February 2004 Archives by author

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Starting: Sun Feb 1 02:09:05 MST 2004
Ending: Sun Feb 29 22:22:26 MST 2004
Messages: 2777

• [Asterisk-Users] Specials on VOIP Gear   John Brown (CV)
• [Asterisk-Users] Hyuandi pstn handsets   John Brown (CV)
• [Asterisk-Users] Job Opening   John Brown (CV)
• [Asterisk-Users] About Grandstream ATA-286 and ring voltage   John Brown (CV)
• [Asterisk-Users] how to route based on DNIS   John Brown (CV)
• [Asterisk-Users] HT 286 Any information about will be great !!!   John Brown (CV)
• [Asterisk-Users] E911 support   John Brown (CV)
• [Asterisk-Users] unsubscribe   与那嶺　弘
• [Asterisk-Users] cisco 7912 problem with chan_sccp   Øyvind Johnsen
• [Asterisk-Users] cisco 7912 problem with chan_sccp   Øyvind Johnsen
• [Asterisk-Users] cisco 7912 problem with chan_sccp   Øyvind Johnsen
• [Asterisk-Users] LDAP authentication   Gustavo García Bernardo
• [Asterisk-Users] SIP Messages (SIMPLE)   Gustavo García Bernardo
• [Asterisk-Users] max asterisk load   Gustavo García Bernardo
• [Asterisk-Users] max asterisk load   Gustavo García Bernardo
• [Asterisk-Users] Spanish newbie need help   Antonio Diego Almodóvar Cebrián
• [Asterisk-Users] SwissVoice IP10S can't take back call   Daniel ANDRE
• [Asterisk-Users] IVR (does not exist in any format)(No such file or directory)   Key Aavoja
• [Asterisk-Users] SIP overlap (early dial) 484 response   Key Aavoja
• [Asterisk-Users] Re: Need to interface to BRIs   Master Abi
• [Asterisk-Users] Hangup to CDR recording timing   Master Abi
• [Asterisk-Users] Loading module chan_capi.so failed!   Nico (Dominik) Ach
• [Asterisk-Users] chan_capi problem   Nico (Dominik) Ach
• [Asterisk-Users] Cisco 7960 SIP image (off-topic)   Adams, Gavin
• [Asterisk-Users] cvs update and new x100p cards broke menu playback   Sean Adams
• [Asterisk-Users] Dial via sip gateway?   Rich Adamson
• [Asterisk-Users] Dial via sip gateway?   Rich Adamson
• [Asterisk-Users] Dial via sip gateway?   Rich Adamson
• [Asterisk-Users] Mediatrix 1204 SIP FXO 4-port gateway review   Rich Adamson
• [Asterisk-Users] Dial via sip gateway?   Rich Adamson
• [Asterisk-Users] Dial via sip gateway?   Rich Adamson
• [Asterisk-Users] Dial via sip gateway?   Rich Adamson
• [Asterisk-Users] Dial via sip gateway?   Rich Adamson
• [Asterisk-Users] How do I provide redundancy and reliability w/ Asterisk?   Rich Adamson
• [Asterisk-Users] Mediatrix 1102 Auth   Rich Adamson
• [Asterisk-Users] Still looking for small fxo sip gateway   Rich Adamson
• [Asterisk-Users] SIP debug logs   Rich Adamson
• [Asterisk-Users] Cisco 7960 quick dial   Rich Adamson
• [Asterisk-Users] Mediatrix sip fxo gateway workaround?   Rich Adamson
• [Asterisk-Users] Still looking for small fxo sip gateway   Rich Adamson
• [Asterisk-Users] [OT] Oldest Telephone   Rich Adamson
• [Asterisk-Users] [OT] Oldest Telephone   Rich Adamson
• [Asterisk-Users] CALEA?   Rich Adamson
• [Asterisk-Users] Cisco 7960 quick dial   Rich Adamson
• [Asterisk-Users] Voicemail volume level?   Rich Adamson
• [Asterisk-Users] Sip flow diagram?   Rich Adamson
• [Asterisk-Users] Sip flow diagram? Answers   Rich Adamson
• [Asterisk-Users] Possible Sip logic bug?   Rich Adamson
• Fw: [Asterisk-Users] Possible Sip logic bug?   Rich Adamson
• [Asterisk-Users] Calls dropping off   Rich Adamson
• [Asterisk-Users] Cisco 7960 bug in 6.1 evident in Asterisk   Rich Adamson
• Fw: [Asterisk-Users] Possible Sip logic bug?   Rich Adamson
• Fw: [Asterisk-Users] Possible Sip logic bug?   Rich Adamson
• [Asterisk-Users] iax2 jitter stats confusion   Rich Adamson
• [Asterisk-Users] Mediatrix 1204 sip g/w now working   Rich Adamson
• [Asterisk-Users] Digium connectivity issue?   Rich Adamson
• [Asterisk-Users] Correct cvs checkout?   Rich Adamson
• [Asterisk-Users] ZapRAS + RADIUS authentication   Rich Adamson
• [Asterisk-Users] IAXTEL and the registration traffic   Rich Adamson
• [Asterisk-Users] FXO gateways on Asterisk   Rich Adamson
• [Asterisk-Users] pstn 800#'s   Rich Adamson
• [Asterisk-Users] Cell Phones and Tone Recognition   Rich Adamson
• [Asterisk-Users] SIP REGISTER   Rich Adamson
• [Asterisk-Users] sip-to-sip hangup problems?   Rich Adamson
• [Asterisk-Users] Woodpeckers   Rich Adamson
• [Asterisk-Users] Woodpeckers   Rich Adamson
• [Asterisk-Users] Woodpeckers   Rich Adamson
• [Asterisk-Users] switch ivr menu on/off remotely?   Rich Adamson
• [Asterisk-Users] 7960 - multiple lines - sip.conf   Rich Adamson
• [Asterisk-Users] SIP extension "busy" when not available ??   Rich Adamson
• [Asterisk-Users] Avaya question   Rich Adamson
• [Asterisk-Users] Is IAXtel down?   Rich Adamson
• [Asterisk-Users] Comments on Voice Quality IP Hard Phones   Rich Adamson
• [Asterisk-Users] Mediatrix 1204 FXO GW ring cadence question   Rich Adamson
• [Asterisk-Users] Cisco 7960 sip v6.2 is out   Rich Adamson
• [Asterisk-Users] Cisco Kit for Grabs   Adthrawn
• [Asterisk-Users] Asterisk Hardware Solution   Adthrawn
• [Asterisk-Users] Callerid detection   Rami AlZaid
• [Asterisk-Users] Callerid detection   Rami AlZaid
• [Asterisk-Users] The Smallest Asterisk Server Ever?   Chris Albertson
• [Asterisk-Users] The Smallest Asterisk Server Ever?   Chris Albertson
• [Asterisk-Users] Smallest server continued...   Chris Albertson
• [Asterisk-Users] voip phones   Chris Albertson
• [Asterisk-Users] Dialing 800 numbers with VOIP   Chris Albertson
• [Asterisk-Users] Dialing 800 numbers with VOIP   Chris Albertson
• [Asterisk-Users] NIC card failure [was: System freeze]   Chris Albertson
• [Asterisk-Users] Dialing 800 numbers with VOIP   Chris Albertson
• [Asterisk-Users] festival voices   Chris Albertson
• [Asterisk-Users] Howto apply a patch, diff file   Chris Albertson
• [Asterisk-Users] Budgetone phones from FWD   Chris Albertson
• [Asterisk-Users] vm to email ?   Chris Albertson
• [Asterisk-Users] Woodpeckers   Chris Albertson
• [Asterisk-Users] Woodpeckers   Chris Albertson
• [Asterisk-Users] Asterisk & Panasonic KXTD - Vonage   Youness El Andaloussi
• [Asterisk-Users] TDM card loses Dial tone   Youness El Andaloussi
• [Asterisk-Users] Cepstral TTS Code   Andreas Anderson
• [Asterisk-Users] 7960 MGCP dialtone problems, part 1 [long]   Andreas Anderson
• [Asterisk-Users] alert-info and Cisco 7960 phones (6.1)   Andreas Anderson
• [Asterisk-Users] Re: Boards falling out...   Colin Anderson
• [Asterisk-Users] (no subject)   Micke Andersson
• [Asterisk-Users] SIP REGISTER   Micke Andersson
• [Asterisk-Users] SIP REGISTER   Micke Andersson
• [Asterisk-Users] mysql_freinds   Micke Andersson
• [Asterisk-Users] SIP REGISTER   Micke Andersson
• [Asterisk-Users] Interrupted musiconhold sound when silence suppression is enabled   Andres
• [Asterisk-Users] Calls dropping off   Andres
• [Asterisk-Users] Interrupted musiconhold sound when silence suppression is enabled   Andres
• [Asterisk-Users] nortel i2004 ip phone   Andres
• [Asterisk-Users] Re: asterisk-grandstream call   Andres
• [Asterisk-Users] Dell 1750 server and Asteriks...   Andres
• [Asterisk-Users] Analog Cordless Phone Recommendations   Andres
• [Asterisk-Users] INFO/DTMF retransmissions in * not absorbed?   Andres
• [Serusers] Re: [Asterisk-Users] INFO/DTMF retransmissions in * not absorbed?   Andres
• [Asterisk-Users] Grandstream transfer into outer space   Andres
• [Asterisk-Users] Asterisk ACD - Avaya ACD   Joelson S. Apon
• [Asterisk-Users] can a variable be redefined within extensions.conf   Lance Arbuckle
• [Asterisk-Users] has Allison recorded "Do Not Disturb"   Lance Arbuckle
• [Asterisk-Users] passing variables to a macro   Lance Arbuckle
• [Asterisk-Users] Snom 100 Code Recommendation   Lance Arbuckle
• [Asterisk-Users] dial timeout not working   Lance Arbuckle
• [Asterisk-Users] dial timeout not working   Lance Arbuckle
• [Asterisk-Users] ringing   Lance Arbuckle
• [Asterisk-Users] Need to interface to BRIs   Jim Archer
• [Asterisk-Users] Need to interface to BRIs   Jim Archer
• [Asterisk-Users] Need to interface to BRIs   Jim Archer
• [Asterisk-Users] Asterisk for a call center?   Jim Archer
• [Asterisk-Users] Hardware requirements   Jim Archer
• [Asterisk-Users] Are IAX2 providers ready for prime time?   Jim Archer
• [Asterisk-Users] Are IAX2 providers ready for prime time?   Jim Archer
• [Asterisk-Users] mgcp endpoint question   Marc Archer
• [Asterisk-Users] Agents / ackcall   Christopher Arnold
• [Asterisk-Users] One2One application?   Christopher Arnold
• [Asterisk-Users] About HT-286   Carlos Arnt
• [Asterisk-Users] Grandstream and HT-286   Carlos Arnt
• [Asterisk-Users] Sipura 2000 SPA-2000, Help Question.   Carlos Arnt
• [Asterisk-Users] HT 286 Any information about will be great !!!   Carlos Arnt
• [Asterisk-Users] Best VOIP Analog adapter ???   Carlos Arnt
• [Asterisk-Users] AutoAttendent ON/OFF control by Attendent   AstGrp
• [Asterisk-Users] CVS Changes (NAT-SIP)   AstGrp
• [Asterisk-Users] Call center integration - passing caller idinto an external app.   AstGrp
• [Asterisk-Users] Music on Hold - Context   AstGrp
• [Asterisk-Users] Music on Hold - Context   AstGrp
• [Asterisk-Users] T1 Help   AstGrp
• [Asterisk-Users] Callerid & AGI Thougts   AstGrp
• [Asterisk-Users] VoicemailMain2   AstGrp
• [Asterisk-Users] multiple lines on 7960's   AstGrp
• [Asterisk-Users] Call Redirection   AstGrp
• [Asterisk-Users] Call Redirection   AstGrp
• [Asterisk-Users] VoiceMail   AstGrp
• [Asterisk-Users] app_directory.c   AstGrp
• [Asterisk-Users] Sip Register Fail - NAT   AstGrp
• [Asterisk-Users] Sip Register Fail - NAT   AstGrp
• [Asterisk-Users] Macro Forward Calls   AstGrp
• [Asterisk-Users] Asterisk as a SIP server behind nat, clients on the outside connecting to Asterisk   AstGrp
• [Asterisk-Users] Translator 'g729tolinb'   AstPBX
• [Asterisk-Users] Call File Troubles   Asterisk
• [Asterisk-Users] Call File Troubles   Asterisk
• [Asterisk-Users] IAX call problems   Rattana BIV
• [Asterisk-Users] IAX call problems   Rattana BIV
• [Asterisk-Users] IAX call problems   Rattana BIV
• [Asterisk-Users] Record conversation   Rattana BIV
• [Asterisk-Users] Record communication   Rattana BIV
• [Asterisk-Users] Problem with call to IAX   Rattana BIV
• [Asterisk-Users] Problem with call to IAX   Rattana BIV
• [Asterisk-Users] Asterisk PABX switch   Nicholas Bachmann
• [Asterisk-Users] Hotel wake-up   Nicholas Bachmann
• [Asterisk-Users] RE:Asterisk PABX switch   Nicholas Bachmann
• [Asterisk-Users] Hotel wake-up   Nicholas Bachmann
• [Asterisk-Users] Hotel wake-up   Nicholas Bachmann
• [Asterisk-Users] Surveys   Nick Bachmann
• [Asterisk-Users] Woodpeckers   Nick Bachmann
• [Asterisk-Users] Adtran 750 Configuration   John Baker
• [Asterisk-Users] Adtran 750 Configuration   John Baker
• [Asterisk-Users] X100P - Asterisk - Asterisk - X100P setup help   John Baker
• [Asterisk-Users] Sip transfers   John Baker
• [Asterisk-Users] busy status   John Baker
• [Asterisk-Users] Intercom system (not paging system)   John Baker
• [Asterisk-Users] Cisco 7960 - how to enable "messages" key   John Baker
• [Asterisk-Users] Direct mailbox transfer   John Baker
• [Asterisk-Users] Adtran 750 - what do I need   John Baker
• [Asterisk-Users] Mailbox full ?   John Baker
• [Asterisk-Users] Simulating the "lighted line in use" type of phone   John Baker
• [Asterisk-Users] InfoElement in chan_capi   Markus Barckmann
• [Asterisk-Users] Dial-out and Dial-in modem problems.   Ariel Batista
• [Asterisk-Users] Sip problem with IpDialog phone.   Ariel Batista
• [Asterisk-Users] Sipura 2000 SPA-2000, Help Question.   Ariel Batista
• [Asterisk-Users] Snom 200 Map key lights problem.   Ariel Batista
• [Asterisk-Users] zaphfc bri with overlap sending/receiving   Jan Baumann
• [Asterisk-Users] zaphfc bri: crackling sound   Jan Baumann
• [Asterisk-Users] Asterisk as a SIP server behind nat, clients on the outside connecting to Asterisk   Steve Beaumont
• [Asterisk-Users] Asterisk pins CPU   Jason Becker
• [Asterisk-Users] Asterisk pins CPU   Jason Becker
• [Asterisk-Users] Best OS for Asterisk   Jason Becker
• [Asterisk-Users] Re: Compiling while * is running   Stephen R. Besch
• [Asterisk-Users] Re: Question on setting up asterisk with hunting lines   Stephen R. Besch
• [Asterisk-Users] Re: iax, trunking, etc.   Stephen R. Besch
• [Asterisk-Users] Re: Anyone used a Grandstream ATA286 with Asterisk   Stephen R. Besch
• [Asterisk-Users] Re: Boards falling out...   Stephen R. Besch
• [Asterisk-Users] Re: Boards falling out...   Stephen R. Besch
• [Asterisk-Users] Re: Compiling while * is running   Stephen R. Besch
• [Asterisk-Users] Re: Boards falling out...   Stephen R. Besch
• [Asterisk-Users] Annoying Beeps   Stephen R. Besch
• [Asterisk-Users] Re: Annoying Beeps   Stephen R. Besch
• [Asterisk-Users] Re: Annoying Beeps   Stephen R. Besch
• [Asterisk-Users] Re: GS BT-100 echo   Stephen R. Besch
• [Asterisk-Users] Re: X100P / Echo / ZTMONITOR CAN2,3, etc.   Stephen R. Besch
• [Asterisk-Users] Re: Double digits seen using Grandstream phones   Stephen R. Besch
• [Asterisk-Users] Red Alarms and mysterious disconnects   Stephen R. Besch
• [Asterisk-Users] Re: Speech between Grandstream phones sounds like talking under water   Stephen R. Besch
• [Asterisk-Users] Re: Woodpeckers   Stephen R. Besch
• [Asterisk-Users] Re: Not Woodpeckers   Stephen R. Besch
• [Asterisk-Users] Re: calls dropped with grandstream   Stephen R. Besch
• [Asterisk-Users] Re: Grandstream transfer into outer space   Stephen R. Besch
• [Asterisk-Users] Re: Grandstream transfer into outer space   Stephen R. Besch
• [Asterisk-Users] Failed to start asterisk   Michiel Betel
• [Asterisk-Users] RE: simple H323 question   Michiel Betel
• [Asterisk-Users] TDM card loses Dial tone   Bob Bevins
• [Asterisk-Users] X100P - Asterisk - Asterisk - X100P setup help   Maninder Bhatia
• [Asterisk-Users] Unable to create vpb channel   Daniel Bichara
• [Asterisk-Users] Termination - Cuba   Daniel Bichara
• [Asterisk-Users] Termination - Cuba   Daniel Bichara
• [Asterisk-Users] PRI error or what?   Daniel Bichara
• [Asterisk-Users] IAX Connection - Voicepulse   Daniel Bichara
• [Asterisk-Users] Voicepulse Connection   Daniel Bichara
• [Asterisk-Users] Connection Problem - GrandStream   Daniel Bichara
• [Asterisk-Users] Codec Order / Preference   Daniel Bichara
• [Asterisk-Users] Codec Order / Preference   Daniel Bichara
• [Asterisk-Users] Brazilian Protocol   Daniel Bichara
• [Asterisk-Users] Brazilian Protocol   Daniel Bichara
• [Asterisk-Users] R2lib   Daniel Bichara
• [Asterisk-Users] System freeze   Jonathan Biggs
• [Asterisk-Users] Hide outgoing CallerID   Mickey Binder
• [Asterisk-Users] Hide outgoing CallerId on Zap interface   Mickey Binder
• [Asterisk-Users] Hide outgoing CallerId on Zap interface   Mickey Binder
• [Asterisk-Users] Hide outgoing CallerId on Zap interface   Mickey Binder
• [Asterisk-Users] Hide outgoing CallerId on Zap interface   Mickey Binder
• [Asterisk-Users] Cisco 7960 quick dial   Bisker, Scott (7805)
• [Asterisk-Users] System freeze   Bisker, Scott (7805)
• [Asterisk-Users] Dial-out and Dial-in modem problems.   Bisker, Scott (7805)
• [Asterisk-Users] Wierd Zap Channel Behavior   Bisker, Scott (7805)
• [Asterisk-Users] T1 Help   Bisker, Scott (7805)
• [Asterisk-Users] Call Pick on Cisco 7960's   Bisker, Scott (7805)
• [Asterisk-Users] Cisco 7960 SIP image (off-topic)   Bisker, Scott (7805)
• [Asterisk-Users] dialing delay question.   John Bittner
• [Asterisk-Users] RE:voiceglo sip config   John Bittner
• [Asterisk-Users] RE:voiceglo sip config   John Bittner
• [Asterisk-Users] dialout redunancy.   John Bittner
• [Asterisk-Users] dialout redunancy.   John Bittner
• [Asterisk-Users] RE: Rhino channel bank and aastra PT390 phones   John Bittner
• [Asterisk-Users] RE: Rhino channel bank and aastra PT390 phones   John Bittner
• [Asterisk-Users] RE: Rhino channel bank and aastra PT390 phones   John Bittner
• [Asterisk-Users] ACD   John Bittner
• [Asterisk-Users] Failed to start asterisk   John Bittner
• [Asterisk-Users] queues   John Bittner
• [Asterisk-Users] HELP!!!! Having problems Starting Asterisk   Robert Boardman
• [Asterisk-Users] HELP!!!! Having problems Starting Asterisk   Robert Boardman
• [Asterisk-Users] HELP!!!! Having problems Starting Asterisk   Robert Boardman
• [Asterisk-Users] Cisco 30VIP Phones   Robert Boardman
• [Asterisk-Users] HELP!!!! Having problems Starting Asterisk   Robert Boardman
• [Asterisk-Users] Incomming Distinctive ringing   Robert Boardman
• [Asterisk-Users] Chan_skinny   Robert Boardman
• [Asterisk-Users] 12SP   Robert Boardman
• [Asterisk-Users] Asterisk 0.7.1 RPMS Updated to Rel 4   Greg Boehnlein
• [Asterisk-Users] The Smallest Asterisk Server Ever?   Greg Boehnlein
• [Asterisk-Users] The Smallest Asterisk Server Ever?   Greg Boehnlein
• [Asterisk-Users] The Smallest Asterisk Server Ever?   Greg Boehnlein
• [Asterisk-Users] [OT] Oldest Telephone   Greg Boehnlein
• [Asterisk-Users] Asterisk 0.7.2   Greg Boehnlein
• [Asterisk-Users] Asterisk 0.7.2 RPMS Updated   Greg Boehnlein
• [Asterisk-Users] Asterisk 0.7.2 RPMS Updated   Greg Boehnlein
• [Asterisk-Users] Asterisk 0.7.2 RPMS Updated   Greg Boehnlein
• [Asterisk-Users] talking clock   Greg Boehnlein
• [Asterisk-Users] talking clock   Greg Boehnlein
• [Asterisk-Users] Asterisk 0.7.2 RPMS Updated   Greg Boehnlein
• [Asterisk-Users] Asterisk 0.7.2 RPMS Updated   Greg Boehnlein
• [Asterisk-Users] s/asterisk mailinglists/asterisk forum/g ?   Greg Boehnlein
• [Asterisk-Users] X100P   Greg Boehnlein
• [Asterisk-Users] Best OS for Asterisk   Greg Boehnlein
• [Asterisk-Users] Dialing 800 numbers with VOIP   Greg Boehnlein
• [Asterisk-Users] The Smallest Asterisk Server Ever?   Greg Boehnlein
• [Asterisk-Users] Wait command in auto attendant causes sched.c error   Greg Boehnlein
• [Asterisk-Users] Dialing 800 numbers with VOIP   Greg Boehnlein
• [Asterisk-Users] VOIP Carrier recommendations?   Greg Boehnlein
• [Asterisk-Users] Speech between Grandstream phones sounds like talking under water   Greg Boehnlein
• [Asterisk-Users] Agents / ackcall   Greg Boehnlein
• [Asterisk-Users] Agents / ackcall   Greg Boehnlein
• [Asterisk-Users] Agents / ackcall   Greg Boehnlein
• [Asterisk-Users] Agents / ackcall   Greg Boehnlein
• [Asterisk-Users] Agents / ackcall   Greg Boehnlein
• [Asterisk-Users] Agents / ackcall   Greg Boehnlein
• [Asterisk-Users] Agents / ackcall   Greg Boehnlein
• [Asterisk-Users] Agents / ackcall   Greg Boehnlein
• [Asterisk-Users] Queue Modified ACD for Asterisk 0.7.2   Greg Boehnlein
• [Asterisk-Users] OT: SNOM and TAPI   Greg Boehnlein
• [Asterisk-Users] Understanding AgentCallbackLogin   Greg Boehnlein
• [Asterisk-Users] Understanding AgentCallbackLogin   Greg Boehnlein
• [Asterisk-Users] Understanding AgentCallbackLogin   Greg Boehnlein
• [Asterisk-Users] Understanding AgentCallbackLogin   Greg Boehnlein
• [Asterisk-Users] Understanding AgentCallbackLogin   Greg Boehnlein
• [Asterisk-Users] Understanding AgentCallbackLogin   Greg Boehnlein
• [Asterisk-Users] Simulating the "lighted line in use" type of phone   Greg Boehnlein
• [Asterisk-Users] (OT) HOWTO: 802.3af POE w/ 79xx   Greg Boehnlein
• [Asterisk-Users] Cisco 7960 - how to enable "messages" key   Greg Boehnlein
• [Asterisk-Users] Release phone call   B. J. Bomar
• [Asterisk-Users] Release phone call   B. J. Bomar
• [Asterisk-Users] System freeze   B. J. Bomar
• [Asterisk-Users] Call Queues   B. J. Bomar
• [Asterisk-Users] Call Pick on Cisco 7960's   B. J. Bomar
• [Asterisk-Users] Call Pick on Cisco 7960's   B. J. Bomar
• [Asterisk-Users] Asterisk as non root   Fran Boon
• [Asterisk-Users] Asterisk 0.7.2 RPMS Updated   Fran Boon
• [Asterisk-Users] Re: Speex == Screech using version 1.1.4   Fran Boon
• [Asterisk-Users] running asterisk as non-root   Fran Boon
• [Asterisk-Users] SIP config documentation   Fran Boon
• [Asterisk-Users] SIP config documentation   Fran Boon
• [Asterisk-Users] Re: [Asterisk-Users] Re: [Asterisk-Users] SIP config documentation   Fran Boon
• [Asterisk-Users] Re: [Asterisk-Users] Re: [Asterisk-Users] SIP config documentation   Fran Boon
• [Asterisk-Users] SIP config documentation   Fran Boon
• [Asterisk-Users] Pickup   Fran Boon
• [Asterisk-Users] Codec Order / Preference   Fran Boon
• [Asterisk-Users] voicemail not working with mysql!!!!   Fran Boon
• [Asterisk-Users] voicemail not working with mysql!!!!   Fran Boon
• [Asterisk-Users] exit   Fran Boon
• [Asterisk-Users] retrieve_sip_conf_from_mysql.pl data format   Fran Boon
• [Asterisk-Users] retrieve_sip_conf_from_mysql.pl data format   Fran Boon
• [Asterisk-Users] retrieve_sip_conf_from_mysql.pl data format   Fran Boon
• [Asterisk-Users] SIP phones with dual ethernet.   Nicolas Bougues
• [Asterisk-Users] SIP phones with dual ethernet.   Nicolas Bougues
• [Asterisk-Users] SIP phones with dual ethernet.   Nicolas Bougues
• [Asterisk-Users] New Firmware for Grandstream Phones - Supports CFG by MAC address   Nicolas Bougues
• [Asterisk-Users] Speech between Grandstream phones sounds like talking under water   Nicolas Bougues
• [Asterisk-Users] Disabling echo cancellation on fax ?   Nicolas Bougues
• [Asterisk-Users] About Grandstream ATA-286 and ring voltage   Nicolas Bougues
• [Asterisk-Users] faxing with asterisk   Stephen Bradwell
• [Asterisk-Users] Playing announcement to called user prior to Confirmation   Matteo Brancaleoni
• [Asterisk-Users] busy tones   Matteo Brancaleoni
• [Asterisk-Users] Mediatrix 1102 Auth   Matteo Brancaleoni
• [Asterisk-Users] Choppy Problem!!   Matteo Brancaleoni
• [Asterisk-Users] Mediatrix 1102 Auth   Matteo Brancaleoni
• [Asterisk-Users] GS and NAT   Matteo Brancaleoni
• [Asterisk-Users] Multiple switch staments   Matteo Brancaleoni
• [Asterisk-Users] Music on Hold - Context   Matteo Brancaleoni
• [Asterisk-Users] Need to interface to BRIs   Matteo Brancaleoni
• [Asterisk-Users] Zhone + call transfer   Matteo Brancaleoni
• [Asterisk-Users] Re: Need to interface to BRIs   Matteo Brancaleoni
• [Asterisk-Users] Lucent Definity CallerID   Matthew Branton
• [Asterisk-Users] Lucent Definity CallerID {Scanned}   Matthew Branton
• [Asterisk-Users] Lucent Definity CallerID {Scanned}   Matthew Branton
• [Asterisk-Users] Agent Queuing on multiple machines   Matthew Branton
• [Asterisk-Users] Lucent Definity CallerID {Scanned}   Matthew Branton
• [Asterisk-Users] running asterisk as non-root   Birk Bremer
• [Asterisk-Users] Anybody managed to call a phone through sipgate.de   Birk Bremer
• [Asterisk-Users] Anybody managed to call a phone through sipgate.de   Birk Bremer
• [Asterisk-Users] Anybody managed to call a phone through sipgate.de   Birk Bremer
• [Asterisk-Users] Anybody managed to call a phone through sipgate.de   Birk Bremer
• [Asterisk-Users] Anybody managed to call a phone through sipgate.de   Birk Bremer
• [Asterisk-Users] Trying to dial out the Other Zap   Brian
• [Asterisk-Users] Cell Phones and Tone Recognition   Brian
• [Asterisk-Users] Cell Phones and Tone Recognition   Brian
• [Asterisk-Users] Cell Phones and Tone Recognition   Brian
• [Asterisk-Users] Cell Phones and Tone Recognition   Brian
• [Asterisk-Users] 4 port FXO   Chad Brown
• [Asterisk-Users] VoiceMail   Chad Brown
• [Asterisk-Users] Voicemail Follow-up   Chad Brown
• [Asterisk-Users] Bizarre ring   Derek Bruce
• [Asterisk-Users] Bizarre ring   Derek Bruce
• [Asterisk-Users] Cisco AS5350 Gateway Intergration   Derek Bruce
• [Asterisk-Users] Remote retrieval of voicemail, a question   Brian Buhrow
• [Asterisk-Users] FXO Gateway of choice is?   Brian Buhrow
• [Asterisk-Users] festival voices   Tony Buser
• [Asterisk-Users] festival voices   Tony Buser
• [Asterisk-Users] festival voices   Tony Buser
• [Asterisk-Users] phpagi DIAL command not working   Tony Buser
• [Asterisk-Users] phpagi DIAL command not working   Tony Buser
• [Asterisk-Users] compiling gastman   Tony Buser
• [Asterisk-Users] Skinny and SIP   Ryan Butler
• [Asterisk-Users] Problems with ATA's locking up..   CW_ASN
• [Asterisk-Users] Problems with ATA's locking up..   CW_ASN
• [Asterisk-Users] Problems with ATA's locking up..   CW_ASN
• [Asterisk-Users] Get new PRI working   CW_ASN
• [Asterisk-Users] Can audio streams go client to cleint with IAX?   Brian Capouch
• [Asterisk-Users] Cepstral TTS Code   Brian Capouch
• [Asterisk-Users] Cepstral TTS Code   Brian Capouch
• [Asterisk-Users] Cepstral TTS Code   Brian Capouch
• [Asterisk-Users] CallWaiting CallerID: Available on all channel types?   Brian Capouch
• [Asterisk-Users] Speex == Screech using version 1.1.4   Brian Capouch
• [Asterisk-Users] cannot find -lXext when building * ?   Brian Capouch
• [Asterisk-Users] Anyone working with NUFONE?   Brian Capouch
• [Asterisk-Users] Anyone working with NUFONE?   Brian Capouch
• [Asterisk-Users] Simulating the "lighted line in use" type of phone   Brian Capouch
• [Asterisk-Users] Simulating the "lighted line in use" type of phone   Brian Capouch
• [Asterisk-Users] Sorry, OT (NuFone)   Brian Capouch
• [Asterisk-Users] A working number at enum.fierymoon.com?   Brian Capouch
• [Asterisk-Users] Testing ENUM   Brian Capouch
• [Asterisk-Users] Superbowl = Linux Shake up to the world..   Juan Cardenas
• [Asterisk-Users] Playing GSM files(s)   Juan Cardenas
• [Asterisk-Users] Background command, error   Juan Cardenas
• [Asterisk-Users] max asterisk load   Juan Cardenas
• [Asterisk-Users] Re: IAX Voicepulse - no DTMF response   Juan Cardenas
• [Asterisk-Users] DTMF Issues with SJPHONE   Carl
• [Asterisk-Users] Iconnect behind NAT   Carl
• [Asterisk-Users] Iconnect behind NAT   Carl
• [Asterisk-Users] Help needed setting up H323 gateway.   Carl
• [Asterisk-Users] Delta Three/iConnectHere Outgoing Caller ID?   Nate Carlson
• [Asterisk-Users] The Smallest Asterisk Server Ever?   David J Carter
• [Asterisk-Users] incoming call to internal user   David J Carter
• [Asterisk-Users] New Firmware for Grandstream Phones - Supports CFG by MAC address   David J Carter
• [Asterisk-Users] Jump to extension from voice menu   David J Carter
• [Asterisk-Users] HELP!!!! Having problems Starting Asterisk   David J Carter
• [Asterisk-Users] Budgetone phones from FWD   David J Carter
• [Asterisk-Users] softphone configs?   David J Carter
• [Asterisk-Users] Unable to create channel of type 'Zap'   David J Carter
• [Asterisk-Users] Off topic question   David J Carter
• [Asterisk-Users] Anybody managed to call a phone through sipgate.de   David J Carter
• [Asterisk-Users] Anybody managed to call a phone through sipgate.de   David J Carter
• [Asterisk-Users] Anybody managed to call a phone through sipgate.de   David J Carter
• [Asterisk-Users] Anybody managed to call a phone through sipgate.de   David J Carter
• [Asterisk-Users] wisip firmware, updates, features??   David J Carter
• [Asterisk-Users] Asterisk 0.7.2   Miguel Cavazos
• [Asterisk-Users] Voiceglo questions, IAX   Miguel Cavazos
• [Asterisk-Users] Wifi Phones   Miguel Cavazos
• [Asterisk-Users] Wifi Phones   Miguel Cavazos
• [Asterisk-Users] wisip firmware, updates, features??   Miguel Cavazos
• [Asterisk-Users] Cisco 7960 SIP image (off-topic)   Heison Chak
• [Asterisk-Users] Surveys   Heison Chak
• [Asterisk-Users] Codec Order / Preference   Heison Chak
• [Asterisk-Users] SIP over NAT   Heison Chak
• [Asterisk-Users] calls dropped with grandstream   Heison Chak
• [Asterisk-Users] Call priority   Heison Chak
• [Asterisk-Users] Can asterisk make a call to a phone?   John Chambers
• [Asterisk-Users] Can audio streams go client to cleint with IAX?   T. Chan
• [Asterisk-Users] Can audio streams go client to cleint with IAX?   T. Chan
• [Asterisk-Users] Fast question on extension matching   T. Chan
• [Asterisk-Users] Fast question on extension matching   T. Chan
• [Asterisk-Users] Faxing in passthrough and codec in passthrough   T. Chan
• [Asterisk-Users] RE: codec negotiation prob solved   T. Chan
• [Asterisk-Users] RE: simple H323 question   T. Chan
• [Asterisk-Users] RE: simple H323 question   T. Chan
• [Asterisk-Users] Problem with * - Nat - Internet - Nat - X-lite   Carlos Chavez
• [Asterisk-Users] DIAX does not receive calls...   Carlos Chavez
• [Asterisk-Users] DIAX does not receive calls...   Carlos Chavez
• [Asterisk-Users] GS Budgetone 101 canot receive calls   Carlos Chavez
• [Asterisk-Users] Re: how to dial and accept a call with only   Sean Cheesman
• [Asterisk-Users] simple test setup   Sean Cheesman
• [Asterisk-Users] zaptel on Debian   Sean Cheesman
• [Asterisk-Users] modprobe wcfxs   Sean Cheesman
• [Asterisk-Users] modprobe wcfxs   Sean Cheesman
• [Asterisk-Users] TDM400 showing up as Tiger Jet   Sean Cheesman
• [Asterisk-Users] Inbound IAX to SIP   Sean Cheesman
• [Asterisk-Users] Gastman doesn't draw lines properly between resources ...   Sean Cheesman
• [Asterisk-Users] New Wiki page: Dimensioning an Asterisk system   Sean Cheesman
• [Asterisk-Users] New Wiki page: Dimensioning an Asterisk system   Sean Cheesman
• [Asterisk-Users] Nested include statements in extensions.conf?   Sean Cheesman
• [Asterisk-Users] WTB: Grandstream Budgetone   Brian Christie
• [Asterisk-Users] extension mobility   Chris Clifton
• [Asterisk-Users] extension mobility   Chris Clifton
• [Asterisk-Users] iax, trunking, etc.   Chris Clifton
• [Asterisk-Users] Cisco 7960 bug in 6.1 evident in Asterisk   Chris Clifton
• [Asterisk-Users] Cisco 7960 bug in 6.1 evident in Asterisk   Chris Clifton
• [Asterisk-Users] busy status   Chris Clifton
• [Asterisk-Users] busy status   Chris Clifton
• [Asterisk-Users] busy status   Chris Clifton
• [Asterisk-Users] busy status   Chris Clifton
• [Asterisk-Users] SIP phones with dual ethernet.   Chris Clifton
• [Asterisk-Users] Dialing 800 numbers with VOIP   Chris Clifton
• [Asterisk-Users] merlin legend / * as ld gw   Chris Clifton
• [Asterisk-Users] VOIP Carrier recommendations?   Chris Clifton
• [Asterisk-Users] line status   Chris Clifton
• [Asterisk-Users] Simulating the "lighted line in use" type of phone   Chris Clifton
• [Asterisk-Users] parking ?   Chris Clifton
• [Asterisk-Users] Conference and transfer   Chris Clifton
• [Asterisk-Users] Sorry, OT (NuFone)   Chris Clifton
• [Asterisk-Users] pstn 800#'s   Netlabz - Chris Clifton
• [Asterisk-Users] vm to email ?   Netlabz - Chris Clifton
• [Asterisk-Users] Re: vm to email ?   Netlabz - Chris Clifton
• [Asterisk-Users] multiple lines on 7960's   Netlabz - Chris Clifton
• [Asterisk-Users] 7960 - multiple lines - sip.conf   Netlabz - Chris Clifton
• [Asterisk-Users] 7960 - multiple lines - sip.conf   Netlabz - Chris Clifton
• [Asterisk-Users] Room Monitor   Jamin W. Collins
• [Asterisk-Users] Room Monitor   Jamin W. Collins
• [Asterisk-Users] Room Monitor   Jamin W. Collins
• [Asterisk-Users] Room Monitor   Jamin W. Collins
• [Asterisk-Users] Room Monitor   Jamin W. Collins
• [Asterisk-Users] Room Monitor   Jamin W. Collins
• [Asterisk-Users] IPKall->FWD->Asterisk   Joshua Colp
• [Asterisk-Users] IPKall->FWD->Asterisk   Joshua Colp
• [Asterisk-Users] Enumlookup and Asterisk   Daniel Concepcion
• [Asterisk-Users] x100p card conflicts with DSL modem   Dave Cotton
• [Asterisk-Users] Do you Linux softphone..   Dave Cotton
• [Asterisk-Users] A missing argument   Dave Cotton
• [Asterisk-Users] A missing argument   Dave Cotton
• [Asterisk-Users] Web based UA   Dave Cotton
• [Asterisk-Users] Digium connectivity issue?   David Coulson
• [Asterisk-Users] play_and_record: No audio available   Ryan Courtnage
• [Asterisk-Users] play_and_record: No audio available   Ryan Courtnage
• [Asterisk-Users] Marketing collateral, etc.   Ryan Courtnage
• [Asterisk-Users] Mac X-Lite and Asterisk   Ryan Courtnage
• [Asterisk-Users] vmail.cgi -> .php?   Ryan Courtnage
• [Asterisk-Users] vmail.cgi -> .php?   Ryan Courtnage
• [Asterisk-Users] Fast question on extension matching   Chris Craft
• [Asterisk-Users] Simulating the "lighted line in use" type of phone   Craig
• [Asterisk-Users] Re: DISA   Steve Creel
• [Asterisk-Users] alert-info and Cisco 7960 phones (6.1)   Steve Creel
• [Asterisk-Users] chan_local and variables   Steve Creel
• [Asterisk-Users] chan_local and variables   Steve Creel
• [Asterisk-Users] merlin legend / * as ld gw   Steve Creel
• [Asterisk-Users] Round-robin chan_zap groups...   Steve Creel
• [Asterisk-Users] Cisco 7960 SIP image (off-topic)   Steve Creel
• [Asterisk-Users] Simple Front Gate Intercom   Steve Creel
• [Asterisk-Users] Simple Front Gate Intercom   Steve Creel
• [Asterisk-Users] Simple Front Gate Intercom   Steve Creel
• [Asterisk-Users] Easy access to visual busy status and call transfer buttons   Jeff Crews
• [Asterisk-Users] Controlling queue size and queue options   Jeff Crews
• [Asterisk-Users] Compiling while * is running   Steven Critchfield
• [Asterisk-Users] Compiling while * is running   Steven Critchfield
• [Asterisk-Users] compile error (still having problems)   Steven Critchfield
• [Asterisk-Users] compile error (still having problems)   Steven Critchfield
• [Asterisk-Users] cdr mysql problem   Steven Critchfield
• [Asterisk-Users] Calls dropping off   Steven Critchfield
• [Asterisk-Users] Compiling while * is running   Steven Critchfield
• [Asterisk-Users] Automated Dialing / Recording ?   Steven Critchfield
• [Asterisk-Users] Automated Dialing / Recording ?   Steven Critchfield
• [Asterisk-Users] Dialling Hook Flash on Zaptel   Steven Critchfield
• [Asterisk-Users] Dialling Hook Flash on Zaptel   Steven Critchfield
• [Asterisk-Users] Dialling Hook Flash on Zaptel   Steven Critchfield
• [Asterisk-Users] The Smallest Asterisk Server Ever?   Steven Critchfield
• [Asterisk-Users] The Smallest Asterisk Server Ever?   Steven Critchfield
• [Asterisk-Users] Re: Boards falling out...   Steven Critchfield
• [Asterisk-Users] CALEA?   Steven Critchfield
• [Asterisk-Users] Re: Execute command in shell   Steven Critchfield
• [Asterisk-Users] Annoying Beeps   Steven Critchfield
• [Asterisk-Users] Asterisk on ebay.   Steven Critchfield
• [Asterisk-Users] port number keeps changing   Steven Critchfield
• [Asterisk-Users] System freeze   Steven Critchfield
• [Asterisk-Users] Dial-out and Dial-in modem problems.   Steven Critchfield
• [Asterisk-Users] X100P Cards have gone belly up?   Steven Critchfield
• [Asterisk-Users] NIC card failure [was: System freeze]   Steven Critchfield
• [Asterisk-Users] Sending DTMF out-of-band over IAX2   Steven Critchfield
• [Asterisk-Users] Call center integration - passing caller id into an external app.   Steven Critchfield
• [Asterisk-Users] Ghost Calls   Steven Critchfield
• [Asterisk-Users] I need patch for musiconhold-multiful format   Steven Critchfield
• [Asterisk-Users] T1 PRI CallerID   Steven Critchfield
• [Asterisk-Users] T1 PRI CallerID   Steven Critchfield
• [Asterisk-Users] Asterisk and Wildcard T100P   Steven Critchfield
• [Asterisk-Users] Asterisk and Wildcard T100P   Steven Critchfield
• [Asterisk-Users] System freeze   Steven Critchfield
• [Asterisk-Users] billing question   Steven Critchfield
• [Asterisk-Users] ZapRAS + RADIUS authentication   Steven Critchfield
• [Asterisk-Users] Analogical FXO vs. BRI dialing speed   Steven Critchfield
• [Asterisk-Users] Cisco 30VIP Phones   Steven Critchfield
• [Asterisk-Users] ZapRAS + RADIUS authentication   Steven Critchfield
• [Asterisk-Users] Got my DID, getting an error.   Steven Critchfield
• [Asterisk-Users] How can you savage a failed call transfer   Steven Critchfield
• [Asterisk-Users] Re: Need to interface to BRIs   Steven Critchfield
• [Asterisk-Users] Analogical FXO vs. BRI dialing speed   Steven Critchfield
• [Asterisk-Users] Dell 1750 server and Asteriks...   Steven Critchfield
• [Asterisk-Users] Callerid & AGI Thougts   Steven Critchfield
• [Asterisk-Users] agi scripting in perl - dealiing with unexpected disconnects gracefully / spurious DTMF   Steven Critchfield
• [Asterisk-Users] Memory usage   Steven Critchfield
• [Asterisk-Users] Where to download app_pgsql   Steven Critchfield
• [Asterisk-Users] Dell 1750 server and Asteriks...   Steven Critchfield
• [Asterisk-Users] Thread-safe applications   Steven Critchfield
• [Asterisk-Users] Processor load spikes   Steven Critchfield
• [Asterisk-Users] Thread-safe applications   Steven Critchfield
• [Asterisk-Users] TE410P E1 PRI problem   Tomica Crnek
• [Asterisk-Users] cdr mysql problem   Tomica Crnek
• [Asterisk-Users] cdr mysql problem   Tomica Crnek
• [Asterisk-Users] cdr mysql problem   Tomica Crnek
• [Asterisk-Users] cdr mysql problem   Tomica Crnek
• [Asterisk-Users] billing information from telecom   Tomica Crnek
• [Asterisk-Users] Data call transfer   Tomica Crnek
• [Asterisk-Users] Data call transfer   Tomica Crnek
• [Asterisk-Users] Incomplete dialed number in CDR   Tomica Crnek
• [Asterisk-Users] Dual line Skinny   Tomica Crnek
• [Asterisk-Users] Dual line Skinny   Tomica Crnek
• [Asterisk-Users] Calls dropping off   Tomica Crnek
• [Asterisk-Users] Calls dropping off   Tomica Crnek
• [Asterisk-Users] Asterisk - oh323 - Cisco CallManager   Tomica Crnek
• [Asterisk-Users] Asterisk - oh323 - Cisco CallManager   Tomica Crnek
• [Asterisk-Users] ZapRAS + RADIUS authentication   Tomica Crnek
• [Asterisk-Users] ZapRAS + RADIUS authentication   Tomica Crnek
• [Asterisk-Users] PRI error or what?   Tomica Crnek
• [Asterisk-Users] PRI error or what?   Tomica Crnek
• [Asterisk-Users] pri error   Tomica Crnek
• [Asterisk-Users] 2 questions about ISDN BRI   Tomica Crnek
• [Asterisk-Users] 2 questions about ISDN BRI   Tomica Crnek
• [Asterisk-Users] 2 questions about ISDN BRI   Tomica Crnek
• [Asterisk-Users] 2 questions about ISDN BRI   Tomica Crnek
• [Asterisk-Users] Port bind   Vic Cross
• [Asterisk-Users] chan_sccp: incoming calls on multiple lines   Vic Cross
• [Asterisk-Users] Trouble emailing Digium   Vic Cross
• [Asterisk-Users] chan_sccp: incoming calls on multiple lines   Vic Cross
• [Asterisk-Users] Interrupted musiconhold sound when silence suppression is enabled   Vic Cross
• [Asterisk-Users] Caller-ID is being sent wrong. How to fix it?   Vic Cross
• [Asterisk-Users] Dual line Skinny   Vic Cross
• [Asterisk-Users] Dual line Skinny   Vic Cross
• [Asterisk-Users] SIP extension "busy" when not available ??   Vic Cross
• [Asterisk-Users] Problem with SIP 407   Vic Cross
• [Asterisk-Users] cisco 7912 problem with chan_sccp   Vic Cross
• [Asterisk-Users] Off topic question   Vic Cross
• [Asterisk-Users] Need to interface to BRIs   Jan Czmok
• [Asterisk-Users] Skinny and SIP   Jan Czmok
• [Asterisk-Users] cisco 7912 problem with chan_sccp   Jan Czmok
• [Asterisk-Users] Grandstream + X100P incoming calls being dropped   Ed Dack
• [Asterisk-Users] Wifi Phones   Yifang Dai
• [Asterisk-Users] Wifi Phones   Yifang Dai
• [Asterisk-Users] Release phone call   Glenn Dalgliesh
• [Asterisk-Users] Release phone call   Glenn Dalgliesh
• [Asterisk-Users] Vegastream 50 FXO with Asterisk   Glenn Dalgliesh
• [Asterisk-Users] asterisk-grandstream call   Glenn Dalgliesh
• [Asterisk-Users] asterisk-grandstream call   Glenn Dalgliesh
• [Asterisk-Users] TDM card loses Dial tone   Glenn Dalgliesh
• [Asterisk-Users] ADSI ports   Glenn Dalgliesh
• [Asterisk-Users] ADSI ports   Glenn Dalgliesh
• [Asterisk-Users] 4 port FXO   Glenn Dalgliesh
• [Asterisk-Users] IAX call problems   Dan
• [Asterisk-Users] diax softphone   Dan
• [Asterisk-Users] diax softphone   Dan
• [Asterisk-Users] X100P and PSTN line Callwaiting   Dan
• [Asterisk-Users] IAX2 (was: diax softphone )   Dan
• [Asterisk-Users] IAX2 (was: diax softphone )   Dan
• [Asterisk-Users] DIAX 0.9.6b call reception   Dan
• [Asterisk-Users] Re: DIAX 0.9.6b call reception   Dan
• [Asterisk-Users] IAXTEL and the registration traffic   Dan
• [Asterisk-Users] Analog Cordless Phone Recommendations   Dan
• [Asterisk-Users] IAXTEL and the registration traffic   Dan
• [Asterisk-Users] IAXTEL and the registration traffic   Dan
• [Asterisk-Users] DIAX does not receive calls...   Dan
• [Asterisk-Users] DIAX does not receive calls...   Dan
• [Asterisk-Users] IAX2 dosent work for DIAX   Dan
• [Asterisk-Users] Re: Adtran 750 DID question.   Kekin Dand
• [Asterisk-Users] Audiocodes FXO - loop channels   Daniel Cubero Salas, Ing
• [Asterisk-Users] speech recognition system   Daniele
• [Asterisk-Users] New Firmware for Grandstream Phones - Supports CFG by MAC address   Jens Davidsen
• [Asterisk-Users] New Firmware for Grandstream Phones -Supports CFG by MAC address   Jens Davidsen
• [Asterisk-Users] iax2 jitter buffer help   Stephen Davies
• [OT] South african laws - was [Asterisk-Users] iax2 jitter buffer help   Stephen Davies
• [Asterisk-Users] I finally did IT!!!! Internal dial tone   Stephen Davies
• [Asterisk-Users] IAXTEL and the registration traffic   Stephen Davies
• [Asterisk-Users] Setup ?   Derek Dempsay
• [Asterisk-Users] Superbowl = Linux Shake up to the world..   Joe Dennick
• [Asterisk-Users] how to dial and accept a call with only x100p card on Redhat linux 9.0?   Joe Dennick
• [Asterisk-Users] cdr mysql problem   Joe Dennick
• [Asterisk-Users] Vegastream 50 BRI   Michael Devenijn
• [Asterisk-Users] Re: DISA   Ed Devine
• [Asterisk-Users] Re: DISA   Ed Devine
• [Asterisk-Users] dtmf recording record and playback   Ed Devine
• [Asterisk-Users] Dual Xeon   Ed Devine
• [Asterisk-Users] exit   Ed Devine
• [Asterisk-Users] Problems with ATA's locking up..   Thomas Dingermann
• [Asterisk-Users] Annoying Beeps   Shawn L. Djernes
• [Asterisk-Users] Sip flow diagram?   Steve Dolloff
• [Asterisk-Users] Memory usage   Steve Dolloff
• [Asterisk-Users] Memory usage   Steve Dolloff
• [Asterisk-Users] chan_sip support for SIP:Remote-Party-ID, specificallyCLID priva cy   Steve Dolloff
• [Asterisk-Users] Basic Sip proxy setup question   Jeff Donovan
• [Asterisk-Users] unable to open ../voicemail/context/exten/msg0000   Jeff Donovan
• [Asterisk-Users] Codecs compile error on yellowdog   Jeff Donovan
• [Asterisk-Users] Re: Codecs compile error on yellowdog   Jeff Donovan
• [Asterisk-Users] Web based UA   Michael Van Donselaar
• [Asterisk-Users] USB Phones   Michael Van Donselaar
• [Asterisk-Users] iaxComm updates at sourceforge   Michael Van Donselaar
• [Asterisk-Users] no extension in callerid of outgoing calls ...   Johannes von Drachenfels
• AW: [Asterisk-Users] no extension in callerid of outgoing calls ...   Johannes von Drachenfels
• [Asterisk-Users] Speech between Grandstream phones sounds like talking under water   Rana Dutt
• [Asterisk-Users] Speech between Grandstream phones sounds like talking under water   Rana Dutt
• [Asterisk-Users] Double digits seen using Grandstream phones   Rana Dutt
• [Asterisk-Users] Re: Double digits seen using Grandstream phones   Rana Dutt
• [Asterisk-Users] Buzzing on Grandstream phones   Rana Dutt
• [Asterisk-Users] Howto apply a patch, diff file   Rana Dutt
• [Asterisk-Users] softphone configs?   Rana Dutt
• [Asterisk-Users] Slow, distorted speech on voice mail messages   Rana Dutt
• [Asterisk-Users] Voice mail sound distortion has everyone laughing   Rana Dutt
• [Asterisk-Users] Could voice mail problem be related to RAM?   Rana Dutt
• [Asterisk-Users] Conference and transfer   Rana Dutt
• [Asterisk-Users] Conference and transfer   Rana Dutt
• [Asterisk-Users] Message waiting light not coming on   Rana Dutt
• [Asterisk-Users] Message waiting light not coming on   Rana Dutt
• [Asterisk-Users] Dialing out after caller leaves message   Rana Dutt
• [Asterisk-Users] Dialing out after caller leaves message   Rana Dutt
• [Asterisk-Users] Playing announcement to called user prior to Confirmation   Kris Edwards
• [Asterisk-Users] Whats wrong with dialplan?   Blake Van Eekeren
• [Asterisk-Users] NewB: Cisco 7910   Stuart Elvish
• [Asterisk-Users] Cisco 7960 SIP image (off-topic)   Matthew Enger
• [Asterisk-Users] extravagant behavior, nat problem ?   Eric_Doiron
• [Asterisk-Users] Can audio streams go client to cleint with IAX?   Marc Fargas
• [Asterisk-Users] Can audio streams go client to cleint with IAX?   Marc Fargas
• [Asterisk-Users] Can audio streams go client to cleint with IAX?   Marc Fargas
• [Asterisk-Users] Can audio streams go client to cleint with IAX?   Marc Fargas
• [Asterisk-Users] Execute command in shell   Marc Fargas
• [Asterisk-Users] Execute command in shell   Marc Fargas
• [Asterisk-Users] Re: Execute command in shell   Marc Fargas
• [Asterisk-Users] Mailbox full ?   Marc Fargas
• [Asterisk-Users] Microsoft Portrait 2.2   Marc Fargas
• [Asterisk-Users] Executing external script   Marc Fargas
• [Asterisk-Users] SIP over NAT   Marc Fargas
• [Asterisk-Users] Flash application on H323   Marc Fargas
• [Asterisk-Users] Problem with SIP 407   Marc Fargas
• [Asterisk-Users] Problem with SIP 407   Marc Fargas
• [Asterisk-Users] Problem with SIP 407   Marc Fargas
• [Asterisk-Users] Jitter Buffer Configuration (typo in iax.conf)   Michael T Farnworth
• [Asterisk-Users] Where to download app_pgsql   Michael T Farnworth
• [Asterisk-Users] Where to download app_pgsql   Michael T Farnworth
• [Asterisk-Users] app_sql_postgres doesn't clean up   Michael T Farnworth
• [Asterisk-Users] Grandstream transfer into outer space   Michael T Farnworth
• [Asterisk-Users] Conference and transfer   Michael T Farnworth
• [Asterisk-Users] Silencing Background App during touch tone detection   Mark Farver
• [Asterisk-Users] Silencing Background App during touch tone detection   Mark Farver
• [Asterisk-Users] HFC-S cards?   FastJack
• [Asterisk-Users] ISDN LCR USING ASTERISK   FastJack
• [Asterisk-Users] ISDN LCR USING ASTERISK   FastJack
• [Asterisk-Users] HFC-S cards?   FastJack
• [Asterisk-Users] Re: HFC-S cards?   FastJack
• [Asterisk-Users] Zaptel BRI and HFC-S cards   FastJack
• [Asterisk-Users] tapi for asterisk*   FastJack
• [Asterisk-Users] Call Redirection   FastJack
• [Asterisk-Users] SEGFAULT (capi amd hfc-s NT)   FastJack
• [Asterisk-Users] calling between two zap points with zaphfc   FastJack
• [Asterisk-Users] OT: SNOM and TAPI   FastJack
• [Asterisk-Users] *8# and zaphfc in NT-mode   FastJack
• [Asterisk-Users] Fax   Simon Faulkner
• [Asterisk-Users] Fax   Simon Faulkner
• [Asterisk-Users] Firefly and IXATEL   Barry Fawthrop
• [Asterisk-Users] Asterisk Faxing   Barry Fawthrop
• [Asterisk-Users] Asterisk Faxing   Barry Fawthrop
• [Asterisk-Users] Asterisk Faxing   Barry Fawthrop
• [Asterisk-Users] Asterisk and Faxing   Barry Fawthrop
• Spam Alert: [Asterisk-Users] Is IAXtel down?   Barry Fawthrop
• [Asterisk-Users] Big Install examples please   Barry Fawthrop
• [Asterisk-Users] cdr_addon_mysql problem linking   Dan Fernandez
• [Asterisk-Users] FIXED : cdr_addon_mysql problem linking   Dan Fernandez
• [Asterisk-Users] Sorry, OT (NuFone)   Joseph Finley
• [Asterisk-Users] Sorry, OT (NuFone)   Joseph Finley
• [Asterisk-Users] CALEA?   Ryan Finnesey
• [Asterisk-Users] Voip in the EU   Ryan Finnesey
• [Asterisk-Users] Voip in the EU   Ryan Finnesey
• [Asterisk-Users] Voip in the EU   Ryan Finnesey
• [Asterisk-Users] Cisco 7960 SIP image (off-topic)   Ryan Finnesey
• [Asterisk-Users] Cisco 7960 SIP image (off-topic)   Ryan Finnesey
• [Asterisk-Users] VoIP Reseller Programs   Ryan Finnesey
• [Asterisk-Users] Can audio streams go client to cleint with IAX?   Jim Flagg
• [Asterisk-Users] Can audio streams go client to cleint with IAX?   Jim Flagg
• [Asterisk-Users] Voiceglo questions, IAX   Jim Flagg
• [Asterisk-Users] CVS Changes (NAT-SIP)   Jim Flagg
• [Asterisk-Users] SIP and DTA-310 was [ MGCP w/8x8 DTA-310 and as5300 pstn gateway]   Jim Flagg
• [Asterisk-Users] SIP and DTA-310 was [ MGCP w/8x8 DTA-310 andas5300 pstn gateway]   Jim Flagg
• [Asterisk-Users] Marketing collateral, etc.   Jim Flagg
• [Asterisk-Users] Room Monitor   Jim Flagg
• [Asterisk-Users] X100P Cards have gone belly up?   Ryan R. Fligg
• [Asterisk-Users] Voicemail Password Digit Timeout   Ryan R. Fligg
• [Asterisk-Users] Dialogic D300SC-E1   Alessio Focardi
• [Asterisk-Users] Dialogic D300SC-E1   Alessio Focardi
• [Asterisk-Users] Make outbound calls only from certain hosts   Alessio Focardi
• [Asterisk-Users] Specify address with base=0xNNNNN   Alessio Focardi
• [Asterisk-Users] extravagant behavior, nat problem ?   Alessio Focardi
• [Asterisk-Users] extravagant behavior, nat problem ?   Alessio Focardi
• [Asterisk-Users] Executing external script   Alessio Focardi
• [Asterisk-Users] Executing external script   Alessio Focardi
• [Asterisk-Users] Incoming context based on ISDN MSN   Alessio Focardi
• [Asterisk-Users] cisco 7912 voicemail/dnd issue   Lele Forzani
• [Asterisk-Users] cisco 7912 voicemail/dnd issue   Lele Forzani
• [Asterisk-Users] Calls dropping off   Steve Foy
• [Asterisk-Users] Calls dropping off   Steve Foy
• [Asterisk-Users] Calls dropping off   Steve Foy
• [Asterisk-Users] Calls dropping off   Steve Foy
• [Asterisk-Users] SIP debug logs   Steve Foy
• [Asterisk-Users] Asterisk 0.7.2   Steve Foy
• [Asterisk-Users] Calls dropping off   Steve Foy
• [Asterisk-Users] Calls dropping off   Steve Foy
• [Asterisk-Users] Calls dropping off   Steve Foy
• [Asterisk-Users] Calls dropping off   Steve Foy
• [Asterisk-Users] Calls dropping off   Steve Foy
• [Asterisk-Users] Asterisk Venture   Steve Foy
• [Asterisk-Users] Caller-ID is being sent wrong. How to fix it?   John Fraizer
• [Asterisk-Users] Caller-ID is being sent wrong. How to fix it?   John Fraizer
• [Asterisk-Users] Caller-ID is being sent wrong. How to fix it?   John Fraizer
• [Asterisk-Users] Caller-ID is being sent wrong. How to fix it?   John Fraizer
• [Asterisk-Users] Caller-ID is being sent wrong. How to fix it?   John Fraizer
• [Asterisk-Users] Caller-ID is being sent wrong. How to fix it?   John Fraizer
• [Asterisk-Users] Caller-ID is being sent wrong. How to fix it?   John Fraizer
• [Asterisk-Users] Caller-ID is being sent wrong. How to fix it?   John Fraizer
• [Asterisk-Users] SIP phones with dual ethernet.   John Fraizer
• [Asterisk-Users] Best OS for Asterisk   John Fraizer
• [Asterisk-Users] Cisco 7960 - how to enable "messages" key   John Fraizer
• [Asterisk-Users] Constant crashes with Asterisk 0.7.2   John Fraizer
• [Asterisk-Users] Error messages I don't understand.   John Fraizer
• [Asterisk-Users] Direct mailbox transfer   John Fraizer
• [Asterisk-Users] Constant crashes with Asterisk 0.7.2   John Fraizer
• [Asterisk-Users] Digium connectivity issue?   John Fraizer
• [Asterisk-Users] Asterisk and "dial by email?"   John Fraizer
• [Asterisk-Users] Digium connectivity issue?   John Fraizer
• [Asterisk-Users] Digium connectivity issue?   John Fraizer
• [Asterisk-Users] HELP!!!! Having problems Starting Asterisk   John Fraizer
• [Asterisk-Users] VOIP Carrier recommendations?   John Fraizer
• [Asterisk-Users] Got my DID, getting an error.   John Fraizer
• [Asterisk-Users] SIP REGISTER   John Fraizer
• [Asterisk-Users] Buzzing on Grandstream phones   John Fraizer
• [Asterisk-Users] SIP REGISTER   John Fraizer
• [Asterisk-Users] "multicasting" conference calls   John Fraizer
• [Asterisk-Users] X100P and DTMF sending   John Fraizer
• [Asterisk-Users] RE: "multicasting" conference calls   John Fraizer
• [Asterisk-Users] SIP Codec selection order   John Fraizer
• [Asterisk-Users] Nested include statements in extensions.conf?   John Fraizer
• [Asterisk-Users] Nested include statements in extensions.conf?   John Fraizer
• [Asterisk-Users] Nested include statements in extensions.conf?   John Fraizer
• [Asterisk-Users] Anyone working with NUFONE?   John Fraizer
• [Asterisk-Users] Could voice mail problem be related to RAM?   John Fraizer
• [Asterisk-Users] calls dropped with grandstream   John Fraizer
• [Asterisk-Users] Cisco 7940/60 Intercom   John Fraizer
• [Asterisk-Users] Call priority   John Fraizer
• [Asterisk-Users] Simulating the "lighted line in use" type of phone   John Fraizer
• [Asterisk-Users] Conference and transfer   John Fraizer
• [Asterisk-Users] E911 support   John Fraizer
• [Asterisk-Users] RE: Message waiting light not coming on   John Fraizer
• [Asterisk-Users] Iconnect behind NAT   John Fraizer
• [Asterisk-Users] sip:user@domain.tld   John Fraizer
• [Asterisk-Users] Iconnect behind NAT   John Fraizer
• [Asterisk-Users] Iconnect behind NAT   John Fraizer
• [Asterisk-Users] Hotel wake-up   John Fraizer
• [Asterisk-Users] OT Superbowl = Linux Shake up to the world..   Brent Franks
• [Asterisk-Users] Calls dropping off   Brent Franks
• [Asterisk-Users] Adtran 750 Configuration   Brent Franks
• [Asterisk-Users] dialout redunancy.   Brent Franks
• [Asterisk-Users] Error Logging (stops Randomly)   Brent Franks
• [Asterisk-Users] Digium connectivity issue?   Brent Franks
• [Asterisk-Users] x100p dropping incoming calls   Brent Franks
• [Asterisk-Users] Re: X100P / Echo / ZTMONITOR CAN2,3, etc.   Brent Franks
• [Asterisk-Users] max asterisk load   Brent Franks
• [Asterisk-Users] OT: Cisco 79XX operation   Brent Franks
• [Asterisk-Users] Attended Transfer Question   Brent Franks
• [Asterisk-Users] Unable to create channem of type 'Zap'   Brent Franks
• [Asterisk-Users] Unable to create channem of type 'Zap'   Brent Franks
• [Asterisk-Users] Unable to create channem of type 'Zap'   Brent Franks
• [Asterisk-Users] Unable to create channel of type 'Zap'   Brent Franks
• [Asterisk-Users] Comments on Voice Quality IP Hard Phones   Brent Franks
• [Asterisk-Users] Adding another X100P Card   Steven E. Frazier
• [Asterisk-Users] Answer from a specific Number   Steven E. Frazier
• [Asterisk-Users] Adding another X100P after X100P and TDM400P is already configured   Steven E. Frazier
• [Asterisk-Users] Festival: read text from external fil   Lars Fredriksson
• [Asterisk-Users] Incoming SIP-calls and Festival   Lars Fredriksson
• [Asterisk-Users] Problems to register with SIP provider   Lars Fredriksson
• [Asterisk-Users] VM: Multilanguage and digits   Lars Fredriksson
• [Asterisk-Users] VM: Multilanguage and digits   Lars Fredriksson
• [Asterisk-Users] Asterisk and Wildcard T100P   Mike Fryer
• [Asterisk-Users] Dialing 800 numbers with VOIP   Rob Fugina
• [Asterisk-Users] Voicemail Password Digit Timeout   Rob Fugina
• [Asterisk-Users] Round-robin chan_zap groups...   Rob Fugina
• [Asterisk-Users] Shopping for ISDN card   Rob Fugina
• [Asterisk-Users] Simple Front Gate Intercom   Rob Fugina
• [Asterisk-Users] Simple Front Gate Intercom   Rob Fugina
• [Asterisk-Users] Simple Front Gate Intercom   Rob Fugina
• [Asterisk-Users] New to T-1/Channel Bank hardware -- help?   Rob Fugina
• [Asterisk-Users] Hotel wake-up   Rob Fugina
• [Asterisk-Users] Hotel wake-up   Rob Fugina
• [Asterisk-Users] Hotel wake-up   Rob Fugina
• [Asterisk-Users] Hotel wake-up   Rob Fugina
• [Asterisk-Users] Hotel wake-up   Rob Fugina
• [Asterisk-Users] Hotel wake-up   Rob Fugina
• [Asterisk-Users] Sip transfers   Sean Garland
• [Asterisk-Users] Direct mailbox transfer   Sean Garland
• [Asterisk-Users] Direct mailbox transfer   Sean Garland
• [Asterisk-Users] Telemarketer handling   Sean Garland
• [Asterisk-Users] extension mobility   Gary
• [Asterisk-Users] Motherboard and fxo suggestion.   Geo_p15tt
• [Asterisk-Users] Incoming context based on ISDN MSN   Jean-Denis Girard
• [Asterisk-Users] Anyone working with NUFONE?   Jean-Denis Girard
• [Asterisk-Users] Incoming context based on ISDN MSN   Jean-Denis Girard
• [Asterisk-Users] Message waiting light not coming on   Jean-Denis Girard
• [Asterisk-Users] Digium connectivity issue?   James Golovich
• [Asterisk-Users] Asterisk monitor with Daemontools   James Golovich
• [Asterisk-Users] agi scripting in perl - dealiing withunexpected disconnects gracefully / spurious DTMF   James Golovich
• [Asterisk-Users] Room Monitor   James Golovich
• [Asterisk-Users] * dropps outbound calls over PSTN   James Golovich
• [Asterisk-Users] Dialing 800 numbers with VOIP   James Golovich
• [Asterisk-Users] "Call did not go through"   James Golovich
• [Asterisk-Users] Asterisk Faxing   James Golovich
• [Asterisk-Users] Asterisk Faxing   James Golovich
• [Asterisk-Users] RE: "multicasting" conference calls   James Golovich
• [Asterisk-Users] A missing argument   James Golovich
• [Asterisk-Users] Re: Outgoing parallelism   James Golovich
• [Asterisk-Users] Brazilian Protocol   Marcio Gomes
• [Asterisk-Users] R2lib   Marcio Gomes
• [Asterisk-Users] Details on TE410P Digium cards   David Gomillion
• [Asterisk-Users] dialing delay question.   David Gomillion
• [Asterisk-Users] Dialling Hook Flash on Zaptel   David Gomillion
• [Asterisk-Users] Nortel and Asterisk interconnection   David Gomillion
• [Asterisk-Users] The Smallest Asterisk Server Ever?   David Gomillion
• [Asterisk-Users] Dialling Hook Flash on Zaptel   David Gomillion
• [Asterisk-Users] Hangup   Eduardo Goncalves
• [Asterisk-Users] Mailbox full ?   Girish Gopinath
• [Asterisk-Users] softphone configs?   Girish Gopinath
• [Asterisk-Users] softphone configs?   Girish Gopinath
• [Asterisk-Users] cdr->dst incorrect?   Girish Gopinath
• [Asterisk-Users] RE: sip router gateway   Girish Gopinath
• [Asterisk-Users] DTMF Issues with SJPHONE   Girish Gopinath
• [Asterisk-Users] DTMF Issues with SJPHONE   Girish Gopinath
• [Asterisk-Users] OT Superbowl = Linux Shake up to the world..   Adam Goryachev
• [Asterisk-Users] Code Hosting...   Adam Goryachev
• [Asterisk-Users] Get new PRI working   Adam Goryachev
• [Asterisk-Users] Get new PRI working   Adam Goryachev
• [Asterisk-Users] Get new PRI working   Adam Goryachev
• [Asterisk-Users] Cisco 7960 - how to enable "messages" key   Michael Graff
• [Asterisk-Users] Luxoncomm 3800 series FXO/FXS adapters?   Michael Graves
• [Asterisk-Users] Fax Extension   Michael Graves
• [Asterisk-Users] Telian small FXS/FXO adapters   Michael Graves
• [Asterisk-Users] Asterisk and Vonage -- possible? Other DID providers with Atlanta presence?   Michael Graves
• [Asterisk-Users] Pingtel Phones?   Michael Graves
• [Asterisk-Users] Asterisk and Vonage -- no make that VoicePulse Connect :)   Michael Graves
• [Asterisk-Users] pstn 800#'s   Michael Graves
• [Asterisk-Users] codec negotiation   Michael Graves
• [Asterisk-Users] Make a phone dial remotely?   Michael Graves
• [Asterisk-Users] Configuring Pingtel Xpressa   Michael Graves
• [Asterisk-Users] IAX Connection - Voicepulse   Michael Graves
• [Asterisk-Users] Are IAX2 providers ready for prime time?   Michael Graves
• [Asterisk-Users] Is IAXtel down?   Michael Graves
• [Asterisk-Users] Web based UA   Michael Graves
• [Asterisk-Users] PCphoneline FXO to FXS box??   Michael Graves
• [Asterisk-Users] moh badness: mpg123 0.59s   Dorian Gray
• [Asterisk-Users] Force SIP Phones to Register   Tom Green
• [Asterisk-Users] New Zealand users/contractors   Gwilym Griffith-Jones
• [Asterisk-Users] Re: ISDN update   Cees de Groot
• [Asterisk-Users] DIAX 0.9.6b call reception   Cees de Groot
• [Asterisk-Users] Re: DIAX 0.9.6b call reception   Cees de Groot
• [Asterisk-Users] Re: Need to interface to BRIs   Cees de Groot
• [Asterisk-Users] Re: Need to interface to BRIs   Cees de Groot
• [Asterisk-Users] Echo on Diva Server 2M with melware CAPI   Cees de Groot
• [Asterisk-Users] help a poor newbie out with SIP choppy one-way problem   Nicolas Gudino
• [Asterisk-Users] dtmf recording record and playback   Nicolas Gudino
• [Asterisk-Users] agi scripting in perl - dealiing withunexpected disconnects gracefully / spurious DTMF   Nicolas Gudino
• [Asterisk-Users] Unable to create channem of type 'Zap'   Nicolas Gudino
• [Asterisk-Users] Meetme without zaptel hardware   CW_ASN - Gus
• [Asterisk-Users] question for oh323 users   CW_ASN - Gus
• [Asterisk-Users] Problems with ATA's locking up..   CW_ASN - Gus
• [Asterisk-Users] Re: question for oh323 users   CW_ASN - Gus
• [Asterisk-Users] Weird sdp output   CW_ASN - Gus
• [Asterisk-Users] progress on DTMF   Jeff Gustafson
• [Asterisk-Users] progress on DTMF   Jeff Gustafson
• [Asterisk-Users] x101p beeps/sceeching   Jeff Gustafson
• [Asterisk-Users] Solved! x101p beeps/sceeching   Jeff Gustafson
• [Asterisk-Users] Wanpipe cards?   Jeff Gustafson
• [Asterisk-Users] ztmonitor and the x101p   Jeff Gustafson
• [Asterisk-Users] ztmonitor and the x101p   Jeff Gustafson
• [Asterisk-Users] RE: two UA with the same usr/pwd   Ed Guy
• [Asterisk-Users] RE: two UA with the same usr/pwd   Ed Guy
• [Asterisk-Users] How to compile bri-stuff.0.0.2.rc12   M H
• [Asterisk-Users] Wifi Phones   HQ
• [Asterisk-Users] Do you Linux softphone..   Walker Haddock
• AW: [Asterisk-Users] Data call transfer   Thomas Haeger
• AW: [Asterisk-Users] Need to interface to BRIs   Thomas Haeger
• [Asterisk-Users] nasty segfault with previous strange Zaptel warnings   Thomas Haeger
• [Asterisk-Users] callerid will not be set   Thomas Haeger
• [Asterisk-Users] how to dial and accept a call with only x100p card on Redhat linux 9.0?   Mark Hagler
• [Asterisk-Users] Loading module chan_capi.so failed!   Bodo Hahnke
• [Asterisk-Users] Loading module chan_capi.so failed! still some problems ...   Bodo Hahnke
• [Asterisk-Users] Asterisk hangs up when a call comes in   Bodo Hahnke
• [Asterisk-Users] Anybody managed to call a phone through sipgate.de   David Hajek
• [Asterisk-Users] Calls dropping off   Bill Hamel
• [Asterisk-Users] Is there a MaxQueueTime for Queues ?   Bill Hamel
• [Asterisk-Users] Agent / Queue help   Bill Hamel
• [Asterisk-Users] Problem playing the first voice mail prompt   Bill Hamlin
• [Asterisk-Users] help a poor newbie out with SIP choppy one-way problem   Freddi Hansen
• [Asterisk-Users] HFC-Cologne TE mode, non-I4L mode.   Jens P. Hansen
• [Asterisk-Users] HANGUPCAUSE   Tais M. Hansen
• [Asterisk-Users] Transfer   Tais M. Hansen
• [Asterisk-Users] Transfer   Tais M. Hansen
• [Asterisk-Users] Transfer   Tais M. Hansen
• [Asterisk-Users] The Smallest Asterisk Server Ever?   Matthew Hardeman
• [Asterisk-Users] can * support ANI   Doug Harris
• [Asterisk-Users] Can audio streams go client to cleint with IAX?   Adam Hart
• [Asterisk-Users] Configuring Firefly Network in *   Adam Hart
• [Asterisk-Users] Can audio streams go client to cleint with IAX?   Adam Hart
• [Asterisk-Users] Can audio streams go client to cleint with IAX?   Adam Hart
• [Asterisk-Users] Can audio streams go client to cleint with IAX?   Adam Hart
• [Asterisk-Users] Can audio streams go client to cleint with IAX?   Adam Hart
• [Asterisk-Users] Can audio streams go client to cleint with IAX?   Adam Hart
• [Asterisk-Users] CALEA?   Adam Hart
• [Asterisk-Users] IAX Softphone Errors   Adam Hart
• [Asterisk-Users] Firefly 1.4 released   Adam Hart
• [Asterisk-Users] How much processing power is needed?   Adam Hart
• [Asterisk-Users] Minor update of Firefly   Adam Hart
• [Asterisk-Users] Patching Asterisk for OpenH323 ASN.1 Vulnerabilities   Adam Hart
• [Asterisk-Users] Patching Asterisk for OpenH323 ASN.1 Vulnerabilities   Adam Hart
• [Asterisk-Users] Patching Asterisk for OpenH323 ASN.1 Vulnerabilities   Adam Hart
• [Asterisk-Users] Grandstream -> firefly call translator problem   Adam Hart
• [Asterisk-Users] Grandstream -> firefly call translator problem   Adam Hart
• [Asterisk-Users] RE: Apple OS-X   Charles Hatchette
• [Asterisk-Users] SIP gateway question   Christian Hecimovic
• [Asterisk-Users] Mediatrix 1204 sip g/w now working   Christian Hecimovic
• [Asterisk-Users] Telemarketer handling   Christian Hecimovic
• [Asterisk-Users] 4 port FXO   Christian Hecimovic
• [Asterisk-Users] Cisco 7940 SIP Registrations   Andreas Hein
• [Asterisk-Users] New Search engine for the list - Final resting place   Kim Hendrikse
• [Asterisk-Users] Re: New Search engine for the list - Whoops!   Kim Hendrikse
• [Asterisk-Users] Mailing list search engine   Kim Hendrikse
• [Asterisk-Users] NEC IP phone compatibility?   Jay Hennigan
• [Asterisk-Users] New Zealand   Carlos Hernandez
• [Asterisk-Users] Zaptel BRI and HFC-S cards in NT-Mode   Carlos Hernandez
• [Asterisk-Users] Pingtel Phones?   Andy Hester
• [Asterisk-Users] Configuring Pingtel Xpressa   Andy Hester
• [Asterisk-Users] Delta Three/iConnectHere Outgoing Caller ID?   Chris Higgins
• [Asterisk-Users] Dial via sip gateway?   Greg Hill
• [Asterisk-Users] Dial via sip gateway?   Greg Hill
• [Asterisk-Users] Dial via sip gateway?   Greg Hill
• [Asterisk-Users] setting up ---- newbie   Greg Hill
• [Asterisk-Users] diax softphone   Greg Hill
• [Asterisk-Users] Boards falling out...   Greg Hill
• [Asterisk-Users] Voiceglo questions   Greg Hill
• [Asterisk-Users] RE:voiceglo sip config   Greg Hill
• [Asterisk-Users] RE:voiceglo sip config   Greg Hill
• [Asterisk-Users] Room Monitor   Greg Hill
• [Asterisk-Users] Woodpeckers   Greg Hill
• [Asterisk-Users] SIP debug logs   Ejay Hire
• [Asterisk-Users] voip phones   Ejay Hire
• [Asterisk-Users] Boards falling out...   Ejay Hire
• [Asterisk-Users] Calls dropping off   Ejay Hire
• [Asterisk-Users] ATA 186 Registration!!!!   Ejay Hire
• [Asterisk-Users] E911 support   Ejay Hire
• [Asterisk-Users] Cisco 7960 SIP image (off-topic)   Chris Hirsch
• [Asterisk-Users] Cisco 7960 SIP image (off-topic)   Chris Hirsch
• [Asterisk-Users] Integrating with an existing PBX   Barton Hodges
• [Asterisk-Users] Call Queues   Jonathan Stanton at Home
• [Asterisk-Users] Hide outgoing CallerId on Zap interface   Jonathan Stanton at Home
• [Asterisk-Users] Asterisk as proxy?   Tor Houghton
• [Asterisk-Users] voip phones   Billy Huddleston
• [Asterisk-Users] Problems with ATA's locking up..   Billy Huddleston
• [Asterisk-Users] Problems with ATA's locking up..   Billy Huddleston
• [Asterisk-Users] Problems with ATA's locking up..   Billy Huddleston
• [Asterisk-Users] Problems with ATA's locking up..   Billy Huddleston
• [Asterisk-Users] Re-Invites and Studder.   Billy Huddleston
• [Asterisk-Users] Apple OS-X   Martin Hunt
• [Asterisk-Users] * dropps outbound calls over PSTN   Ralf Illing
• [Asterisk-Users] * dropps outbound calls over PSTN   Ralf Illing
• [Asterisk-Users] Cisco 7960 quick dial   Jose Inzunza/YM/RWDOE
• [Asterisk-Users] Details on TE410P Digium cards   Dan Iordanescu
• [Asterisk-Users] TE410P PRI T1 problem   Dan Iordanescu
• [Asterisk-Users] Termination - Cuba   Linux Dominicana - J.M.F.A
• [Asterisk-Users] sipphone dialing out problem   Deepakumar JV
• [Asterisk-Users] talking clock   Deepakumar JV
• [Asterisk-Users] talking clock   Deepakumar JV
• [Asterisk-Users] talking clock   Deepakumar JV
• [Asterisk-Users] talking clock   Deepakumar JV
• [Asterisk-Users] H323 calls via provider   Deepakumar JV
• [Asterisk-Users] DTMF over SIP to a Cisco gateway   Deepakumar JV
• [Asterisk-Users] incoming DTMF on a SIP call   Deepakumar JV
• [Asterisk-Users] TDM400 showing up as Tiger Jet   Deepakumar JV
• [Asterisk-Users] X100P / Echo / ZTMONITOR CAN2,3, etc.   Deepakumar JV
• [Asterisk-Users] Digium connectivity issue?   Deepakumar JV
• [Asterisk-Users] voicemail extension - hangup   Deepakumar JV
• Fw: [Asterisk-Users] voicemail extension - hangup   Deepakumar JV
• [Asterisk-Users] oh323 codec negotiation   Deepakumar JV
• [Asterisk-Users] Asterisk monitor with Daemontools   Jacky
• [Asterisk-Users] Termination - Cuba   Amaury Jacquot
• [Asterisk-Users] Brazilian Protocol   Amaury Jacquot
• [Asterisk-Users] Record communication   Mike Jagdis
• [Asterisk-Users] An example config for using a Wildcard X100P and a SIP phone?   Jason
• [Asterisk-Users] An example config for using a Wildcard X100P and a SIP phone?   Jason
• [Asterisk-Users] Setting up with an Eicon DIVA PCI card?   Jason
• [Asterisk-Users] SIP newbie question:SIP proxy is necessary?   Jeff
• [Asterisk-Users] Asterisk and Vonage -- no make that VoicePulse Connect :)   Jeff
• [Asterisk-Users] two UA with the same usr/pwd   Jeff
• [Asterisk-Users] RE: two UA with the same usr/pwd   Jeff
• [Asterisk-Users] Language setting   Jeroen
• [Asterisk-Users] [DENICenum-l] Open Workshop on IP voice and associated convergent services]   Rainer Jochem
• [Asterisk-Users] SUBSCRIBE in chan_sip - anyone?   Olle E. Johansson
• [Asterisk-Users] SUBSCRIBE in chan_sip - anyone?   Olle E. Johansson
• [Asterisk-Users] SUBSCRIBE in chan_sip - anyone?   Olle E. Johansson
• [Asterisk-Users] Dial via sip gateway?   Olle E. Johansson
• [Asterisk-Users] SIP debug logs   Olle E. Johansson
• [Asterisk-Users] Caller-ID is being sent wrong. How to fix it?   Olle E. Johansson
• [Asterisk-Users] Caller-ID is being sent wrong. How to fix it?   Olle E. Johansson
• [Asterisk-Users] Snom 200 MWI Button   Olle E. Johansson
• [Asterisk-Users] Calling SIP   Olle E. Johansson
• [Asterisk-Users] Calling SIP   Olle E. Johansson
• [Asterisk-Users] Anybody going to the Spring VON converence [OT]   Olle E. Johansson
• [Asterisk-Users] two UA with the same usr/pwd   Olle E. Johansson
• [Asterisk-Users] SIP config documentation   Olle E. Johansson
• [Asterisk-Users] SIP config documentation   Olle E. Johansson
• [Asterisk-Users] Re: incominglimits and outgoinglimits (New subject)   Olle E. Johansson
• [Asterisk-Users] Re: [Asterisk-Users] SIP config documentation   Olle E. Johansson
• [Asterisk-Users] SIP extension "busy" when not available ??   Olle E. Johansson
• [Asterisk-Users] How to best debug SIP registration failure   Olle E. Johansson
• [Asterisk-Users] Re: How to best debug SIP registration failure   Olle E. Johansson
• [Asterisk-Users] VM: Multilanguage and digits   Olle E. Johansson
• [Asterisk-Users] Changes in capi.conf   Olle E. Johansson
• [Asterisk-Users] Re: calls dropped with grandstream   Olle E. Johansson
• [Asterisk-Users] Problem with SIP 407   Olle E. Johansson
• [Asterisk-Users] Problem with SIP 407   Olle E. Johansson
• [Asterisk-Users] Message waiting light not coming on   Olle E. Johansson
• [Asterisk-Users] chan_sip support for SIP:Remote-Party-ID, specifically CLID priva cy   Olle E. Johansson
• [Asterisk-Users] Grandstream transfer into outer space   Olle E. Johansson
• [Asterisk-Users] Re: Grandstream transfer into outer space   Olle E. Johansson
• [Asterisk-Users] chan_sip support for SIP:Remote-Party-ID, sp ecifically CLID priva cy   Olle E. Johansson
• [Asterisk-Users] retrieve_sip_conf_from_mysql.pl data format   Olle E. Johansson
• [Asterisk-Users] retrieve_sip_conf_from_mysql.pl data format   Olle E. Johansson
• [Asterisk-Users] Smallest server continued...   Brian Johnson
• [Asterisk-Users] simple test setup   Brian Johnson
• [Asterisk-Users] cannot find -lXext when building * ?   C. Johnson
• [Asterisk-Users] Sending DTMF out-of-band over IAX2   Mark Johnston
• [Asterisk-Users] Skinny and SIP   Asterisk Jones
• [Asterisk-Users] Cisco 7940/60 Intercom   Asterisk Jones
• [Asterisk-Users] SUBSCRIBE in chan_sip - anyone?   Clif Jones
• [Asterisk-Users] Compiling while * is running   Clif Jones
• [Asterisk-Users] Still looking for small fxo sip gateway   Clif Jones
• [Asterisk-Users] Mediatrix sip fxo gateway workaround?   Clif Jones
• Fw: [Asterisk-Users] Possible Sip logic bug?   Clif Jones
• [Asterisk-Users] Cisco 7960 bug in 6.1 evident in Asterisk   Clif Jones
• [Asterisk-Users] Cisco 7960 bug in 6.1 evident in Asterisk   Clif Jones
• Fw: [Asterisk-Users] Possible Sip logic bug?   Clif Jones
• [Asterisk-Users] Having problems with RTP packets and Hold   Clif Jones
• [Fwd: [Asterisk-Users] Having problems with RTP packets and Hold]   Clif Jones
• [Fwd: [Asterisk-Users] Having problems with RTP packets and H old]   Clif Jones
• [Asterisk-Users] Can't connect KPhone to asterisk   Clif Jones
• [Asterisk-Users] Pingtel Phones?   Clif Jones
• [Asterisk-Users] FXO gateways on Asterisk   Clif Jones
• [Serusers] Re: [Asterisk-Users] INFO/DTMF retransmissions in * not absorbed?   Clif Jones
• [Asterisk-Users] RFC 2833 / Timestamp   Clif Jones
• [Asterisk-Users] Can audio streams go client to cleint with IAX?   Jeremy Jones
• [Asterisk-Users] The Smallest Asterisk Server Ever?   Jeremy Jones
• [Asterisk-Users] DTA-310 Outbound Dialing   Jeremy Jones
• [Asterisk-Users] DTA-310 Outbound Dialing   Jeremy Jones
• [Asterisk-Users] Transfer   Senad Jordanovic
• [Asterisk-Users] Calls dropping off   Senad Jordanovic
• [Asterisk-Users] Calls dropping off   Senad Jordanovic
• [Asterisk-Users] Calls dropping off   Senad Jordanovic
• [Asterisk-Users] Transfer   Senad Jordanovic
• [Asterisk-Users] Qualify statement   Senad Jordanovic
• [Asterisk-Users] Port bind   Senad Jordanovic
• [Asterisk-Users] Calls dropping off   Senad Jordanovic
• [Asterisk-Users] Asterisk under UML?   Senad Jordanovic
• [Asterisk-Users] how to password protect a meetme conference?   Senad Jordanovic
• [Asterisk-Users] Transfer   Senad Jordanovic
• [Asterisk-Users] SPA 2000 ringing   Senad Jordanovic
• [Asterisk-Users] SPA 2000 ringing   Senad Jordanovic
• [Asterisk-Users] Web based UA   Bartosz Jozwiak
• [Asterisk-Users] Kansas SIP or IAX Provider?   Arnold Cavazos Jr.
• [Asterisk-Users] SIP debug logs   James H. Cloos Jr.
• [Asterisk-Users] Re: sementation fault with mpg123   James H. Cloos Jr.
• [Asterisk-Users] sipphone dialing out problem   James H. Cloos Jr.
• [Asterisk-Users] Re: Execute command in shell   James H. Cloos Jr.
• [Asterisk-Users] Re: Execute command in shell   James H. Cloos Jr.
• [Asterisk-Users] Re: Asterisk under UML?   James H. Cloos Jr.
• [Asterisk-Users] Fast question on extension matching   James H. Cloos Jr.
• [Asterisk-Users] Re: Speex == Screech using version 1.1.4   James H. Cloos Jr.
• [Asterisk-Users] 1.freenum.org. [was: Re: Dialing 800 numbers with VOIP]   James H. Cloos Jr.
• [Asterisk-Users] Re: Jump to extension from voice menu   James H. Cloos Jr.
• [Asterisk-Users] Re: Residential Plans for Asterisk Users   James H. Cloos Jr.
• [Asterisk-Users] Re: High Density configuration for Voice & Fax   James H. Cloos Jr.
• [Asterisk-Users] Re: High Density configuration for Voice & Fax   James H. Cloos Jr.
• [Asterisk-Users] Re: Hide outgoing CallerId on Zap interface   James H. Cloos Jr.
• [Asterisk-Users] Good source for moh files   James H. Cloos Jr.
• [Asterisk-Users] Re: Double digits seen using Grandstream phones   James H. Cloos Jr.
• [Asterisk-Users] IAXTEL and the registration traffic   James H. Cloos Jr.
• [Asterisk-Users] Re: pstn 800#'s   James H. Cloos Jr.
• [Asterisk-Users] Re: Pingtel Opensource PBX Announcement   James H. Cloos Jr.
• [Asterisk-Users] Re: exit   James H. Cloos Jr.
• [Asterisk-Users] ISDN update   Klaus-Peter Junghanns
• [Asterisk-Users] ISDN update   Klaus-Peter Junghanns
• [Asterisk-Users] Fax   Klaus-Peter Junghanns
• [Asterisk-Users] Need to interface to BRIs   Klaus-Peter Junghanns
• [Asterisk-Users] Need to interface to BRIs   Klaus-Peter Junghanns
• [Asterisk-Users] Re: Need to interface to BRIs   Klaus-Peter Junghanns
• [Asterisk-Users] Analogical FXO vs. BRI dialing speed   Klaus-Peter Junghanns
• [Asterisk-Users] Re: Need to interface to BRIs   Klaus-Peter Junghanns
• [Asterisk-Users] chan_capi problem   Klaus-Peter Junghanns
• [Asterisk-Users] Zaptel BRI and HFC-S cards in NT-Mode   Klaus-Peter Junghanns
• [Asterisk-Users] Zaptel BRI and HFC-S cards in NT-Mode   Klaus-Peter Junghanns
• [Asterisk-Users] Zaptel BRI and HFC-S cards in NT-Mode   Klaus-Peter Junghanns
• [Asterisk-Users] Changes in capi.conf   Klaus-Peter Junghanns
• [Asterisk-Users] cdr_addon_mysql problem linking   Gregory Junker
• [Asterisk-Users] modprobe wcfxs   How Peng Kaiam
• [Asterisk-Users] Can't connect KPhone to asterisk   Maciek Kaminski
• [Asterisk-Users] Can't connect KPhone to asterisk   Maciek Kaminski
• [Asterisk-Users] Can't connect KPhone to asterisk   Maciek Kaminski
• [Asterisk-Users] festival in agi?   Maciek Kaminski
• [Asterisk-Users] s/asterisk mailinglists/asterisk forum/g ?   Roy Sigurd Karlsbakk
• [Asterisk-Users] Asterisk Venture   Roy Sigurd Karlsbakk
• [Asterisk-Users] sip router gateway   Anand S. Katti
• [Asterisk-Users] RE: sip router gateway   Anand S. Katti
• [Asterisk-Users] The Smallest Asterisk Server Ever?   Tony Kava
• [Asterisk-Users] Dell 1750 server and Asteriks...   Tony Kava
• [Asterisk-Users] Dell 1750 server and Asteriks...   Tony Kava
• [Asterisk-Users] Unable to create vpb channel   Steven Kawuma
• [Asterisk-Users] Unable to create vpb channel   Steven Kawuma
• [Asterisk-Users] Boards falling out...   Greg Kedrovsky
• [Asterisk-Users] Boards falling out...   Greg Kedrovsky
• [Asterisk-Users] Boards falling out...   Greg Kedrovsky
• [Asterisk-Users] Re: Boards falling out...   Greg Kedrovsky
• [Asterisk-Users] Simple Front Gate Intercom   Greg Kedrovsky
• [Asterisk-Users] Simple Front Gate Intercom   Greg Kedrovsky
• [Asterisk-Users] Simple Front Gate Intercom   Greg Kedrovsky
• [Asterisk-Users] Simple Front Gate Intercom   Greg Kedrovsky
• [Asterisk-Users] Simple Front Gate Intercom   Greg Kedrovsky
• [Asterisk-Users] Simple Front Gate Intercom   Greg Kedrovsky
• [Asterisk-Users] Simple Front Gate Intercom   Greg Kedrovsky
• [Asterisk-Users] exit   Greg Kedrovsky
• [Asterisk-Users] exit   Greg Kedrovsky
• [Asterisk-Users] exit   Greg Kedrovsky
• [Asterisk-Users] exit   Greg Kedrovsky
• [Asterisk-Users] CALEA?   Steve Kennedy
• [Asterisk-Users] Best OS for Asterisk   Steve Kennedy
• [Asterisk-Users] Voip in the EU   Steve Kennedy
• [Asterisk-Users] Voip in the EU   Steve Kennedy
• [Asterisk-Users] Voip in the EU   Steve Kennedy
• [Asterisk-Users] Voip in the EU   Steve Kennedy
• [Asterisk-Users] Voip in the EU   Steve Kennedy
• [Asterisk-Users] Woodpeckers   Steve Kennedy
• [Asterisk-Users] Caller ID with Name X100   Kevin
• [Asterisk-Users] VOIP Deployment Concerns --- Adelphia Cable   Lawrence Keyes
• [Asterisk-Users] Flash to PBX with X100P - how to?   Maciej Kietlinski
• [Asterisk-Users] Whats wrong with dialplan?   Bob Klepfer
• [Asterisk-Users] Switch brands, speeds, etc.   Bob Klepfer
• [Asterisk-Users] IAX call problems   Philipp von Klitzing
• [Asterisk-Users] Calls dropping off   Philipp von Klitzing
• [Asterisk-Users] Calls dropping off   Philipp von Klitzing
• [Asterisk-Users] New Firmware for Grandstream Phones - Supports CFG by MAC address   Philipp von Klitzing
• [Asterisk-Users] asterisk and fax over ip - concept   Philipp von Klitzing
• [Asterisk-Users] Calling SIP   Philipp von Klitzing
• [Asterisk-Users] Log entry   Philipp von Klitzing
• [Asterisk-Users] setting up callback   Philipp von Klitzing
• [Asterisk-Users] chan_local and variables   Philipp von Klitzing
• [Asterisk-Users] Speech between Grandstream phones sounds like talking under water   Philipp von Klitzing
• [Asterisk-Users] Mailbox full ?   Philipp von Klitzing
• [Asterisk-Users] Re: Need to interface to BRIs   Philipp von Klitzing
• [Asterisk-Users] Record communication   Philipp von Klitzing
• [Asterisk-Users] RE: two UA with the same usr/pwd   Philipp von Klitzing
• [Asterisk-Users] Record communication   Philipp von Klitzing
• [Asterisk-Users] codec negotiation prob solved?   Philipp von Klitzing
• [Asterisk-Users] IAX Connection - Voicepulse   Philipp von Klitzing
• [Asterisk-Users] Using * for large VoIP implementation   Philipp von Klitzing
• [Asterisk-Users] New Wiki page: Dimensioning an Asterisk system   Philipp von Klitzing
• [Asterisk-Users] 2 questions about ISDN BRI   Philipp von Klitzing
• [Asterisk-Users] How to best debug SIP registration failure   Philipp von Klitzing
• [Asterisk-Users] Pickup   Philipp von Klitzing
• [Asterisk-Users] Nested include statements in extensions.conf?   Philipp von Klitzing
• [Asterisk-Users] line status   Philipp von Klitzing
• [Asterisk-Users] Call priority   Philipp von Klitzing
• [Asterisk-Users] Simulating the "lighted line in use" type of phone   Philipp von Klitzing
• [Asterisk-Users] cdr->dst incorrect?   Philipp von Klitzing
• [Asterisk-Users] Detection of extension   Philipp von Klitzing
• [Asterisk-Users] Web based UA   Philipp von Klitzing
• [Asterisk-Users] chan_capi 0.3.1 segfault backtrace   Philipp von Klitzing
• [Asterisk-Users] Conference and transfer   Philipp von Klitzing
• [Asterisk-Users] SIP Extrange Problem   Philipp von Klitzing
• [Asterisk-Users] callerid will not be set   Philipp von Klitzing
• [Asterisk-Users] Big Install examples please   Philipp von Klitzing
• [Asterisk-Users] Big Install examples please   Philipp von Klitzing
• [Asterisk-Users] Does Digium TDM400P + X100P make a switchboard?   Philipp von Klitzing
• [Asterisk-Users] Problem connecting to Asterisk Server   Philipp von Klitzing
• [Asterisk-Users] Anybody managed to call a phone through sipgate.de   Philipp von Klitzing
• [Asterisk-Users] A General question   Philipp von Klitzing
• [Asterisk-Users] help *** newbie   Martin Klozik
• [Asterisk-Users] Asuscom HiSax based ISDN BRI card - one way latency   Martin 'poorman' Klozik
• AW: [Asterisk-Users] Eicon Diva Server   Sascha Knific
• AW: [Asterisk-Users] Loading module chan_capi.so failed!   Sascha Knific
• AW: [Asterisk-Users] Anybody managed to call a phone through sipgate.de   Sascha Knific
• [Asterisk-Users] OT:Linux(or *BSD) SNMP tools (Was: Re: rtp sound quality?)   Bob Knight
• [Asterisk-Users] Mediatrix 1204 SIP FXO 4-port gateway review   Bob Knight
• [Asterisk-Users] Sip flow diagram?   Bob Knight
• [Asterisk-Users] Snom 100 Code Recommendation   Bob Knight
• [Asterisk-Users] s/asterisk mailinglists/asterisk forum/g ?   Bob Knight
• [Asterisk-Users] "Stuck" TE410P cards   Bob Knight
• [Asterisk-Users] Anybody going to the Spring VON converence [ OT]   Bob Knight
• [Asterisk-Users] Anybody going to the Spring VON converence [OT]   Bob Knight
• [Asterisk-Users] Hotel wake-up   Bob Knight
• [Asterisk-Users] Asterisk and Vonage -- possible? Other DID providers with Atlanta presence?   Tom Knox
• [Asterisk-Users] Asterisk and Vonage -- no make that VoicePulse Connect :)   Tom Knox
• [Asterisk-Users] Got my DID, getting an error.   Tom Knox
• [Asterisk-Users] Got my DID, getting an error.   Tom Knox
• [Asterisk-Users] Re: Got my DID, getting an error.   Tom Knox
• [Asterisk-Users] EAGI errors?   Tom Knox
• [Asterisk-Users] Are IAX2 providers ready for prime time?   Tom Knox
• [Asterisk-Users] Using Festival inside an agi   Tom Knox
• [Asterisk-Users] Re: Using Festival inside an agi   Tom Knox
• [Asterisk-Users] Asterisk GUI Client - New verison 0.9   Dustin Knuttgen
• [Asterisk-Users] Snom 200 MWI Button   Dustin Knuttgen
• [Asterisk-Users] Pictures of new multiport FXO/FXS from digum   Gene Kochanowsky
• [Asterisk-Users] 8 lines - best approach   Gene Kochanowsky
• [Asterisk-Users] TDM400 FXO???   Gene Kochanowsky
• [Asterisk-Users] TDM400 FXO???   Gene Kochanowsky
• [Asterisk-Users] voip phones   Michael Koehler
• [Asterisk-Users] voip phones   Michael Koehler
• [Asterisk-Users] voip phones   Michael Koehler
• [Asterisk-Users] Re: asterisk-grandstream call   Michael Koehler
• [Asterisk-Users] Asterisk PABX switch   Nikolay Koev
• [Asterisk-Users] RE:Asterisk PABX switch   Nikolay Koev
• [Asterisk-Users] How do I provide redundancy and reliability w/ Asterisk?   Andrew Kohlsmith
• [Asterisk-Users] How do I provide redundancy and reliability w/ Asterisk?   Andrew Kohlsmith
• [Asterisk-Users] Boards falling out...   Andrew Kohlsmith
• [Asterisk-Users] Re: iax, trunking, etc.   Andrew Kohlsmith
• [Asterisk-Users] Re: Boards falling out...   Andrew Kohlsmith
• [Asterisk-Users] ParkAndAnnounce - Get Parking Extension   Andrew Kohlsmith
• [Asterisk-Users] Answer from a specific Number   Andrew Kohlsmith
• [Asterisk-Users] iax2 jitter stats confusion   Andrew Kohlsmith
• [Asterisk-Users] modprobe wcfxs   Andrew Kohlsmith
• [Asterisk-Users] Re: [Asterisk-Dev] s/asterisk mailinglists/asterisk forum/g ?   Andrew Kohlsmith
• [Asterisk-Users] s/asterisk mailinglists/asterisk forum/g ?   Andrew Kohlsmith
• [Asterisk-Users] X100P / Echo / ZTMONITOR CAN2,3, etc.   Andrew Kohlsmith
• [Asterisk-Users] Jitter Buffer Configuration (typo in iax.conf)   Andrew Kohlsmith
• [Asterisk-Users] RE: Rhino channel bank and aastra PT390 phones   Andrew Kohlsmith
• [Asterisk-Users] TE405P and dual Athlon systems   Andrew Kohlsmith
• [Asterisk-Users] Wifi Phones   Andrew Kohlsmith
• [Asterisk-Users] Wifi Phones   Andrew Kohlsmith
• [Asterisk-Users] Woodpeckers   Andrew Kohlsmith
• [Asterisk-Users] Asterisk Faxing   Andrew Kohlsmith
• [Asterisk-Users] Simulating the "lighted line in use" type of phone   Andrew Kohlsmith
• [Asterisk-Users] Grandstream transfer into outer space   Andrew Kohlsmith
• [Asterisk-Users] Grandstream transfer into outer space   Andrew Kohlsmith
• [Asterisk-Users] Dialing out after caller leaves message   Andrew Kohlsmith
• [Asterisk-Users] Still looking for small fxo sip gateway   Kostur, Andre
• [Asterisk-Users] Vegastream 50 FXO with Asterisk   Kostur, Andre
• [Asterisk-Users] check if que has members   Tjardick van der Kraan
• [Asterisk-Users] INFO/DTMF retransmissions in * not absorbed?   Jiri Kuthan
• [Asterisk-Users] OT: SNOM and TAPI   Jiri Kuthan
• [Asterisk-Users] INFO/DTMF retransmissions in * not absorbed?   Jiri Kuthan
• [Serusers] Re: [Asterisk-Users] INFO/DTMF retransmissions in * not absorbed?   Jiri Kuthan
• [Serusers] Re: [Asterisk-Users] INFO/DTMF retransmissions in * not absorbed?   Jiri Kuthan
• [Asterisk-Users] Patching Asterisk for OpenH323 ASN.1 Vulnera bilities   LEOLCH at hgc.com.hk
• [Asterisk-Users] Asterisk on ebay.   Francois Lachance
• [Asterisk-Users] NIC card failure [was: System freeze]   Scott Laird
• [Asterisk-Users] Cisco 7960 - how to enable "messages" key   Siggi Langauf
• [Asterisk-Users] Voiceglo questions   Robert Hajime Lanning
• [Asterisk-Users] Re: DISA   Robert Hajime Lanning
• [Asterisk-Users] Re: DISA   Robert Hajime Lanning
• [Asterisk-Users] Caller-ID is being sent wrong. How to fix it?   Robert Hajime Lanning
• [Asterisk-Users] Caller-ID is being sent wrong. How to fix it?   Robert Hajime Lanning
• [Asterisk-Users] Caller-ID is being sent wrong. How to fix it?   Robert Hajime Lanning
• [Asterisk-Users] Caller-ID is being sent wrong. How to fix it?   Robert Hajime Lanning
• [Asterisk-Users] Caller-ID is being sent wrong. How to fix it?   Robert Hajime Lanning
• [Asterisk-Users] port number keeps changing   Robert Hajime Lanning
• [Asterisk-Users] Dialing 800 numbers with VOIP   Robert Hajime Lanning
• [Asterisk-Users] Dialing 800 numbers with VOIP   Robert Hajime Lanning
• [Asterisk-Users] If one extension is busy...   Robert Hajime Lanning
• [Asterisk-Users] Cisco 7940 SIP Registrations   Keith Lard
• [Asterisk-Users] Howto apply a patch, diff file   Jan Larsen
• [Asterisk-Users] Howto apply a patch, diff file   Jan Larsen
• [Asterisk-Users] Changes in capi.conf   Jan Larsen
• [Asterisk-Users] "CLASS" codes or "VERTICAL SERVICE CODES"   Jan Larsen
• [Asterisk-Users] painful first steps with * and SIP phones   Jan Larsen
• [Asterisk-Users] New Insatll of *   Jan Larsen
• [Asterisk-Users] Newbie Question. Is asterisk right for my scenario?   Anthony Law
• [Asterisk-Users] question for oh323 users   Anthony Law
• [Asterisk-Users] question for oh323 users   Anthony Law
• [Asterisk-Users] question for oh323 users   Anthony Law
• [Asterisk-Users] Re: question for oh323 users   Anthony Law
• [Asterisk-Users] question for oh323 users   Anthony Law
• [Asterisk-Users] Cisco 7960G ordering Question   Robert Lawrence
• [Asterisk-Users] TDM card loses Dial tone   Robert Lawrence
• [Asterisk-Users] Galaxy Voice - Good or Bad?   Robert Lawrence
• [Asterisk-Users] Hotel wake-up   Robert Lawrence
• [Asterisk-Users] Need knowledgeable comments, bug 981 and dual redirect   Matt Lawson
• [Asterisk-Users] port number keeps changing   Matt Lawson
• [Asterisk-Users] Dialing 800 numbers with VOIP   Matt Lawson
• [Asterisk-Users] Please Explain newchan->pvt->pvt   Matt Lawson
• [Asterisk-Users] Re: speex with VoicePulse   Matt Lawson
• [Asterisk-Users] What can cause a Red alarm?   Matt Lawson
• [Asterisk-Users] Zombies got me!   Matt Lawson
• [Asterisk-Users] Re: Zombies got me! - Fixed!   Matt Lawson
• [Asterisk-Users] Re: Zombies got me! - Fixed!   Matt Lawson
• [Asterisk-Users] Woodpeckers   Matt Lawson
• [Asterisk-Users] Re: Asterisk-Users digest, Vol 1 #2879 - 10 msgs   Matt Lawson
• [Asterisk-Users] Re: Voicepulse Connection   Matt Lawson
• [Asterisk-Users] Re: Confusion with IAX PBX-PBX   Matt Lawson
• [Asterisk-Users] TDM400 showing up as Tiger Jet   Gregg Lebovitz
• [Asterisk-Users] upgrade problems   Chris Lee
• [Asterisk-Users] Whats wrong with dialplan?   Chris Lee
• [Asterisk-Users] Whats wrong with dialplan?   Chris Lee
• [OT] South african laws - was [Asterisk-Users] iax2 jitter buffer help   Chris Lee
• [Asterisk-Users] Asterisk as non root   Chris Lee
• [Asterisk-Users] Asterisk as non root   Chris Lee
• [Asterisk-Users] Help with Sip call problems - Whats not working?   Chris Lee
• [Asterisk-Users] Help with Sip call problems - Whats not working?   Chris Lee
• [Asterisk-Users] two phones one host   Chris Lee
• [Asterisk-Users] * nat internet nat sip phone howto   Chris Lee
• [Asterisk-Users] Confusion with IAX PBX-PBX   Chris Lee
• [Asterisk-Users] DSL (DMT) goes down when X100 plugged in   Chris Lee
• [Asterisk-Users] Request for enhancement - IP dependent ports   Chris Lee
• [Asterisk-Users] Norstar Integration with Asterisk via FXO or BRI ISDN   Christopher Lee
• [Asterisk-Users] Dialling Hook Flash on Zaptel   Christopher Lee
• [Asterisk-Users] Dialling Hook Flash on Zaptel   Christopher Lee
• [Asterisk-Users] Dialling Hook Flash on Zaptel   Christopher Lee
• [Asterisk-Users] Cisco 7960 quick dial   Christopher Lee
• [Asterisk-Users] Cisco 7960 quick dial   Christopher Lee
• [Asterisk-Users] Cisco 7960 quick dial   Christopher Lee
• [Asterisk-Users] Cisco 7960 quick dial   Christopher Lee
• [Asterisk-Users] Trouble emailing Digium   Christopher Lee
• [Asterisk-Users] Analog Cordless Phone Recommendations   Christopher Lee
• [Asterisk-Users] Analog Cordless Phone Recommendations   Christopher Lee
• [Asterisk-Users] Analog Cordless Phone Recommendations   Christopher Lee
• [Asterisk-Users] Analog Cordless Phone Recommendations   Christopher Lee
• [Asterisk-Users] IAX2 Call menu handling problem with Norstar   Christopher Lee
• [Asterisk-Users] IAX2 Call menu handling problem with Norstar   Christopher Lee
• [Asterisk-Users] IAX2 Call menu handling problem with Norstar - Partial Solution   Christopher Lee
• [Asterisk-Users] Simulating the "lighted line in use" type of phone   Christopher Lee
• [Asterisk-Users] Simulating the "lighted line in use" type of phone   Christopher Lee
• [Asterisk-Users] Zaptel BRI and HFC-S cards in NT-Mode   Ernst Lehmann
• [Asterisk-Users] Zaptel BRI and HFC-S cards in NT-Mode   Ernst Lehmann
• [Asterisk-Users] Zaptel BRI and HFC-S cards in NT-Mode - problem solved   Ernst Lehmann
• [Asterisk-Users] Zaptel BRI and HFC-S cards in NT-Mode - Dialout problem   Ernst Lehmann
• [Asterisk-Users] Asterisk & Panasonic KXTD - Vonage   Jacques Leisy
• [Asterisk-Users] Asterisk & Panasonic KXTD - Vonage   Jacques Leisy
• [Asterisk-Users] Hide outgoing CallerId on Zap interface   Jacques Leisy
• [Asterisk-Users] Digium connectivity issue?   Jacques Leisy
• [Asterisk-Users] Calling SIP   Jacques Leisy
• [Asterisk-Users] Calling SIP   Jacques Leisy
• [Asterisk-Users] What is the best way to debug the DTMF tones on a Zap interface   Jacques Leisy
• [Asterisk-Users] cdr mysql problem   Tilghman Lesher
• [Asterisk-Users] Fax Extension   Tilghman Lesher
• [Asterisk-Users] cdr mysql problem   Tilghman Lesher
• [Asterisk-Users] Playing announcement to called user prior toConfirmation   Tilghman Lesher
• [Asterisk-Users] x100p card conflicts with DSL modem   Tilghman Lesher
• [Asterisk-Users] sementation fault with mpg123   Tilghman Lesher
• [Asterisk-Users] talking clock   Tilghman Lesher
• [Asterisk-Users] CALEA?   Tilghman Lesher
• [Asterisk-Users] talking clock   Tilghman Lesher
• [Asterisk-Users] The Evil of type=friend explained, again (was Re: [Asterisk-Users] Minor Registration Problem With Polycom Soun dpoint IP 500)   Tilghman Lesher
• [Asterisk-Users] Asterisk as non root   Tilghman Lesher
• [Asterisk-Users] Asterisk as non root   Tilghman Lesher
• [Asterisk-Users] modprobe wcfxs   Tilghman Lesher
• [Asterisk-Users] Voicemail/Playback Questions   Tilghman Lesher
• [Asterisk-Users] X100P   Tilghman Lesher
• [Asterisk-Users] Dial-out and Dial-in modem problems.   Tilghman Lesher
• [Asterisk-Users] Wait command in auto attendant causes sched.c error   Tilghman Lesher
• [Asterisk-Users] Wait command in auto attendant causes sched.c error   Tilghman Lesher
• [Asterisk-Users] x101p beeps/sceeching   Tilghman Lesher
• [Asterisk-Users] Anybody going to the Spring VON converence [ OT]   Tilghman Lesher
• [Asterisk-Users] X100P / Echo / ZTMONITOR CAN2,3, etc.   Tilghman Lesher
• [Asterisk-Users] X100P / Echo / ZTMONITOR CAN2,3, etc.   Tilghman Lesher
• [Asterisk-Users] Re: Need to interface to BRIs   Tilghman Lesher
• [Asterisk-Users] cannot find -lXext when building * ?   Tilghman Lesher
• [Asterisk-Users] Re: Need to interface to BRIs   Tilghman Lesher
• [Asterisk-Users] Asterisk quietly segfaults   Tilghman Lesher
• [Asterisk-Users] cannot find -lXext when building * ?   Tilghman Lesher
• [Asterisk-Users] compiling * pipe error   Tilghman Lesher
• [Asterisk-Users] Round-robin chan_zap groups...   Tilghman Lesher
• [Asterisk-Users] compiling gastman   Tilghman Lesher
• [Asterisk-Users] Re: System call forked - more stuff   Tilghman Lesher
• [Asterisk-Users] switch ivr menu on/off remotely?   Tilghman Lesher
• [Asterisk-Users] VoiceMail   Tilghman Lesher
• [Asterisk-Users] Thread-safe applications   Tilghman Lesher
• [Asterisk-Users] Can't compile Zaptel drivers   Tilghman Lesher
• [Asterisk-Users] Enhancements Coming To VoicePulse Connect!   Ernest W. Lessenger
• [Asterisk-Users] IAX Connection - Voicepulse   Ernest W. Lessenger
• [Asterisk-Users] Thread-safe applications   Ernest W. Lessenger
• [Asterisk-Users] Thread-safe applications   Ernest W. Lessenger
• [Asterisk-Users] Thread-safe applications   Ernest W. Lessenger
• [Asterisk-Users] VoIP Reseller Programs   Ernest W. Lessenger
• [Asterisk-Users] Comments on Voice Quality IP Hard Phones   Ernest W. Lessenger
• [Asterisk-Users] Comments on Voice Quality IP Hard Phones   Ernest W. Lessenger
• [Asterisk-Users] Detection of extension   Ernest W. Lessenger
• [Asterisk-Users] Detection of extension   Ernest W. Lessenger
• [Asterisk-Users] Comments on Voice Quality IP Hard Phones   Ernest W. Lessenger
• [Asterisk-Users] Calls always parked on 701   Ernest W. Lessenger
• [Asterisk-Users] Calls always parked on 701   Ernest W. Lessenger
• [Asterisk-Users] Voicemail cutting off messages on SIP   Ernest W. Lessenger
• [Asterisk-Users] Cisco 7960 bug in 6.1 evident in Asterisk   Todd Lieberman
• [Asterisk-Users] Force SIP Phones to Register   Todd Lieberman
• [Asterisk-Users] Zhone + call transfer   Todd Lieberman
• [Asterisk-Users] Predictive Dialing   Todd Lieberman
• [Asterisk-Users] how to route based on DNIS   Todd Lieberman
• [Asterisk-Users] Best VOIP Analog adapter ???   Todd Lieberman
• [Asterisk-Users] asterisk-grandstream call   Paul Liew
• [Asterisk-Users] Problem connecting to Asterisk Server   Abraham Lincoln
• [Asterisk-Users] Analogical FXO vs. BRI dialing speed   Jean-Marc V. Liotier
• [Asterisk-Users] Analogical FXO vs. BRI dialing speed   Jean-Marc V. Liotier
• [Asterisk-Users] Are IAX2 providers ready for prime time?   Jean-Marc V. Liotier
• [Asterisk-Users] Are IAX2 providers ready for prime time?   Jean-Marc V. Liotier
• [Asterisk-Users] Simple Front Gate Intercom   Jean-Marc V. Liotier
• [Asterisk-Users] Connecting an ISDN (1 BRI) DECT multi phone base. NO ONE ???   Jean-Marc V. Liotier
• [Asterisk-Users] Voicetronix Audio Problems when making two or more simultanoues calls   David Liu
• [Asterisk-Users] Minor Registration Problem With Polycom Soundpoint IP 500   David Liu
• [Asterisk-Users] Minor Registration Problem With Polycom Soundpoint IP 500   David Liu
• [Asterisk-Users] Minor Registration Problem With Polycom Soundpoint IP 500   David Liu
• [Asterisk-Users] Minor Registration Problem With Polycom Soundpoint IP 500   David Liu
• [Asterisk-Users] Minor Registration Problem With Polycom Soun dpoint IP 500   David Liu
• [Asterisk-Users] Vegastream 50 FXO with Asterisk   David Liu
• [Asterisk-Users] Cisco 7960G ordering Question   David Liu
• [Asterisk-Users] Pls help for Musiconhold   David Liu
• [Asterisk-Users] Room Monitor   David Liu
• [Asterisk-Users] OT: Cisco 79XX operation   David Liu
• [Asterisk-Users] SIP over NAT   David Liu
• [Asterisk-Users] VM: Multilanguage and digits   Thorsten Lockert
• [Asterisk-Users] RE: Asterisk-Users digest, Vol 1 #2719 - 10 msgs   Alex Lopez
• [Asterisk-Users] I finally did IT!!!! Internal dial tone   Alex Lopez
• [Asterisk-Users] I finally did IT!!!! Internal dial tone   Alex Lopez
• [Asterisk-Users] SIP phones with dual ethernet.   Loucks, Jason
• [Asterisk-Users] Newbie Qu.   Carl Lougher
• [Asterisk-Users] RE:Poor Voicemail / Ivr announcement quality   Carl Lougher
• [Asterisk-Users] P2P RTP without SIP re-invites   Low, Adam
• [Asterisk-Users] P2P RTP without SIP re-invites   Low, Adam
• [Asterisk-Users] Asterisk + oh323 docs ?   Low, Adam
• [Asterisk-Users] Record conversation   Low, Adam
• [Asterisk-Users] Vegastream 50 FXO with Asterisk   Low, Adam
• [Asterisk-Users] Cisco 7960 - how to enable "messages" key   Low, Adam
• [Asterisk-Users] Analog Cordless Phone Recommendations   Low, Adam
• [Asterisk-Users] cannot find -lXext when building * ?   Low, Adam
• [Asterisk-Users] chan_sip support for SIP:Remote-Party-ID, specifically CLID priva cy   Low, Adam
• [Asterisk-Users] chan_sip support for SIP:Remote-Party-ID, sp ecificallyCLID priva cy   Low, Adam
• [Asterisk-Users] chan_sip support for SIP:Remote-Party-ID, sp ecifically CLID priva cy   Low, Adam
• [Asterisk-Users] chan_sip support for SIP:Remote-Party-ID, sp ecifically CLID priva cy   Low, Adam
• [Asterisk-Users] Best VOIP Analog adapter ???   Low, Adam
• [Asterisk-Users] Asterisk Venture   John Benson (Solutios Ltd)
• [Asterisk-Users] P2P RTP without SIP re-invites   David Luyens
• [Asterisk-Users] Eicon Diva Server card, where to purchase ?   Downlink AS / MSI
• [Asterisk-Users] Best OS for Asterisk   Ian B. MacDonald
• [Asterisk-Users] x100p dropping incoming calls   Stuart Mackintosh
• [Asterisk-Users] Mailbox full ?   Stuart Mackintosh
• [Asterisk-Users] Queue timeout reason   Stuart Mackintosh
• [Asterisk-Users] Speech between Grandstream phones sounds like talking under water   Stuart Mackintosh
• [Asterisk-Users] IAX2 queue member no auth found   Stuart Mackintosh
• [Asterisk-Users] IAX2 queue member no auth found   Stuart Mackintosh
• [Asterisk-Users] Echo   Jess Magnaye
• [Asterisk-Users] SIP phones with dual ethernet.   Jess Magnaye
• [Asterisk-Users] Using * for large VoIP implementation   Jess Magnaye
• [Asterisk-Users] Video Conference   Jess Magnaye
• [Asterisk-Users] extension mobility   Paul Mahler
• [Asterisk-Users] simple test setup   Paul Mahler
• [Asterisk-Users] SIP phones with dual ethernet.   Paul Mahler
• [Asterisk-Users] how to password protect a meetme conference?   Paul Mahler
• [Asterisk-Users] SIP phones with dual ethernet.   Paul Mahler
• [Asterisk-Users] Cisco 7960 - how to enable "messages" key   Paul Mahler
• [Asterisk-Users] Cisco 7960 - how to enable "messages" key   Paul Mahler
• [Asterisk-Users] Cisco 7960 - how to enable "messages" key   Paul Mahler
• [Asterisk-Users] Kansas SIP or IAX Provider?   Paul Mahler
• [Asterisk-Users] Kansas SIP or IAX Provider? - area codes corrected   Paul Mahler
• [Asterisk-Users] record application in extensions.conf -- how to stop recording?   Paul Mahler
• [Asterisk-Users] voip phones   Isamar Maia
• [Asterisk-Users] voip phones   Isamar Maia
• [Asterisk-Users] Cisco 7960 : No one-way audio   Isamar Maia
• [Asterisk-Users] Quintum Tenor a800 and *   Isamar Maia
• [Asterisk-Users] Prepaid Calling Card   Isamar Maia
• [Asterisk-Users] Cisco 7960 bug in 6.1 evident in Asterisk   Isamar Maia
• [Asterisk-Users] Trouble emailing Digium   Isamar Maia
• [Asterisk-Users] X100P   Isamar Maia
• [Asterisk-Users] Dialing 800 numbers with VOIP   Isamar Maia
• [Asterisk-Users] 800 numbers / Skinny -> IAX2   Isamar Maia
• [Asterisk-Users] VideoPhone   Isamar Maia
• [Asterisk-Users] call2ua Dial parameter   Isamar Maia
• [Asterisk-Users] Grandstream / SIP <-> IAX2 / Voicepulse   Isamar Maia
• [Asterisk-Users] Call priority   Isamar Maia
• [Asterisk-Users] Automated Dialing / Recording ?   C. Maj
• [Asterisk-Users] More external call control   C. Maj
• [Asterisk-Users] Asterisk for a call center?   C. Maj
• [Asterisk-Users] PRI error or what?   C. Maj
• [Asterisk-Users] Anyway to automate an AgentCallBackLogin   C. Maj
• [Asterisk-Users] Help setting up asterisk   C. Maj
• [Asterisk-Users] The Smallest Asterisk Server Ever?   Panny Malialis
• [Asterisk-Users] The Smallest Asterisk Server Ever?   Panny Malialis
• [Asterisk-Users] Asterisk compatibility list   Paulo Mannheimer
• [Asterisk-Users] Conference server   Paulo Mannheimer
• [Asterisk-Users] Recording   Paulo H. Mannheimer
• [Asterisk-Users] Asterisk Crash   Cristian Manoni
• [Asterisk-Users] Choppy Problem!!   Cristian Manoni
• [Asterisk-Users] question for oh323 users   Michael Manousos
• [Asterisk-Users] asterisk-oh323, new version 0.5.8   Michael Manousos
• [Asterisk-Users] Re: question for oh323 users   Michael Manousos
• [Asterisk-Users] asterisk-oh323 new update, v0.5.9   Michael Manousos
• [Asterisk-Users] asterisk-oh323 new update, v0.5.9   Michael Manousos
• [Asterisk-Users] Asterisk - oh323 - Cisco CallManager   Michael Manousos
• [Asterisk-Users] OH323 errors   Michael Manousos
• [Asterisk-Users] oh323 codec negotiation   Michael Manousos
• [Asterisk-Users] SIP - Native Bridge Error   Wes Marderness
• [Asterisk-Users] SIP - NATIVE BRIDGE ERROR   Wes Marderness
• [Asterisk-Users] SIP - NATIVE BRIDGE ERROR   Wes Marderness
• [Asterisk-Users] Help with Sip call problems - Whats not working?   Wes Marderness
• [Asterisk-Users] g729 license   Wes Marderness
• [Asterisk-Users] g729 license   Wes Marderness
• [Asterisk-Users] IVR (does not exist in any format)(No such file or directory)   Wes Marderness
• [Asterisk-Users] SIP Re-Invites & Timeout   Wes Marderness
• [Asterisk-Users] softphone configs?   Mark Messmore, Technical Support, University Telcom Inc.
• [Asterisk-Users] softphone configs?   Mark Messmore, Technical Support, University Telcom Inc.
• [Asterisk-Users] softphone configs?   Mark Messmore, Technical Support, University Telcom Inc.
• [Asterisk-Users] softphone configs?   Mark Messmore, Technical Support, University Telcom Inc.
• [Asterisk-Users] Mac X-Lite and Asterisk   Mark Messmore, Technical Support, University Telcom Inc.
• [Asterisk-Users] ACD   Mark Messmore, Technical Support, University Telcom Inc.
• [Asterisk-Users] calls dropped with grandstream   Mark Messmore, Technical Support, University Telcom Inc.
• [Asterisk-Users] qview.pl   Mark Messmore, Technical Support, University Telcom Inc.
• [Asterisk-Users] RE: qview.pl   Mark Messmore, Technical Support, University Telcom Inc.
• [Asterisk-Users] VOIP/IAX Termination   Matthew B Marlowe
• [Asterisk-Users] VoicePulse IAX2 lag   Matthew B Marlowe
• [Asterisk-Users] Playing announcement to called user prior toConfirmation   Matthew B Marlowe
• [Asterisk-Users] Playing announcement to called user prior toConfirmation   Matthew B Marlowe
• [Asterisk-Users] Playing announcement to called user prior toConfirmation   Matthew B Marlowe
• [Asterisk-Users] Using a Dial Statement with option m and t   Matthew B Marlowe
• [Asterisk-Users] Using a Dial Statement with option m and t   Matthew B Marlowe
• [Asterisk-Users] voip phones   Matthew B Marlowe
• [Asterisk-Users] Music on hold inside of an agi script while processing programs in background   Matthew B Marlowe
• [Asterisk-Users] Music on hold inside of an agi script while processing programs in background   Matthew B Marlowe
• [Asterisk-Users] Music on hold inside of an agi script while processing programs in background   Matthew B Marlowe
• [Asterisk-Users] Music on hold inside of an agi script while processing programs in background   Matthew B Marlowe
• [Asterisk-Users] Music on hold inside of an agi script whileprocessing programs in background   Matthew B Marlowe
• [Asterisk-Users] http://www.oneunified.net   Matthew B Marlowe
• [Asterisk-Users] RE:voiceglo sip config   Matthew B Marlowe
• [Asterisk-Users] dialout redunancy.   Matthew B Marlowe
• [Asterisk-Users] dialout redunancy.   Matthew B Marlowe
• [Asterisk-Users] New Firmware for Grandstream Phones - Supports CFG by MAC address   Matthew B Marlowe
• [Asterisk-Users] New Firmware for Grandstream Phones -Supports CFG by MAC address   Matthew B Marlowe
• [Asterisk-Users] re: SIP 481 subscription does not exist with SJPhone   Matthew B Marlowe
• [Asterisk-Users] Got my DID, getting an error.   Matthew B Marlowe
• [Asterisk-Users] VOIP Carrier recommendations?   Matthew B Marlowe
• [Asterisk-Users] Anyway to automate an AgentCallBackLogin   Matthew B Marlowe
• [Asterisk-Users] max asterisk load   Matthew B Marlowe
• [Asterisk-Users] Specials on VOIP Gear   Matthew B Marlowe
• [Asterisk-Users] Speech between Grandstream phones sounds like talking under water   Matthew B Marlowe
• [Asterisk-Users] Budgetone phones from FWD   Matthew B Marlowe
• [Asterisk-Users] running script based on if a peer is up or down   Matthew B Marlowe
• [Asterisk-Users] Enhancements Coming To VoicePulse Connect!   Matthew B Marlowe
• [Asterisk-Users] Enhancements Coming To VoicePulse Connect!   Matthew B Marlowe
• [Asterisk-Users] IAX Connection - Voicepulse   Matthew B Marlowe
• [Asterisk-Users] IAX Connection - Voicepulse   Matthew B Marlowe
• [Asterisk-Users] IAX Connection - Voicepulse   Matthew B Marlowe
• [Asterisk-Users] IAX Connection - Voicepulse   Matthew B Marlowe
• [Asterisk-Users] IAX Connection - Voicepulse   Matthew B Marlowe
• [Asterisk-Users] Call Redirection   Matthew B Marlowe
• [Asterisk-Users] Agents / ackcall   Matthew B Marlowe
• [Asterisk-Users] Simple Call Center Setup?   Matthew B Marlowe
• [Asterisk-Users] Agents / ackcall   Matthew B Marlowe
• [Asterisk-Users] Agents / ackcall   Matthew B Marlowe
• [Asterisk-Users] IAX Connection - Voicepulse   Matthew B Marlowe
• [Asterisk-Users] Call Redirection   Matthew B Marlowe
• [Asterisk-Users] Agents / ackcall   Matthew B Marlowe
• [Asterisk-Users] Voicemail Follow-up   Matthew B Marlowe
• [Asterisk-Users] Voicemail Follow-up   Matthew B Marlowe
• [Asterisk-Users] Agents / ackcall   Matthew B Marlowe
• [Asterisk-Users] Agents / ackcall   Matthew B Marlowe
• [Asterisk-Users] Agents / ackcall   Matthew B Marlowe
• [Asterisk-Users] Agents / ackcall   Matthew B Marlowe
• [Asterisk-Users] Understanding AgentCallbackLogin   Matthew B Marlowe
• [Asterisk-Users] VoIP Reseller Programs   Matthew B Marlowe
• [Asterisk-Users] Understanding AgentCallbackLogin   Matthew B Marlowe
• [Asterisk-Users] Understanding AgentCallbackLogin   Matthew B Marlowe
• [Asterisk-Users] Grandstream transfer into outer space   Matthew B Marlowe
• [Asterisk-Users] Grandstream transfer into outer space   Matthew B Marlowe
• [Asterisk-Users] Grandstream transfer into outer space   Matthew B Marlowe
• [Asterisk-Users] Grandstream transfer into outer space   Matthew B Marlowe
• [Asterisk-Users] Asterisk Venture   Matthew B Marlowe
• [Asterisk-Users] Grandstream transfer into outer space   Matthew B Marlowe
• [Asterisk-Users] Grandstream transfer into outer space   Matthew B Marlowe
• [Asterisk-Users] Re: Grandstream transfer into outer space   Matthew B Marlowe
• [Asterisk-Users] Grandstream transfer into outer space   Matthew B Marlowe
• [Asterisk-Users] Grandstream transfer into outer space   Matthew B Marlowe
• [Asterisk-Users] GS Budgetone 101 canot receive calls   Matthew B Marlowe
• [Asterisk-Users] Big Install examples please   Matthew B Marlowe
• [Asterisk-Users] vmail.cgi -> .php?   Matthew B Marlowe
• [Asterisk-Users] vmail.cgi -> .php?   Matthew B Marlowe
• [Asterisk-Users] Hotel wake-up   Matthew B Marlowe
• [Asterisk-Users] Hotel wake-up   Matthew B Marlowe
• [Asterisk-Users] Hotel wake-up   Matthew B Marlowe
• [Asterisk-Users] short ringing   Martin
• [Asterisk-Users] Large scale e.g. university   Martin
• [Asterisk-Users] short ringing   Martin
• [Asterisk-Users] NanoBGA VIA Eden-N Processor   Martin
• [Asterisk-Users] 8 lines - best approach   Darren Martz
• [Asterisk-Users] All incoming Zap calls getting picked up as FAX calls!   Darren Martz
• [Asterisk-Users] central voicemail with remote offices   Darren Martz
• [Asterisk-Users] All incoming Zap calls getting picked up asFAX calls!   Darren Martz
• [Asterisk-Users] central voicemail with remote offices   Darren Martz
• [Asterisk-Users] central voicemail with remote offices   Darren Martz
• [Asterisk-Users] Configuring Firefly Network in *   Joel Maslak
• [Asterisk-Users] Dialing 800 numbers with VOIP   Joel Maslak
• [Asterisk-Users] Woodpeckers   Joel Maslak
• [Asterisk-Users] Comments on Voice Quality IP Hard Phones   Joel Maslak
• [Asterisk-Users] Conference and transfer   Joel Maslak
• [Asterisk-Users] Can You Specify Codec Per Extension?   Matt
• [Asterisk-Users] Can You Specify Codec Per Extension?   Matt
• [Asterisk-Users] chan_h323 chan_oh323   Matt
• [Asterisk-Users] FXO Gateway of choice is?   Matt
• [Asterisk-Users] Using * for large VoIP implementation   John Matte
• [Asterisk-Users] Using * for large VoIP implementation   John Matte
• [Asterisk-Users] can a variable be redefined within extensions.conf   Brancaleoni Matteo
• [Asterisk-Users] Can asterisk make a call to a phone?   Brancaleoni Matteo
• [Asterisk-Users] Best OS for Asterisk   Brancaleoni Matteo
• [Asterisk-Users] max asterisk load   Brancaleoni Matteo
• [Asterisk-Users] recording   Brancaleoni Matteo
• [Asterisk-Users] Conference and transfer   Brancaleoni Matteo
• [Asterisk-Users] budgetones + G726   Brancaleoni Matteo
• [Asterisk-Users] zaphfc bri with overlap sending/receiving   Brancaleoni Matteo
• [Asterisk-Users] Re: Need to interface to BRIs   Reinhard Max
• [Asterisk-Users] Mailing list lag again   Reinhard Max
• [Asterisk-Users] Re: HFC-S cards?   Reinhard Max
• [Asterisk-Users] Re: Zaptel BRI and HFC-S cards   Reinhard Max
• [Asterisk-Users] International PSTN dialing   Matt McIntyre
• [Asterisk-Users] Is IAXtel down?   Matt McIntyre
• Spam Alert: [Asterisk-Users] Is IAXtel down?   Matt McIntyre
• [Asterisk-Users] RE: simple H323 question   Ron McMillan
• [Asterisk-Users] Code Hosting...   Jeremy McNamara
• [Asterisk-Users] MeetMe questions   Jeremy McNamara
• [Asterisk-Users] Minor Registration Problem With Polycom Soundpoint IP 500   Jeremy McNamara
• [Asterisk-Users] Minor Registration Problem With Polycom Soun dpoint IP 500   Jeremy McNamara
• [Asterisk-Users] The Evil of type=friend explained, again (was Re: [Asterisk-Users] Minor Registration Problem With Polycom Soun dpoint IP 500)   Jeremy McNamara
• [Asterisk-Users] Dialogic D300SC-E1   Jeremy McNamara
• [Asterisk-Users] The Evil of type=friend explained, again (was Re: [Asterisk-Users] Minor Registration Problem With Polycom Soun dpoint IP 500)   Jeremy McNamara
• [Asterisk-Users] Dialing 800 numbers with VOIP   Jeremy McNamara
• [Asterisk-Users] Revisit the Cisco 7910   Jeremy McNamara
• [Asterisk-Users] Asterisk monitor with Daemontools   Jeremy McNamara
• [Asterisk-Users] Re: 12SP registration   Jeremy McNamara
• [Asterisk-Users] Sorry, OT (NuFone)   Jeremy McNamara
• [Asterisk-Users] Load average ...   Andrew McRory
• [Asterisk-Users] Asterisk rpm packages   Andrew McRory
• [Asterisk-Users] Sip flow diagram?   Jorge Mendoza
• [Asterisk-Users] Detecting answer supervison from an AGI app   Jorge Mendoza
• [Asterisk-Users] Re: incoming call to internal user   Doug Meredith
• [Asterisk-Users] Re: asterisk-grandstream call   Doug Meredith
• [Asterisk-Users] Re: Dialing 800 numbers with VOIP   Doug Meredith
• [Asterisk-Users] Re: Re-Invites and Studder.   Doug Meredith
• [Asterisk-Users] Cisco AS5350 Gateway Intergration   Eric Merkel
• [Asterisk-Users] New Zealand   Wayne Methorst
• [Asterisk-Users] asterisk-grandstream call   Bill Michaelson
• [Asterisk-Users] Re: asterisk-grandstream call   Bill Michaelson
• [Asterisk-Users] asterisk-grandstream call   Bill Michaelson
• [Asterisk-Users] Re: Asterisk-Users digest, Vol 1 #2785 - 6 msgs   Bill Michaelson
• [Asterisk-Users] Re: asterisk-grandstream call   Bill Michaelson
• [Asterisk-Users] Re: Asterisk<->GS and codec selection   Bill Michaelson
• [Asterisk-Users] Digium connectivity issue?   Bill Michaelson
• [Asterisk-Users] Inbound IAX to SIP   Bill Michaelson
• [Asterisk-Users] System cmd usage   Bill Michaelson
• [Asterisk-Users] Re: System call succeed, asterisk sees failure   Bill Michaelson
• [Asterisk-Users] System called seems forked up   Bill Michaelson
• [Asterisk-Users] Re: System called seems forked up   Bill Michaelson
• [Asterisk-Users] Re: System call forked - more stuff   Bill Michaelson
• [Asterisk-Users] outdial broadcast   Bill Michaelson
• [Asterisk-Users] Hotel wake-up   Bill Michaelson
• [Asterisk-Users] outgoing spool parallelism   Bill Michaelson
• [Asterisk-Users] Re: Outgoing parallelism   Bill Michaelson
• [Asterisk-Users] New to the list -> some (unsolved) questions   Martin Mielke
• [Asterisk-Users] Dial via sip gateway?   Dawid Mielnik
• [Asterisk-Users] asterisk and fax over ip - concept   Dawid Mielnik
• [Asterisk-Users] asterisk and fax over ip - concept   Dawid Mielnik
• [Asterisk-Users] Cisco ATA 186   Dawid Mielnik
• [Asterisk-Users] setting up callback   Dawid Mielnik
• [Asterisk-Users] AudioCodes MP-104, register   Dawid Mielnik
• [Asterisk-Users] AudioCodes MP-104 - help !   Dawid Mielnik
• [Asterisk-Users] FXO gateways on Asterisk   Dawid Mielnik
• [Asterisk-Users] softphone configs?   Dawid Mielnik
• [Asterisk-Users] softphone configs?   Dawid Mielnik
• [Asterisk-Users] audiocodes and asterisk interoperability   Dawid Mielnik
• [Asterisk-Users] sip - t.38 through asterisk   Dawid Mielnik
• [Asterisk-Users] Asterisk Faxing   Dawid Mielnik
• [Asterisk-Users] ATA-186 x Legacy PBX   Miguel
• [Asterisk-Users] ATA-186 x Legacy PBX   Miguel
• [Asterisk-Users] ParkAndAnnounce - Get Parking Extension   Ben Miller
• [Asterisk-Users] Help setting up asterisk   Jason Miller
• [Asterisk-Users] Help setting up asterisk   Jason Miller
• [Asterisk-Users] Log entry   Dmitry Mishchenko
• [Asterisk-Users] g729, g723.1 codec translation costs   Dmitry Mishchenko
• [Asterisk-Users] voicemail auth failure   Louis-David Mitterrand
• [Asterisk-Users] FS: Adtran TotalAccess 850 Channel Bank,Router,4x4FXS   Roderick Montgomery
• [Asterisk-Users] Anyone use the Cisco 12SP+ phone w. asterisk?   Chris Moody
• [Asterisk-Users] System freeze   Jonathan Moore
• [Asterisk-Users] Xlite, GSM, Agent Logins and DTMF   Jonathan Moore
• [Asterisk-Users] Kansas SIP or IAX Provider?   Jonathan Moore
• [Asterisk-Users] Wifi Phones   Jonathan Moore
• [Asterisk-Users] Easy access to visual busy status and call transfer buttons   Jonathan Moore
• [Asterisk-Users] Switch brands, speeds, etc.   Jonathan Moore
• [Asterisk-Users] channel bank - Adit 600   Jonathan Moore
• [Asterisk-Users] Constant crashes with Asterisk 0.7.2   Jonathan Moore
• [Asterisk-Users] Wifi Phones   Jonathan Moore
• [Asterisk-Users] DIAX does not receive calls...   Jonathan Moore
• [Asterisk-Users] Wifi Phones   Jonathan Moore
• [Asterisk-Users] Room Monitor   Jonathan Moore
• [Asterisk-Users] Room Monitor   Jonathan Moore
• [Asterisk-Users] Budgetone phones from FWD   Jonathan Moore
• [Asterisk-Users] Room Monitor   Jonathan Moore
• [Asterisk-Users] Wifi Phones   Jonathan Moore
• [Asterisk-Users] Web based UA   Jonathan Moore
• [Asterisk-Users] Comments on Voice Quality IP Hard Phones   Jonathan Moore
• [Asterisk-Users] Web based UA   Jonathan Moore
• [Asterisk-Users] wisip firmware, updates, features??   Jonathan Moore
• [Asterisk-Users] Remote retrieval of voicemail, a question   Joel Barbosa Moraes
• [Asterisk-Users] TDM400 problems   John Morris
• [Asterisk-Users] g729 license   Osvaldo Mundim
• [Asterisk-Users] g729 license   Osvaldo Mundim
• [Asterisk-Users] g729 license   Osvaldo Mundim
• [Asterisk-Users] Mac X-Lite and Asterisk   Osvaldo Mundim
• [Asterisk-Users] Mac X-Lite and Asterisk   Osvaldo Mundim
• [Asterisk-Users] Mac X-Lite and Asterisk   Osvaldo Mundim
• [Asterisk-Users] Mac X-Lite and Asterisk   Osvaldo Mundim
• [Asterisk-Users] segfault with alsa   Masakazu Nakano
• [Asterisk-Users] Web based UA   Masakazu Nakano
• [Asterisk-Users] Apple OS-X   Kurt Nalty
• [Asterisk-Users] Luxoncomm 3800 series FXO/FXS adapters?   Kannaiyan Natesan
• [Asterisk-Users] Still looking for small fxo sip gateway   Kannaiyan Natesan
• [Asterisk-Users] Budgetone phones from FWD   Jason T. Nelson
• [Asterisk-Users] Fax with wildcards   Darren Nickerson
• [Asterisk-Users] Re: High Density configuration for Voice & Fax   Darren Nickerson
• [Asterisk-Users] MWI false light activity - msg0000.txt   Darren Nickerson
• [Asterisk-Users] Problems with chan_sip: random calls have no sound withouth any errors   Geert Nijpels
• [Asterisk-Users] Problems with chan_sip: random calls have no sound withouth any errors   Geert Nijpels
• [Asterisk-Users] SIP debug logs   Geert Nijpels
• [Asterisk-Users] Snom 100 Code Recommendation   Geert Nijpels
• [Asterisk-Users] Dual Xeon   Geert Nijpels
• [Asterisk-Users] Dual Xeon   Geert Nijpels
• [Asterisk-Users] Dial via sip gateway?   Grzegorz Nosek
• [Asterisk-Users] Dialling Hook Flash on Zaptel   Grzegorz Nosek
• [Asterisk-Users] IAX2 (was: diax softphone )   Grzegorz Nosek
• [Asterisk-Users] Callerid detection   Alfred R. Nurnberger
• [Asterisk-Users] Hide outgoing CallerId on Zap interface   Alfred R. Nurnberger
• [Asterisk-Users] Woodpeckers   Alfred R. Nurnberger
• [Asterisk-Users] Infite RTP to wrong address from DG104S   Zot O'Connor
• [Asterisk-Users] Why does the DG104S keep sending?   Zot O'Connor
• [Asterisk-Users] Siemens Gigaset not ringing the cordless phjone for very long   Zot O'Connor
• [Asterisk-Users] ParkAndAnnounce - Get Parking Extension   Carlton J. O'Riley
• [Asterisk-Users] RFC 2833 / Timestamp   Gerard O'Rourke
• [Asterisk-Users] Connecting an ISDN DECT phone base   Frederic Olivie
• [Asterisk-Users] Connecting an ISDN (1 BRI) DECT multi phone base. NO ONE ???   Frederic Olivie
• [Asterisk-Users] Snom 200 MWI Button   Paul Oster
• [Asterisk-Users] Snom 200 MWI Button   Paul Oster
• [Asterisk-Users] SIP Codec selection order   Alex Ovcharenko
• [Asterisk-Users] SUBSCRIBE in chan_sip - anyone?   Florian Overkamp
• [Asterisk-Users] SUBSCRIBE in chan_sip - anyone?   Florian Overkamp
• [Asterisk-Users] Smallest server continued...   Florian Overkamp
• [Asterisk-Users] Whats wrong with dialplan?   Florian Overkamp
• [Asterisk-Users] Problems with ATA's locking up..   Florian Overkamp
• [Asterisk-Users] Problems with ATA's locking up..   Florian Overkamp
• [Asterisk-Users] Problems with ATA's locking up..   Florian Overkamp
• [Asterisk-Users] Problems with ATA's locking up..   Florian Overkamp
• [Asterisk-Users] RTP with ATA186 ?   Florian Overkamp
• [Asterisk-Users] Re: Speex == Screech using version 1.1.4   Florian Overkamp
• [Asterisk-Users] asterisk and fax over ip - concept   Florian Overkamp
• [Asterisk-Users] Loading module chan_capi.so failed!   Florian Overkamp
• [Asterisk-Users] Need to interface to BRIs   Florian Overkamp
• [Asterisk-Users] HFC-S cards?   Florian Overkamp
• [Asterisk-Users] Wifi Phones   Florian Overkamp
• [Asterisk-Users] Re: Zaptel BRI and HFC-S cards   Florian Overkamp
• [Asterisk-Users] Callerid & AGI Thougts   Florian Overkamp
• [Asterisk-Users] IVR (does not exist in any format)(No such file or directory)   Florian Overkamp
• [Asterisk-Users] Simulating the "lighted line in use" type of phone   Florian Overkamp
• [Asterisk-Users] "CLASS" codes or "VERTICAL SERVICE CODES"   Florian Overkamp
• [Asterisk-Users] CISCO ATA 188   Florian Overkamp
• [Asterisk-Users] OTish: Firefly Crashing with *   Florian Overkamp
• [Asterisk-Users] Dialing out after caller leaves message   Florian Overkamp
• [Asterisk-Users] s/asterisk mailinglists/asterisk forum/g ?   Jamie P
• [Asterisk-Users] GSM codec with Cisco equipment   Juan J. Sierralta P.
• [Asterisk-Users] TE405P and dual Athlon systems   Juan J. Sierralta P.
• [Asterisk-Users] GSM codec with Cisco equipment   Juan J. Sierralta P.
• [Asterisk-Users] TE405P and dual Athlon systems   Juan J. Sierralta P.
• [Asterisk-Users] Festival patch ?   Juan J. Sierralta P.
• [Asterisk-Users] MeetMe questions   PBXtech
• [Asterisk-Users] MeetMe questions   PBXtech
• [Asterisk-Users] MeetMe questions   PBXtech
• [Asterisk-Users] MeetMe questions   PBXtech
• [Asterisk-Users] MeetMe questions   PBXtech
• [Asterisk-Users] Surveys   PBXtech
• [Asterisk-Users] help *** newbie   FRANCISCO PEREZ-LANDAETA
• [Asterisk-Users] SMDI on *   Dave Packham
• [Asterisk-Users] Need Origination Number form Bahamas   Dave Packham
• [Asterisk-Users] How to best debug SIP registration failure   George Pajari
• [Asterisk-Users] Re: How to best debug SIP registration failure   George Pajari
• [Asterisk-Users] Re: How to best debug SIP registration failure (Solved)   George Pajari
• [Asterisk-Users] Voiceglo questions   Cameron Palmer
• [Asterisk-Users] Voiceglo questions   Cameron Palmer
• [Asterisk-Users] Cisco 7960 bug in 6.1 evident in Asterisk   Cameron Palmer
• [Asterisk-Users] Voiceglo questions, IAX   Cameron Palmer
• [Asterisk-Users] Cisco 7960 bug in 6.1 evident in Asterisk   Cameron Palmer
• [Asterisk-Users] [OT] Looking for Manual: Clarent CPG 101   Miguel A Paraz
• [Asterisk-Users] Pictures of new multiport FXO/FXS from digum   Terence Parker
• [Asterisk-Users] Unable to create vpb channel   Terence Parker
• [Asterisk-Users] how to password protect a meetme conference?   Patrick
• [Asterisk-Users] Processor load spikes   Patrick
• [Asterisk-Users] Meetme without zaptel hardware   Paul
• [Asterisk-Users] asterisk 0.7.1 - mysql   John Payne
• [Asterisk-Users] setting up ---- newbie   Francisco Perez-Landaeta
• [Asterisk-Users] Asterisk setup.-   Francisco Perez-Landaeta
• [Asterisk-Users] fwd settings   Francisco Perez-Landaeta
• [Asterisk-Users] TECHNOLAND SERVERS.-   Francisco Perez-Landaeta
• [Asterisk-Users] Dialing 800 numbers with VOIP   Tim Petlock
• [Asterisk-Users] Dialing 800 numbers with VOIP   Tim Petlock
• [Asterisk-Users] Spurious DTMF tones heard by the person being called   Tim Petlock
• [Asterisk-Users] Termination - Cuba   Tim Petlock
• [Asterisk-Users] Re: Need to interface to BRIs   Tim Petlock
• [Asterisk-Users] Re: Need to interface to BRIs   Tim Petlock
• [Asterisk-Users] agi scripting in perl - dealiing with unexpected disconnects gracefully / spurious DTMF   Tim Petlock
• [Asterisk-Users] agi scripting in perl - dealiing withunexpected disconnects gracefully / spurious DTMF   Tim Petlock
• [Asterisk-Users] agi scripting in perl - dealiing withunexpected disconnects gracefully / spurious DTMF   Tim Petlock
• [Asterisk-Users] ATA 186 Registration!!!!   Tim Petlock
• [Asterisk-Users] SIP Stuff   Kristofer Pettijohn
• [Asterisk-Users] s/asterisk mailinglists/asterisk forum/g ?   Joe Phillips
• [Asterisk-Users] Budgetone phones from FWD   Joe Phillips
• [Asterisk-Users] anyone using GalaxyVoice?   Mark Phillips
• [Asterisk-Users] Can't compile Zaptel drivers   Mark Phillips
• [Asterisk-Users] Hide outgoing CallerId on Zap interface   Pertti Pikkarainen
• [Asterisk-Users] Hide outgoing CallerId on Zap interface   Pertti Pikkarainen
• [Asterisk-Users] Hide outgoing CallerId on Zap interface   Pertti Pikkarainen
• [Asterisk-Users] Dell 1750 server and Asteriks...   Pertti Pikkarainen
• [Asterisk-Users] Adtran 750 Configuration   Don Pobanz
• [Asterisk-Users] Pingtel Opensource PBX Announcement   Don Pobanz
• [Asterisk-Users] Ghost Calls   Brian Pollack
• [Asterisk-Users] Cisco 7960 - how to enable "messages" key   Brian Pollack
• [Asterisk-Users] Cisco 7960 - how to enable "messages" key   Brian Pollack
• [Asterisk-Users] Boards falling out...   Jon Pounder
• [Asterisk-Users] Digium connectivity issue?   Jon Pounder
• [Asterisk-Users] ADSI ports   Jon Pounder
• [Asterisk-Users] ADSI ports   Jon Pounder
• [Asterisk-Users] Simple Front Gate Intercom   Jon Pounder
• [Asterisk-Users] Simple Front Gate Intercom   Jon Pounder
• [Asterisk-Users] Code Hosting...   Andy Powell
• [Asterisk-Users] Code Hosting...   Andy Powell
• [Asterisk-Users] Snom 100 + H.323   Andy Powell
• [Asterisk-Users] OT: SNOM and TAPI   Andy Powell
• [Asterisk-Users] OT: SNOM and TAPI   Andy Powell
• [Asterisk-Users] Dialing 800 numbers with VOIP   Mukul Prasad
• [Asterisk-Users] speex with VoicePulse   Warren H. Prince
• [Asterisk-Users] Adtran 750 - what do I need   Warren H. Prince
• [Asterisk-Users] Are IAX2 providers ready for prime time?   Warren H. Prince
• [Asterisk-Users] AGI/php help needed with variables   Warren H. Prince
• [Asterisk-Users] GS and NAT   Tomas Prybil
• [Asterisk-Users] GS and NAT   Tomas Prybil
• [Asterisk-Users] SIP phones with dual ethernet.   Tomas Prybil
• [Asterisk-Users] SIP phones with dual ethernet.   Tomas Prybil
• [Asterisk-Users] sethdlc-new compile, does it?   Karl Putland
• [Asterisk-Users] Round-robin chan_zap groups...   Martin Pycko
• [Asterisk-Users] Pingtel SIPxchange IP PBX goes Open Source...   Martin Pycko
• [Asterisk-Users] Not Woodpeckers   Jose Quinteiro
• [Asterisk-Users] multiple context in sip.conf   Antonio Rabena
• [Asterisk-Users] multiple context in sip.conf   Antonio Rabena
• [Asterisk-Users] 100% of cpu   Rafa
• [Asterisk-Users] The Smallest Asterisk Server Ever?   Kevin Ragsdale
• [Asterisk-Users] Registering SJPhone with Asterisk   Yasir Rahman
• [Asterisk-Users] ISDN, CISCO and SCCP call forwarding   Matteo Rancilio
• [Asterisk-Users] busy tones   Matteo Rancilio
• [Asterisk-Users] busy tones   Matteo Rancilio
• [Asterisk-Users] busy tones   Matteo Rancilio
• [Asterisk-Users] Newbie: Chan_capi, early b3 in Italy   Matteo Rancilio
• [Asterisk-Users] ISDN update   Matteo Rancilio
• [Asterisk-Users] ISDN update   Matteo Rancilio
• [Asterisk-Users] chan_sccp: incoming calls on multiple lines   Matteo Rancilio
• [Asterisk-Users] incoming call to internal user   Matteo Rancilio
• [Asterisk-Users] CISCO 7912 SIP Problem   Matteo Rancilio
• [Asterisk-Users] Dell 1750 server and Asteriks...   Robert R. Randall
• [Asterisk-Users] Dell 1750 server and Asteriks...   Robert R. Randall
• [Asterisk-Users] IAX2 dosent work for DIAX   Vikram Rangnekar
• [Asterisk-Users] Re: IAX2 dosent work for DIAX   Vikram Rangnekar
• [Asterisk-Users] extensions   Vikram Rangnekar
• [Asterisk-Users] Re: extensions (FIXED)   Vikram Rangnekar
• [Asterisk-Users] voicemail   Vikram Rangnekar
• [Asterisk-Users] CallerID or Noise ?   Soren Rathje
• [Asterisk-Users] SIP extension "busy" when not available ??   Soren Rathje
• [Asterisk-Users] SIP extension "busy" when not available ??   Soren Rathje
• [Asterisk-Users] SIP extension "busy" when not available ??   Soren Rathje
• [Asterisk-Users] SIP extension "busy" when not available ??   Soren Rathje
• [Asterisk-Users] Bizarre ring   Lee Redmayne
• [Asterisk-Users] Caller ID Oddity   Lee Redmayne
• [Asterisk-Users] Bizarre ring   Lee Redmayne
• [Asterisk-Users] Bizarre ring   Lee Redmayne
• [Asterisk-Users] VOIP Deployment Concerns   Walt Reed
• [Asterisk-Users] Best OS for Asterisk   Walt Reed
• [Asterisk-Users] asterisk and fax over ip - concept   Walt Reed
• [Asterisk-Users] g729 license   Walt Reed
• [Asterisk-Users] Wildcard TDM400P - FXO?   Walt Reed
• [Asterisk-Users] Telemarketer handling   Walt Reed
• [Asterisk-Users] Room Monitor   Walt Reed
• [Asterisk-Users] Cisco 7960 SIP image (off-topic)   Walt Reed
• [Asterisk-Users] Minimum voice mail message limit?   Walt Reed
• [Asterisk-Users] Call Groups and outgoing line selection   Walt Reed
• [Asterisk-Users] Distinctive ring on 2 channels?   Walt Reed
• [Asterisk-Users] Simple Front Gate Intercom   Walt Reed
• [Asterisk-Users] SIP debug logs   Regovich, Timothy
• [Asterisk-Users] Sip flow diagram?   Regovich, Timothy
• [Asterisk-Users] The Evil of type=friend explained, again ( wa s Re: [Asterisk-Users] Minor Registration Problem With Polycom Soun dpoin t IP 500)   Regovich, Timothy
• [Fwd: [Asterisk-Users] Having problems with RTP packets and H old]   Regovich, Timothy
• [Fwd: [Asterisk-Users] Having problems with RTP packets and H old]   Regovich, Timothy
• [Asterisk-Users] Can't connect KPhone to asterisk   Regovich, Timothy
• [Asterisk-Users] Can't connect KPhone to asterisk   Regovich, Timothy
• [Asterisk-Users] Sip problem with IpDialog phone.   Regovich, Timothy
• [Asterisk-Users] SIP REGISTER   Regovich, Timothy
• [Asterisk-Users] Room Monitor   Regovich, Timothy
• [Asterisk-Users] An example config for using a Wildcard X100P and a SIP phone?   Regovich, Timothy
• [Asterisk-Users] An example config for using a Wildcard X100P and a SIP phone?   Regovich, Timothy
• [Asterisk-Users] Codec Order / Preference   Regovich, Timothy
• [Asterisk-Users] Web based UA   Regovich, Timothy
• [Asterisk-Users] Re: Voicemail Password Digit Timeout   Bill Reid
• [Asterisk-Users] IAX Connection - Voicepulse   Greg Retkowski
• [Asterisk-Users] IAX Connection - Voicepulse   Greg Retkowski
• [Asterisk-Users] Eicon Diva Server   Sergio Serrano Revuelto
• [Asterisk-Users] Eicon Diva Server   Sergio Serrano Revuelto
• [Asterisk-Users] RE: [Asterisk-Users] Spanish indications configurationº   Sergio Serrano Revuelto
• [Asterisk-Users] SIP Extrange Problem   Sergio Serrano Revuelto
• [Asterisk-Users] GS Budgetone 101 canot receive calls   Sergio Serrano Revuelto
• [Asterisk-Users] PCphoneline FXO to FXS box??   Sergio Serrano Revuelto
• [Asterisk-Users] Need some information   Jeroen Rikhof
• [Asterisk-Users] OT Superbowl = Linux Shake up to the world..   Steven Ringwald
• [Asterisk-Users] Voicemail/Playback Questions   Steven Ringwald
• [Asterisk-Users] Voicemail/Playback Questions   Steven Ringwald
• [Asterisk-Users] X100 => T100 Upgrade   Steven Ringwald
• [Asterisk-Users] Problem sip registration   Javier Rios
• [Asterisk-Users] About Grandstream ATA-286 and ring voltage   Joshua Roberson
• [Asterisk-Users] is it possible to turn auto answer off and on in the dialplan?   Jeff Roberts
• [Asterisk-Users] TDM card loses Dial tone   Jeff Roberts
• [Asterisk-Users] Brazilian Protocol   Alex G Robertson
• [Asterisk-Users] Anybody going to the Spring VON converence [ OT]   Chris Robertson
• [Asterisk-Users] Get new PRI working   Tim Robinson
• [Asterisk-Users] X100P analogue cards and impedance matching   Tim Robinson
• [Asterisk-Users] X100P and DTMF sending   Tim Robinson
• [Asterisk-Users] X100P and DTMF sending   Tim Robinson
• [Asterisk-Users] 2 questions about ISDN BRI   Tim Robinson
• [Asterisk-Users] SIP overlap (early dial) 484 response   Tim Robinson
• [Asterisk-Users] calls dropped with grandstream   Sean Rodger
• [Asterisk-Users] RE: Re: calls dropped with grandstream   Sean Rodger
• [Asterisk-Users] Residential Plans for Asterisk Users   Steve Rodgers
• [Asterisk-Users] Asterisk and Vonage -- possible? Other DID providers with Atlanta presence?   Steve Rodgers
• [Asterisk-Users] Video Recording   Neutel Rodrigues
• [Asterisk-Users] voicemail issue   Rohde
• [Asterisk-Users] Patching Asterisk for OpenH323 ASN.1 Vulnerabilities   Jim Rosenberg
• [Asterisk-Users] Grandstream transfer into outer space   Jim Rosenberg
• [Asterisk-Users] Patching Asterisk for OpenH323 ASN.1 Vulnerabilities   Jim Rosenberg
• [Asterisk-Users] Patching Asterisk for OpenH323 ASN.1 Vulnerabilities   Jim Rosenberg
• [Asterisk-Users] Grandstream transfer into outer space   Jim Rosenberg
• [Asterisk-Users] Grandstream transfer into outer space   Jim Rosenberg
• [Asterisk-Users] Grandstream transfer into outer space   Jim Rosenberg
• [Asterisk-Users] Patching Asterisk for OpenH323 ASN.1 Vulnera bilities   Jim Rosenberg
• [Asterisk-Users] PCphoneline FXO to FXS box??   Jim Rosenberg
• [Asterisk-Users] Configuring Firefly Network in *   Jason Ross
• [Asterisk-Users] Snom 100 Code Recommendation   Jason Ross
• [Asterisk-Users] OpenBSD 3.4 Patching   Jason Ross
• [Asterisk-Users] Snom 100 Code Recommendation   Jason Ross
• [Asterisk-Users] how to password protect a meetme conference?   Jason Ross
• [Asterisk-Users] OT: Cisco 7940 Smartnet in the UK   Jason Ross
• [Asterisk-Users] OT: Cisco 7940 Smartnet in the UK   Jason Ross
• [Asterisk-Users] Cisco 7960 SIP image (off-topic)   Jason Ross
• [Asterisk-Users] Woodpeckers   Luke Roussell
• [Asterisk-Users] freeBSD zaptel driver   Michael Rowley
• [Asterisk-Users] Audio code registration   Roy
• [Asterisk-Users] Anybody going to the Spring VON converence [OT]   Roy
• [Asterisk-Users] Anybody going to the Spring VON converence [OT]   Roy
• [Asterisk-Users] Asterisk under UML?   Scott Russ
• [Asterisk-Users] Mac X-Lite and Asterisk   Ryan
• [Asterisk-Users] 7960 MGCP dialtone problems, part 1 [long]   John S.
• [Asterisk-Users] 7960 MGCP dialtone problems, part 2 [long]   John S.
• [Asterisk-Users] RFC: Some proposals for the list digests   John S.
• [Asterisk-Users] voip phones   Tim Sailer
• [Asterisk-Users] diax softphone   Tim Sailer
• [Asterisk-Users] zaptel on Debian   Tim Sailer
• [Asterisk-Users] zaptel on Debian   Tim Sailer
• [Asterisk-Users] modprobe wcfxs   Tim Sailer
• [Asterisk-Users] modprobe wcfxs   Tim Sailer
• [Asterisk-Users] modprobe wcfxs   Tim Sailer
• [Asterisk-Users] modprobe wcfxs   Tim Sailer
• [Asterisk-Users] ringing   Tim Sailer
• [Asterisk-Users] ringing   Tim Sailer
• [Asterisk-Users] Calling SIP   Tim Sailer
• [Asterisk-Users] Calling SIP   Tim Sailer
• [Asterisk-Users] Log entry   Tim Sailer
• [Asterisk-Users] Log entry - solved   Tim Sailer
• [Asterisk-Users] Wait command in auto attendant causes sched.c error   Tim Sailer
• [Asterisk-Users] Database items   Tim Sailer
• [Asterisk-Users] GS BT-100 echo   Tim Sailer
• [Asterisk-Users] HELP!!!! Having problems Starting Asterisk   Tim Sailer
• [Asterisk-Users] Digium connectivity issue?   Tim Sailer
• [Asterisk-Users] Digium connectivity issue?   Tim Sailer
• [Asterisk-Users] Re: vm to email ?   Tim Sailer
• [Asterisk-Users] POTS Distinctive Ring   Tim Sailer
• [Asterisk-Users] Simulating the "lighted line in use" type of phone   Tim Sailer
• [Asterisk-Users] Can't compile Zaptel drivers   Tim Sailer
• [Asterisk-Users] Simulating the "lighted line in use" type of phone   Tim Sailer
• [Asterisk-Users] voicemail not working with mysql!!!!   Tim Sailer
• [Asterisk-Users] voicemail not working with mysql!!!!   Tim Sailer
• [Asterisk-Users] Simulating the "lighted line in use" type of phone   Tim Sailer
• [Asterisk-Users] Simulating the "lighted line in use" type of phone   Tim Sailer
• [Asterisk-Users] USB Phones   Tim Sailer
• [Asterisk-Users] USB Phones - Proper URL   Tim Sailer
• [Asterisk-Users] USB Phones - Proper URL   Tim Sailer
• [Asterisk-Users] gnophone   Tim Sailer
• [Asterisk-Users] Cisco AC Power Cubes for Sale   Sales
• [Asterisk-Users] Anyone working with NUFONE?   Sales
• [Asterisk-Users] WTS (20) ATA-186, Various Cisco IP Phones VOIP Gear and P/S   Sales
• [Asterisk-Users] Re: 12SP registration   Roger De Salis
• [Asterisk-Users] cdr->dst incorrect?   SamW
• [Asterisk-Users] cdr->dst incorrect?   SamW
• [Asterisk-Users] cdr->dst incorrect?   SamW
• [Asterisk-Users] G.729, show command or log to confirm it's using the G.729 codec.   Derek Samford
• [Asterisk-Users] Asterisk - Carrier Access Bank Ring through   Derek Samford
• [Asterisk-Users] Unable to create channem of type 'Zap'   Derek Samford
• [Asterisk-Users] H-324M/SIP Gateway   Joao Sampaio
• [Asterisk-Users] FW: H-324M/SIP Gateway   Joao Sampaio
• [Asterisk-Users] IAX DTMF question   Sathya
• [Asterisk-Users] dial plan question   Sathya
• [Asterisk-Users] relaxdtmf - duplicatedigits   Sathya
• [Asterisk-Users] Detection of extension   Christophe Sauthier
• [Asterisk-Users] Detection of extension   Christophe Sauthier
• [Asterisk-Users] Routing NOTIFY SIP messages   John J. Sawa
• [Asterisk-Users] retrieve_sip_conf_from_mysql.pl data format   Chad Sawyer
• [Asterisk-Users] DSL (DMT) goes down when X100 plugged in   Thomas M. Schaefer
• [Asterisk-Users] OS X   Erick Schmidt
• [Asterisk-Users] OS X -- More Specific   Erick Schmidt
• [Asterisk-Users] OS X -- Zaptel   Erick Schmidt
• [Asterisk-Users] C7-Hardware   Roger Schreiter
• [Asterisk-Users] Anyone working with NUFONE?   Steven Schroedl
• [Asterisk-Users] Intercom system (not paging system)   David Schumann
• [Asterisk-Users] Intercom system (not paging system)   David Schumann
• [Asterisk-Users] Asterisk on Feebsd , pls. HELP !   Serge
• [Asterisk-Users] Asterisk on Feebsd , pls. HELP !   Serge
• [Asterisk-Users] Asterisk on Feebsd , pls. HELP !   Serge
• [Asterisk-Users] compile error   Gonzalo Servat
• [Asterisk-Users] The Smallest Asterisk Server Ever?   James Sharp
• [Asterisk-Users] The Smallest Asterisk Server Ever?   James Sharp
• [Asterisk-Users] Newbie Question. Is asterisk right for my scenario?   James Sharp
• [Asterisk-Users] Surveys   James Sharp
• [Asterisk-Users] channel bank - Adit 600   Edwin Silva
• [Asterisk-Users] (no subject)   Cullen Simpson
• [Asterisk-Users] 12SP   Cullen Simpson
• [Asterisk-Users] alert-info and Cisco 7960 phones (6.1)   James Sizemore
• [Asterisk-Users] alert-info and Cisco 7960 phones (6.1)   James Sizemore
• [Asterisk-Users] ATA 186 Registration!!!!   James Sizemore
• [Asterisk-Users] Grandstream transfer into outer space   James Sizemore
• [Asterisk-Users] Calls always parked on 701   James Sizemore
• [Asterisk-Users] Automated Dialing / Recording ?   Rick Smith
• [Asterisk-Users] Automated Dialing / Recording ?   Rick Smith
• [Asterisk-Users] Automated Dialing / Recording ?   Rick Smith
• [Asterisk-Users] "Call did not go through"   Jim Sneeringer
• [Asterisk-Users] Pickup   Jim Sneeringer
• [Asterisk-Users] Unreliable Fax   Jim Sneeringer
• [Asterisk-Users] Calls always parked on 701   Jim Sneeringer
• [Asterisk-Users] A newbie list question   Jim Sneeringer
• [Asterisk-Users] Calls always parked on 701   Jim Sneeringer
• [Asterisk-Users] If one extension is busy...   Jim Sneeringer
• [Asterisk-Users] Official word from GalaxyVoice customer service   Steve Sobol
• [Asterisk-Users] Anybody going to the Spring VON converence [OT]   Steven Sokol
• [Asterisk-Users] Anybody going to the Spring VON converence [OT]   Steven Sokol
• [Asterisk-Users] Direct mailbox transfer   Steven Sokol
• [Asterisk-Users] Anybody going to the Spring VON converence [OT]   Steven Sokol
• [Asterisk-Users] IaxTel: Using IaxTel Numbers As Asterisk DIDs   Steven Sokol
• [Asterisk-Users] Any schedule for Digium's TDM400 FXO modules, or for IAXy?   Steven Sokol
• [Asterisk-Users] IAX Phone Bug Fix   Steven Sokol
• [Asterisk-Users] IAX Phone Update - Slight Change   Steven Sokol
• [Asterisk-Users] USB Phones   Steven Sokol
• [Asterisk-Users] USB Phones - Proper URL   Steven Sokol
• [Asterisk-Users] USB Phones - Proper URL   Steven Sokol
• [Asterisk-Users] X100P   Soragan
• [Asterisk-Users] X100P   Soragan
• [Asterisk-Users] X100P   Soragan
• [Asterisk-Users] X100P   Soragan
• [Asterisk-Users] X100P   Soragan
• [Asterisk-Users] linux 2.6   Soragan
• [Asterisk-Users] TDM400 FXO???   Mark Spencer
• [Asterisk-Users] Boards falling out...   Mark Spencer
• [Asterisk-Users] Asterisk 0.7.2   Mark Spencer
• [Asterisk-Users] Conference server   Mark Spencer
• [Asterisk-Users] Asterisk pins CPU   Mark Spencer
• [Asterisk-Users] X100P   Mark Spencer
• [Asterisk-Users] System freeze   Mark Spencer
• [Asterisk-Users] System freeze   Mark Spencer
• [Asterisk-Users] System freeze   Mark Spencer
• [Asterisk-Users] Digium connectivity issue?   Mark Spencer
• [Asterisk-Users] TE405P and dual Athlon systems   Mark Spencer
• [Asterisk-Users] Incoming context based on ISDN MSN   Robert Sprockeels
• [Asterisk-Users] Incoming context based on ISDN MSN   Robert Sprockeels
• [Asterisk-Users] Incoming context based on ISDN MSN   Robert Sprockeels
• [Asterisk-Users] System cmd usage   Eric Stanley
• [Asterisk-Users] Re: Dialing 800 numbers with VOIP   Kris Stark
• [Asterisk-Users] painful first steps with * and SIP phones   Janusz Starzyk
• [Asterisk-Users] Best OS for Asterisk   Steve
• [Asterisk-Users] Can't connect KPhone to asterisk   Steve
• [Asterisk-Users] System freeze   Steve
• [Asterisk-Users] Room Monitor   Steve
• [Asterisk-Users] asterisk-oh323, new version 0.5.8   Steve
• [Asterisk-Users] Comments on Voice Quality IP Hard Phones   Steve
• [Asterisk-Users] Grandstream transfer into outer space   Steve
• [Asterisk-Users] OS X   Iain Stevenson
• [Asterisk-Users] OS X -- More Specific   Iain Stevenson
• [Asterisk-Users] Cisco ATA 186   Iain Stevenson
• [Asterisk-Users] Digium connectivity issue?   Iain Stevenson
• [Asterisk-Users] Festival: read text from external fil   Iain Stevenson
• [Asterisk-Users] Voip in the EU   Iain Stevenson
• [Asterisk-Users] Voip in the EU   Iain Stevenson
• [Asterisk-Users] Voip in the EU   Iain Stevenson
• [Asterisk-Users] Executing external script   Iain Stevenson
• [Asterisk-Users] DSL (DMT) goes down when X100 plugged in   Iain Stevenson
• [Asterisk-Users] Re: DSL (DMT) goes down when X100 plugged in   Iain Stevenson
• [Asterisk-Users] Message waiting light not coming on   Iain Stevenson
• [Asterisk-Users] Asterisk / T-bird [THREAD NAME CHANGE]   Scott Stingel
• [Asterisk-Users] [OT] Oldest Telephone   Scott Stingel
• [Asterisk-Users] [OT] Oldest Telephone   Scott Stingel
• [Asterisk-Users] [OT] Oldest Telephone   Scott Stingel
• [Asterisk-Users] s/asterisk mailinglists/asterisk forum/g ?   Scott Stingel
• [Asterisk-Users] Can asterisk make a call to a phone?   Scott Stingel
• [Asterisk-Users] "Stuck" TE410P cards   Scott Stingel
• [Asterisk-Users] Playing GSM files(s)   Scott Stingel
• [Asterisk-Users] $$$ Anybody going to the Spring VON converence [OT]   Scott Stingel
• [Asterisk-Users] X100 => T100 Upgrade   Scott Stingel
• [Asterisk-Users] TE410P PRI T1 problem   Scott Stingel
• [Asterisk-Users] Asterisk Venture   Scott Stingel
• [Asterisk-Users] outgoing spool parallelism   Scott Stingel
• [Asterisk-Users] A General question   Scott Stingel
• [Asterisk-Users] Call Queues   Jon Stockill
• [Asterisk-Users] Voip in the EU   Jon Stockill
• [Asterisk-Users] X100P / Echo / ZTMONITOR CAN2,3, etc.   Jeff Stohl
• [Asterisk-Users] X100P / Echo / ZTMONITOR CAN2,3, etc.   Jeff Stohl
• [Asterisk-Users] X100P / Echo / ZTMONITOR CAN2,3, etc.   Jeff Stohl
• [Asterisk-Users] Echo cancellation   Jeff Stohl
• [Asterisk-Users] The Smallest Asterisk Server Ever?   Christian Stredicke
• [Asterisk-Users] Sip flow diagram?   Christian Stredicke
• [Asterisk-Users] Snom 200 MWI Button   Christian Stredicke
• [Asterisk-Users] FW: SNOM 200 silence suppression   Christian Stredicke
• [Asterisk-Users] The Smallest Asterisk Server Ever?   Christian Stredicke
• [Asterisk-Users] alert-info and Cisco 7960 phones (6.1)   Christian Stredicke
• [Asterisk-Users] OT: SNOM and TAPI   Christian Stredicke
• [Asterisk-Users] OT: SNOM and TAPI   Christian Stredicke
• [Asterisk-Users] Carrier Access Access Bank 1, incomming calls only echo problems, and Adit 600   Spencer Stucki
• [Asterisk-Users] IPKall->FWD->Asterisk   William Suffill
• [Asterisk-Users] VOIP Deployment Concerns   William Suffill
• [Asterisk-Users] iax, trunking, etc.   William Suffill
• [Asterisk-Users] Caller-ID is being sent wrong. How to fix it?   William Suffill
• [Asterisk-Users] IAX Softphone Errors   William Suffill
• [Asterisk-Users] s/asterisk mailinglists/asterisk forum/g ?   William Suffill
• [Asterisk-Users] New Firmware for Grandstream Phones - Supports CFG by MAC address   William Suffill
• [Asterisk-Users] Can asterisk make a call to a phone?   William Suffill
• [Asterisk-Users] central voicemail with remote offices   William Suffill
• [Asterisk-Users] Looking for Incoming # for Area Code 713 (Houston, TX)   William Suffill
• [Asterisk-Users] Minimum voice mail message limit?   William Suffill
• [Asterisk-Users] Web based UA   William Suffill
• [Asterisk-Users] Newbie Qu.   William Suffill
• [Asterisk-Users] Asterisk Venture   William Suffill
• [Asterisk-Users] Hotel wake-up   William Suffill
• [Asterisk-Users] OTish: Firefly Crashing with *   William Suffill
• [Asterisk-Users] OTish: Firefly Crashing with *   William Suffill
• [Asterisk-Users] Address Separator hex b causes callerid rejection   Linus Surguy
• [Asterisk-Users] Address Separator hex b causes callerid rejection   Linus Surguy
• [Asterisk-Users] Address Separator hex b causes callerid rejection   Linus Surguy
• [Asterisk-Users] Voip in the EU   Linus Surguy
• [Asterisk-Users] Voip in the EU   Linus Surguy
• [Asterisk-Users] Voip in the EU   Linus Surguy
• [Asterisk-Users] Voiceglo questions   Michael Swan
• [Asterisk-Users] OT: Cisco 79XX operation   Michael Swan
• [Asterisk-Users] Background and multiple digits   Michael Swan
• [Asterisk-Users] Anyone used a Grandstream ATA286 with Asterisk   MLS Drop for SysAdmin
• [Asterisk-Users] Asterisk Randomly Stopping   MLS Drop for SysAdmin
• [Asterisk-Users] System freeze   TC
• [Asterisk-Users] "Stuck" TE410P cards   TC
• [Asterisk-Users] Simple Call Center Setup?   TC
• [Asterisk-Users] calls dropped with grandstream   TC
• [Asterisk-Users] Simulating the "lighted line in use" type of phone   TC
• [Asterisk-Users] Simple Front Gate Intercom   TC
• [Asterisk-Users] Simple Front Gate Intercom   TC
• [Asterisk-Users] Simple Front Gate Intercom   TC
• [Asterisk-Users] Re: Asterisk-Users digest, Vol 1 #2840 - 11 msgs   Howard Tarlow
• [Asterisk-Users] A General question   AR Tarzi
• [Asterisk-Users] billing question   Arretni VoIP Tech
• [Asterisk-Users] G729 and VoiceMail   Arretni VoIP Tech
• [Asterisk-Users] SIP config documentation   Arretni VoIP Tech
• [Asterisk-Users] 4 port FXO   Arretni VoIP Tech
• [Asterisk-Users] max asterisk load   Arretni VoIP Tech
• [Asterisk-Users] Termination - Cuba   Aram Ter-Martirosyan
• [Asterisk-Users] Need Origination Number form Bahamas   Aram Ter-Martirosyan
• [Asterisk-Users] VoicePulse IAX2 lag   Jonathan Tew
• [Asterisk-Users] Fax with wildcards   Thomas
• [Asterisk-Users] Code Hosting...   Andrew Thompson
• [Asterisk-Users] Typo fixes   Andrew Thompson
• [Asterisk-Users] Telemarketer handling   Andrew Thompson
• [Asterisk-Users] Re: Zombies got me! - Fixed!   Andrew Thompson
• [Asterisk-Users] Enhancements Coming To VoicePulse Connect!   Andrew Thompson
• [Asterisk-Users] SPA 2000 ringing   Andrew Thompson
• [Asterisk-Users] Pickup   Andrew Thompson
• [Asterisk-Users] SPA 2000 ringing   Andrew Thompson
• [Asterisk-Users] Simulating the "lighted line in use" type of phone   Andrew Thompson
• [Asterisk-Users] Transferring Incoming Calls Twice   Andrew Thompson
• [Asterisk-Users] Can You Specify Codec Per Extension?   Andrew Thompson
• [Asterisk-Users] Core dump crash   Andrew Thompson
• [Asterisk-Users] exit   Andrew Thompson
• [Asterisk-Users] Problem connecting to Asterisk Server   Andrew Thompson
• [Asterisk-Users] Problem connecting to Asterisk Server   Andrew Thompson
• [Asterisk-Users] Can You Specify Codec Per Extension?   Andrew Thompson
• [Asterisk-Users] billing question   Andrew Thompson
• [Asterisk-Users] A working number at enum.fierymoon.com?   Andrew Thompson
• [Asterisk-Users] Timestamps   Andrew Thompson
• [Asterisk-Users] RE: Problem connecting to ASkterisk Server   Andrew Thompson
• [Asterisk-Users] The Smallest Asterisk Server Ever?   James H. Thompson
• [Asterisk-Users] The Smallest Asterisk Server Ever?   James H. Thompson
• [Asterisk-Users] Sip flow diagram? Answers   James H. Thompson
• [Asterisk-Users] Nortel and Asterisk interconnection   James H. Thompson
• [Asterisk-Users] http://www.oneunified.net   James H. Thompson
• [Asterisk-Users] Intercom system (not paging system)   James H. Thompson
• [Asterisk-Users] Intercom system (not paging system)   James H. Thompson
• [Asterisk-Users] VideoPhone   James H. Thompson
• [Asterisk-Users] Pingtel Opensource PBX Announcement   James H. Thompson
• [Asterisk-Users] Simple Front Gate Intercom   James H. Thompson
• [Asterisk-Users] Simple Front Gate Intercom   James H. Thompson
• [Asterisk-Users] X100P   Anton Tinchev
• [Asterisk-Users] Problems with ATA's locking up..   Anton Tinchev
• [Asterisk-Users] Cisco AS5350 Gateway Intergration   Anton Tinchev
• [Asterisk-Users] How to best debug SIP registration failure   Anton Tinchev
• [Asterisk-Users] New Wiki page: Dimensioning an Asterisk system   Anton Tinchev
• [Asterisk-Users] Some ADSI script programming guide?   Anton Tinchev
• [Asterisk-Users] Luxoncomm 3800 series FXO/FXS adapters?   John Todd
• [Asterisk-Users] Luxoncomm 3800 series FXO/FXS adapters?   John Todd
• [Asterisk-Users] Norstar Integration with Asterisk via FXO or BRI ISDN   John Todd
• [Asterisk-Users] Dialling Hook Flash on Zaptel   John Todd
• [Asterisk-Users] Using a Dial Statement with option m and t   John Todd
• [Asterisk-Users] Still looking for small fxo sip gateway   John Todd
• [Asterisk-Users] Re: Call pre-announcement (was: Asterisk-Users digest, Vol 1 #2711 - 15 msgs)   John Todd
• [Asterisk-Users] voicemail issue   John Todd
• [Asterisk-Users] Detecting answer supervison from an AGI app   John Todd
• [Asterisk-Users] Cisco 7960 bug in 6.1 evident in Asterisk   John Todd
• [Asterisk-Users] Code Hosting...   John Todd
• [Asterisk-Users] Cisco 7960 bug in 6.1 evident in Asterisk   John Todd
• [Asterisk-Users] talking clock   John Todd
• [Asterisk-Users] CALEA?   John Todd
• [Asterisk-Users] ParkAndAnnounce - Get Parking Extension   John Todd
• [Asterisk-Users] Cisco 7960 bug in 6.1 evident in Asterisk   John Todd
• [Asterisk-Users] talking clock   John Todd
• [Asterisk-Users] Apple OS-X   John Todd
• [Asterisk-Users] Re: [Asterisk-Dev] DISA   John Todd
• [Asterisk-Users] Cisco 7960 bug in 6.1 evident in Asterisk   John Todd
• [Asterisk-Users] Re: DISA   John Todd
• [Asterisk-Users] Re: DISA   John Todd
• [Asterisk-Users] Intercom system (not paging system)   John Todd
• [Asterisk-Users] 1.freenum.org. [was: Re: Dialing 800 numbers with VOIP]   John Todd
• [Asterisk-Users] Asterisk Critical Mass: Thursday, Miami, 9:00 PM   John Todd
• [Asterisk-Users] festival voices   John Todd
• [Asterisk-Users] GSM codec with Cisco equipment   John Todd
• [Asterisk-Users] Anybody going to the Spring VON converence [OT]   John Todd
• [Asterisk-Users] Do you Linux softphone..   Chris Tooley
• [Asterisk-Users] Asterisk 0.7.2 RPMS Updated   Chris Tooley
• [Asterisk-Users] Asterisk 0.7.2 RPMS Updated   Chris Tooley
• [Asterisk-Users] Asterisk 0.7.2 RPMS Updated   Chris Tooley
• [Asterisk-Users] Asterisk 0.7.2 RPMS Updated   Chris Tooley
• [Asterisk-Users] Asterisk 0.7.2 RPMS Updated   Chris Tooley
• [Asterisk-Users] PCMCIA   Chris Tooley
• [Asterisk-Users] FCC forces 911 services   Chris Tooley
• [Asterisk-Users] Comments on Voice Quality IP Hard Phones   Chris Tooley
• [Asterisk-Users] Enhancements Coming To VoicePulse Connect!   Steve Totaro
• [Asterisk-Users] Registering Polycom IP 500 with Asterisk [revised]   James Treleaven
• [Asterisk-Users] Re: Registering Polycom IP 500 with Asterisk [re vised]   James Treleaven
• [Asterisk-Users] Noise and scratches when there are two concurrent CAPI calls   Costa Tsaousis
• [Asterisk-Users] High Density configuration for Voice & Fax   Costa Tsaousis
• [Asterisk-Users] CAPI noise and scratches - strange behavior   Costa Tsaousis
• [Asterisk-Users] SIP config documentation   Costa Tsaousis
• [Asterisk-Users] Re: [Asterisk-Users] SIP config documentation   Costa Tsaousis
• [Asterisk-Users] Re: [Asterisk-Users]  SIP config documentation   Costa Tsaousis
• [Asterisk-Users] Re: [Asterisk-Users] Dual Xeon   Costa Tsaousis
• [Asterisk-Users] T100P & T1   Tri Tu
• [Asterisk-Users] Anyone working with NUFONE?   Tri Tu
• [Asterisk-Users] Anyone working with NUFONE?   Tri Tu
• [Asterisk-Users] T1 inbound dialplan   Tri Tu
• [Asterisk-Users] talking clock   Dan Tucny
• [Asterisk-Users] compiling * pipe error   Graham Tyers
• [Asterisk-Users] TDM card loses Dial tone   Ulexus
• [Asterisk-Users] Dialogic D300SC-E1   Steve Underwood
• [Asterisk-Users] Analogical FXO vs. BRI dialing speed   Steve Underwood
• [Asterisk-Users] Voip in the EU   Steve Underwood
• [Asterisk-Users] Cell Phones and Tone Recognition   Steve Underwood
• [Asterisk-Users] Woodpeckers   Steve Underwood
• [Asterisk-Users] Woodpeckers   Steve Underwood
• [Asterisk-Users] Woodpeckers   Steve Underwood
• [Asterisk-Users] Woodpeckers   Steve Underwood
• [Asterisk-Users] Woodpeckers   Steve Underwood
• [Asterisk-Users] Woodpeckers   Steve Underwood
• [Asterisk-Users] Woodpeckers   Steve Underwood
• [Asterisk-Users] Woodpeckers   Steve Underwood
• [Asterisk-Users] Not Woodpeckers   Steve Underwood
• [Asterisk-Users] Pingtel Opensource PBX Announcement   Steve Underwood
• [Asterisk-Users] A missing argument   Steve Underwood
• [Asterisk-Users] Re: [Asterisk-Users] Dual Xeon   Steve Underwood
• [Asterisk-Users] Asterisk and Multicast   Asterisk User
• [Asterisk-Users] ATA 186 Registration!!!!   Erick Weber V.
• [Asterisk-Users] ATA 186 Registration!!!!   Erick Weber V.
• [Asterisk-Users] cisco 7912 voicemail/dnd issue   Swen Veckes
• AW: [Asterisk-Users] cisco 7912 voicemail/dnd issue   Swen Veckes
• AW: [Asterisk-Users] cisco 7912 voicemail/dnd issue   Swen Veckes
• [Asterisk-Users] Unable to create channem of type 'Zap'   Wim Venneman
• [Asterisk-Users] Unable to create channem of type 'Zap'   Wim Venneman
• [Asterisk-Users] Unable to create channem of type 'Zap'   Wim Venneman
• [Asterisk-Users] Unable to create channem of type 'Zap'   Wim Venneman
• [Asterisk-Users] Unable to create channem of type 'Zap'   Wim Venneman
• [Asterisk-Users] Unable to create channel of type 'Zap'   Wim Venneman
• [Asterisk-Users] Unable to create channel of type 'Zap'   Wim Venneman
• [Asterisk-Users] Zaptel BRI and HFC-S cards   Armand A. Verstappen
• [Asterisk-Users] Zaptel BRI and HFC-S cards in NT-Mode   Armand A. Verstappen
• [Asterisk-Users] Zaptel BRI and HFC-S cards in NT-Mode   Armand A. Verstappen
• [Asterisk-Users] exit   Alex Volkov
• [Asterisk-Users] TDM card loses Dial tone   John Vozza
• [Asterisk-Users] talking clock   Craig Waddington
• [Asterisk-Users] fwd settings   Craig Waddington
• [Asterisk-Users] Wifi Phones   Craig Waddington
• [Asterisk-Users] New Wiki page: Dimensioning an Asterisk system   Craig Waddington
• [Asterisk-Users] Compiling while * is running   William Waites
• [Asterisk-Users] Compiling while * is running   William Waites
• [Asterisk-Users] Using a Dial Statement with option m and t   William Waites
• [Asterisk-Users] Smallest server continued...   William Waites
• [Asterisk-Users] Code Hosting...   William Waites
• [Asterisk-Users] asterisk and fax over ip - concept   William Waites
• [Asterisk-Users] asterisk and fax over ip - concept   William Waites
• [Asterisk-Users] Calling from Iaxtel to FWD users always busy   William Waites
• [Asterisk-Users] Asterisk on Feebsd , pls. HELP !   William Waites
• [Asterisk-Users] Oh323 question   Todd Wallace
• [Asterisk-Users] Wrong Pitch 1st subfr.   Todd Wallace
• [Asterisk-Users] Buzzing on Grandstream phones   Todd Wallace
• [Asterisk-Users] OH323 errors   Todd Wallace
• [Asterisk-Users] OH323 errors   Todd Wallace
• [Asterisk-Users] OH323 errors   Todd Wallace
• [Asterisk-Users] Registering SJPhone with Asterisk   Kevin Walsh
• [Asterisk-Users] Boards falling out...   Tom Walsh
• [Asterisk-Users] Cisco 7960 SIP image (off-topic)   Hermann Wecke
• [Asterisk-Users] Calls always parked on 701   Hermann Wecke
• [Asterisk-Users] 7960 x SCCP/Skinny (off-topic)   Hermann Wecke
• [Asterisk-Users] DTA-310 Outbound Dialing   Hermann Wecke
• [Asterisk-Users] CISCO ATA 188   Hermann Wecke
• [Asterisk-Users] Unable to specify channel 1: No such device or address   Hermann Wecke
• [Asterisk-Users] RE: Rhino channel bank and aastra PT390 phones   Dave Weis
• [Asterisk-Users] Wanpipe cards?   Dave Weis
• [Asterisk-Users] extravagant behavior, nat problem ?   Dave Weis
• [Asterisk-Users] iaxtel to asterisk config help.   Scott Weis
• [Asterisk-Users] FXO Gateway of choice is?   Scott Weis
• [Asterisk-Users] [OT] Oldest Telephone   Michael Welter
• [Asterisk-Users] Adtran 750 Configuration   Michael Welter
• [Asterisk-Users] System freeze   Michael Welter
• [Asterisk-Users] System freeze   Michael Welter
• [Asterisk-Users] System freeze   Michael Welter
• [Asterisk-Users] System freeze   Michael Welter
• [Asterisk-Users] T1 PRI CallerID   Michael Welter
• [Asterisk-Users] T1 PRI CallerID   Michael Welter
• [Asterisk-Users] T1 PRI CallerID   Michael Welter
• [Asterisk-Users] System freeze   Michael Welter
• [Asterisk-Users] Asterisk and Wildcard T100P   Michael Welter
• [Asterisk-Users] "Integrated" T1 PRI (voice and data)   Michael Welter
• [Asterisk-Users] Adtran 750 - what do I need   Michael Welter
• [Asterisk-Users] Overhead Paging   Michael Welter
• [Asterisk-Users] Overhead Paging   Michael Welter
• [Asterisk-Users] Woodpeckers   Michael Welter
• [Asterisk-Users] Re: Asterisk-Users digest, Vol 1 #2879 - 10 msgs   Michael Welter
• [Asterisk-Users] Re: Asterisk-Users digest, Vol 1 #2879 - 10 msgs   Michael Welter
• [Asterisk-Users] T1 inbound dialplan   Michael Welter
• [Asterisk-Users] System freeze   Michael Welter
• [Asterisk-Users] Fujitsu 9600   Michael Welter
• [Asterisk-Users] Hotel wake-up   Michael Welter
• [Asterisk-Users] Residential Plans for Asterisk Users   Ben Wern
• [Asterisk-Users] agent autologoff   Brian West
• [Asterisk-Users] voicemail issue   Brian West
• [Asterisk-Users] Cisco 7960 bug in 6.1 evident in Asterisk   Brian West
• [Asterisk-Users] Cisco 7960 quick dial   Brian West
• [Asterisk-Users] Cisco 7960 quick dial   Brian West
• [Asterisk-Users] Code Hosting...   Brian West
• [Asterisk-Users] talking clock   Brian West
• [Asterisk-Users] talking clock   Brian West
• [Asterisk-Users] Cepstral TTS Code   Brian West
• [Asterisk-Users] Re: iax, trunking, etc.   Brian West
• [Asterisk-Users] Cepstral TTS Code   Brian West
• [Asterisk-Users] Cepstral TTS Code   Brian West
• [Asterisk-Users] Cisco 7960 bug in 6.1 evident in Asterisk   Brian West
• [Asterisk-Users] Asterisk 0.7.2   Brian West
• [Asterisk-Users] MeetMe questions   Brian West
• [Asterisk-Users] MeetMe questions   Brian West
• [Asterisk-Users] MeetMe questions   Brian West
• [Asterisk-Users] Re: Call pre-announcement (was: Asterisk-Users digest, Vol 1 #2711 - 15 msgs)   Brian West
• [Asterisk-Users] Music on hold inside of an agi script while processing programs in background   Brian West
• [Asterisk-Users] Music on hold inside of an agi script while processing programs in background   Brian West
• [Asterisk-Users] Music on hold inside of an agi script while processing programs in background   Brian West
• [Asterisk-Users] ParkAndAnnounce - Get Parking Extension   Brian West
• [Asterisk-Users] Cisco 7960 bug in 6.1 evident in Asterisk   Brian West
• [Asterisk-Users] has Allison recorded "Do Not Disturb"   Brian West
• [Asterisk-Users] Asterisk Randomly Stopping   Brian West
• [Asterisk-Users] Answer from a specific Number   Brian West
• [Asterisk-Users] Adding another X100P after X100P and TDM400P is already configured   Brian West
• [Asterisk-Users] Trouble emailing Digium   Brian West
• [Asterisk-Users] Asterisk 0.7.2 RPMS Updated   Brian West
• [Asterisk-Users] Asterisk 0.7.2 RPMS Updated   Brian West
• [Asterisk-Users] Asterisk 0.7.2 RPMS Updated   Brian West
• [Asterisk-Users] Asterisk 0.7.2 RPMS Updated   Brian West
• [Asterisk-Users] Silencing Background App during touch tone detection   Brian West
• [Asterisk-Users] SIP - NATIVE BRIDGE ERROR   Brian West
• [Asterisk-Users] busy status   Brian West
• [Asterisk-Users] busy status   Brian West
• [Asterisk-Users] s/asterisk mailinglists/asterisk forum/g ?   Brian West
• [Asterisk-Users] dial timeout not working   Brian West
• [Asterisk-Users] X100P   Brian West
• [Asterisk-Users] alert-info and Cisco 7960 phones (6.1)   Brian West
• [Asterisk-Users] alert-info and Cisco 7960 phones (6.1)   Brian West
• [Asterisk-Users] I finally did IT!!!! Internal dial tone   Brian West
• [Asterisk-Users] asterisk-oh323 new update, v0.5.9   Brian West
• [Asterisk-Users] festival voices   Brian West
• [Asterisk-Users] Digium connectivity issue?   Brian West
• [Asterisk-Users] Digium connectivity issue?   Brian West
• [Asterisk-Users] multiple context in sip.conf   Brian West
• [Asterisk-Users] System freeze   Howard White
• [Asterisk-Users] X100P + HP DL380   Matt White
• [Asterisk-Users] recording   Darren Wiebe
• [Asterisk-Users] Calling SIP   Darren Wiebe
• [Asterisk-Users] Anyone working with NUFONE?   Darren Wiebe
• [Asterisk-Users] outdial broadcast   Darren Wiebe
• [Asterisk-Users] G729 troubles   Darren Wiebe
• [Asterisk-Users] G729 troubles   Darren Wiebe
• [Asterisk-Users] Iconnect behind NAT   Darren Wiebe
• [Asterisk-Users] HANGUPCAUSE   Eric Wieling
• [Asterisk-Users] Address Separator hex b causes callerid rejection   Eric Wieling
• [Asterisk-Users] Calls dropping off   Eric Wieling
• [Asterisk-Users] agent autologoff   Eric Wieling
• [Asterisk-Users] Problems with chan_sip: random calls have no sound withouth any errors   Eric Wieling
• [Asterisk-Users] Using a Dial Statement with option m and t   Eric Wieling
• [Asterisk-Users] x100p card conflicts with DSL modem   Eric Wieling
• [Asterisk-Users] progress on DTMF   Eric Wieling
• [Asterisk-Users] progress on DTMF   Eric Wieling
• [Asterisk-Users] Minor Registration Problem With Polycom Soundpoint IP 500   Eric Wieling
• [Asterisk-Users] Music on hold inside of an agi script while processing programs in background   Eric Wieling
• [Asterisk-Users] s/asterisk mailinglists/asterisk forum/g ?   Eric Wieling
• [Asterisk-Users] All incoming Zap calls getting picked up as FAX calls!   Eric Wieling
• [Asterisk-Users] Calling SIP   Eric Wieling
• [Asterisk-Users] play_and_record: No audio available   Eric Wieling
• [Asterisk-Users] SIP and DTA-310 was [ MGCP w/8x8 DTA-310 and as5300 pstn gateway]   Eric Wieling
• [Asterisk-Users] X100P Cards have gone belly up?   Eric Wieling
• [Asterisk-Users] ATA-186 x Legacy PBX   Eric Wieling
• [Asterisk-Users] Calls dropping off   Eric Wieling
• [Asterisk-Users] Cell Phones and Tone Recognition   Eric Wieling
• [Asterisk-Users] Calling SIP   Eric Wieling
• [Asterisk-Users] Hyuandi pstn handsets   Eric Wieling
• [Asterisk-Users] Codec Order / Preference   Eric Wieling
• [Asterisk-Users] Codec Order / Preference   Eric Wieling
• [Asterisk-Users] Hotel wake-up   Eric Wieling
• [Asterisk-Users] Re: Outgoing parallelism   Eric Wieling
• [Asterisk-Users] Hotel wake-up   Eric Wieling
• [Asterisk-Users] Zhone + call transfer   Kent Williams
• [Asterisk-Users] Zhone + call transfer   Kent Williams
• [Asterisk-Users] SIP Behind NAT (sipgate.de)   Scott James Williamson
• [Asterisk-Users] SIP Behind NAT (sipgate.de)   Scott James Williamson
• [Asterisk-Users] DNIS on X100P   Chris Wilson
• [Asterisk-Users] How do I provide redundancy and reliability w/ Asterisk?   WipeOut
• [Asterisk-Users] voip phones   WipeOut
• [Asterisk-Users] Do you Linux softphone..   WipeOut
• [Asterisk-Users] Code Hosting...   WipeOut
• [OT] South african laws - was [Asterisk-Users] iax2 jitter buffer help   WipeOut
• [OT] South african laws - was [Asterisk-Users] iax2 jitter buffer help   WipeOut
• [Asterisk-Users] X100P   WipeOut
• [Asterisk-Users] New Firmware for Grandstream Phones - Supports CFG by MAC address   WipeOut
• [Asterisk-Users] how to password protect a meetme conference?   WipeOut
• [Asterisk-Users] port number keeps changing   WipeOut
• [Asterisk-Users] Best OS for Asterisk   WipeOut
• [Asterisk-Users] Jump to extension from voice menu   WipeOut
• [Asterisk-Users] Voip in the EU   WipeOut
• [Asterisk-Users] Voip in the EU   WipeOut
• [Asterisk-Users] Callerid & AGI Thougts   WipeOut
• [Asterisk-Users] Make a phone dial remotely?   WipeOut
• [Asterisk-Users] Are IAX2 providers ready for prime time?   WipeOut
• [Asterisk-Users] Are IAX2 providers ready for prime time?   WipeOut
• [Asterisk-Users] Pingtel Opensource PBX Announcement   WipeOut
• [Asterisk-Users] sip:user@domain.tld   WipeOut
• [Asterisk-Users] RedHat 9 & VSFTPD & Digium Hardware Oddoties   Mindworks Wireless
• [Asterisk-Users] Revisit the Cisco 7910   Yonah Wolf
• [Asterisk-Users] Call center integration - passing caller id into an external app.   Yonah Wolf
• [Asterisk-Users] Wait command in auto attendant causes sched.c error   Christopher J. Wolff
• [Asterisk-Users] Configuring buttons on a CISCO 12SP+ Ip Phone (skinny.conf)   Davide Yachaya
• [Asterisk-Users] asterisk-oh323, new version 0.5.8   Khalid Yaseen
• [Asterisk-Users] Interrupted musiconhold sound when silence suppression is enabled   George Ye
• [Asterisk-Users] Interrupted musiconhold sound when silence supression is enabled   George Ye
• [Asterisk-Users] ATA in MGCP sometimes dropping calls   Anton Yurchenko
• [Asterisk-Users] long delay before asterisk returns 486 busy with sip   Anton Yurchenko
• [Asterisk-Users] Simple Call Center Setup?   Zarjazz
• [Asterisk-Users] Simple Call Center Setup?   Zarjazz
• [Asterisk-Users] how to dial and accept a call with only x100p card on Redhat linux 9.0?   Michael Zheng
• [Asterisk-Users] how to dial and accept a call with only x100p card on Redhat linux 9.0?   Michael Zheng
• [Asterisk-Users] Re: how to dial and accept a call with only   Michael Zheng
• [Asterisk-Users] x100p card conflicts with DSL modem   Michael Zheng
• [Asterisk-Users] x100p card conflicts with DSL modem   Michael Zheng
• [Asterisk-Users] Grandstream -> firefly call translator problem   Paul Zimm
• [Asterisk-Users] Grandstream -> firefly call translator problem   Paul Zimm
• [Asterisk-Users] Grandstream -> firefly call translator problem   Paul Zimm
• [Asterisk-Users] Voicetronix Audio Problems when making two or more simultanoues calls   Peter Zion
• [Asterisk-Users] How to Dial-out from USR Voice Modem   a1s1
• [Asterisk-Users] Does Digium TDM400P + X100P make a switchboard?   a1s1
• [Asterisk-Users] Dialing 800 numbers with VOIP   admin
• [Asterisk-Users] mgcp endpoint question   amg at intercable.com.ve
• [Asterisk-Users] System called seems forked up   andrewg at felinemenace.org
• [Asterisk-Users] Re: System called seems forked up   andrewg at felinemenace.org
• [Asterisk-Users] "multicasting" conference calls   andrewg at felinemenace.org
• [Asterisk-Users] "multicasting" conference calls   andrewg at felinemenace.org
• [Asterisk-Users] Re: How to best debug SIP registration failure   andrewg at felinemenace.org
• [Asterisk-Users] Re: How to best debug SIP registration failure   andrewg at felinemenace.org
• [Asterisk-Users] (no subject)   arohde at comcast.net
• [Asterisk-Users] OT Asterisk Sales Questions (Not for Asterisk itself)   arohde at comcast.net
• [Asterisk-Users] with soekris?   kemal asad
• [Asterisk-Users] Re: Asterisk-Users digest, Vol 1 #2827 - 16 msgs   kemal asad
• [Asterisk-Users] recording   kemal asad
• [Asterisk-Users] OTish: Firefly Crashing with *   asdasd
• [Asterisk-Users] OTish: Firefly Crashing with *   asdasd
• [Asterisk-Users] OTish: Firefly Crashing with *   asdasd
• [Asterisk-Users] Timestamps   asdasd
• [Asterisk-Users] Code Hosting...   ast at uc9.net
• [Asterisk-Users] Re: iax, trunking, etc.   ast at uc9.net
• [Asterisk-Users] http://www.oneunified.net   ast at uc9.net
• [Asterisk-Users] /var/spool/asterisk/outgoing issues   ast at uc9.net
• [Asterisk-Users] 2 Asterisk Boxes 2 Dev Kits   ast at uc9.net
• [Asterisk-Users] RE:voiceglo sip config   asterisk at bsius.com
• [Asterisk-Users] RE:voiceglo sip config   asterisk at bsius.com
• [Asterisk-Users] Wildcard TDM400P - FXO?   asterisk at sasami.anime.net
• [Asterisk-Users] voicemail not working with mysql!!!!   atif
• [Asterisk-Users] voicemail not working with mysql!!!!   atif
• [Asterisk-Users] cannot configure voicemail with mysql   atif
• [Asterisk-Users] Re: Asterisk-Dev digest, Vol 1 #507 - 8 msgs   atif
• [Asterisk-Users] Address Separator hex b causes callerid rejection   bam
• [Asterisk-Users] Address Separator hex b causes callerid rejection   bam
• [Asterisk-Users] Address Separator hex b causes callerid rejection   bam
• [Asterisk-Users] manipulating with numbers - StripMSD, Prefix   bam
• [Asterisk-Users] One way h323 to Cisco 7905?   bam
• [Asterisk-Users] Jump to extension from voice menu   bam
• [Asterisk-Users] T1 Ground Start signalling support ?   tony banks
• [Asterisk-Users] Connecting Nortel DMS100 to Asterisk ?   tony banks
• [Asterisk-Users] Newbie -- TE410P installation   marin blu
• [Asterisk-Users] IAX2 Problem   marin blu
• [Asterisk-Users] Newbie - help   marin blu
• [Asterisk-Users] Iaxphone problem   marin blu
• [Asterisk-Users] Help a Newbie to conf softphone   marin blu
• [Asterisk-Users] H323 SETUP ON ASTERISK??   carl
• [Asterisk-Users] DTMF Issues with SJPHONE   carl
• [Asterisk-Users] DTMF Issues with SJPHONE   carl
• [Asterisk-Users] Iconnect behind NAT   carl
• [Asterisk-Users] iax2 jitter buffer help   clive18 at webmail.co.za
• [Asterisk-Users] iax2 jitter buffer help   clive18 at webmail.co.za
• [OT] South african laws - was [Asterisk-Users] iax2 jitter buffer help   clive18 at webmail.co.za
• [Asterisk-Users] iax2 jitter buffer help   clive18 at webmail.co.za
• [Asterisk-Users] Jitter Buffer Configuration (typo in iax.conf)   clive18 at webmail.co.za
• [Asterisk-Users] asterisk to Clarent H323?   clive18 at webmail.co.za
• [Asterisk-Users] Fax with wildcards   cts4
• [Asterisk-Users] PCI expansion slots.   daryl at introspect.net
• [Asterisk-Users] Can asterisk make a call to a phone?   daryl at introspect.net
• [Asterisk-Users] Dialing 800 numbers with VOIP   daryl at introspect.net
• [Asterisk-Users] Dialing 800 numbers with VOIP   daryl at introspect.net
• [Asterisk-Users] Channel Bank   denzel-infotechs
• [Asterisk-Users] channel bank - Adit 600   denzel-infotechs
• [Asterisk-Users] Spanish indications configurationº   dfm
• [Asterisk-Users] chan_capi problem   dfm
• [Asterisk-Users] max asterisk load   dfm
• [Asterisk-Users] Executing external script   dfm
• [Asterisk-Users] 2 questions about ISDN BRI   dfm
• [Asterisk-Users] cdr mysql problem   dipak0105
• [Asterisk-Users] How much processing power is needed?   dkordick at austin.rr.com
• [Asterisk-Users] Call Queues   dkwok
• [Asterisk-Users] RE: setting up ---- newbie   dkwok
• [Asterisk-Users] Transfer of call from a call queue   dkwok
• [Asterisk-Users] Re: Asterisk-Users digest, Vol 1 #2749 - 7 msgs   dkwok
• [Asterisk-Users] Calling from Iaxtel to FWD users always busy   dkwok
• [Asterisk-Users] FWD/Iaxtel/Asterisk codec use   dkwok
• [Asterisk-Users] How can you savage a failed call transfer   dkwok
• [Asterisk-Users] x100p dropping incoming calls   dkwok
• [Asterisk-Users] codec negotiation   dkwok
• [Asterisk-Users] Re: X100P / Echo / ZTMONITOR CAN2,3, etc.   dkwok
• [Asterisk-Users] Echo cancellation   dkwok
• [Asterisk-Users] RE: codec negotiation prob solved   dkwok
• [Asterisk-Users] RE: "multicasting" conference calls   dkwok
• [Asterisk-Users] Hyuandi pstn handsets   dkwok
• [Asterisk-Users] An example config for using a Wildcard X100P and a SIP phone?   dkwok
• [Asterisk-Users] codec translation   dkwok
• [Asterisk-Users] more codec negotiation problems   dkwok
• [Asterisk-Users] system speed dial   dkwok
• [Asterisk-Users] RE: Message waiting light not coming on   dkwok
• [Asterisk-Users] Failed to start asterisk   dkwok
• [Asterisk-Users] exit   dkwok
• [Asterisk-Users] Call transfer from a queue   dkwok at iware.com.au
• [Asterisk-Users] Re: Asterisk-Users digest, Vol 1 #2849 - 12 msgs   dkwok at iware.com.au
• [Asterisk-Users] Re: Asterisk-Users digest, Vol 1 #2849 - 12 msgs   dkwok at iware.com.au
• [Asterisk-Users] Codec negotiation   dkwok at iware.com.au
• Subject: [Asterisk-Users] Grandstream / SIP <-> IAX2 / Voicepulse   dkwok at iware.com.au
• [Asterisk-Users] RE:Connection Problem - GrandStream   dkwok at iware.com.au
• [Asterisk-Users] help with h.323 outgoing calls   yair hakak
• [Asterisk-Users] re: SIP 481 subscription does not exist with SJPhone   yair hakak
• [Asterisk-Users] help a poor newbie out with SIP choppy one-way problem   yair hakak
• [Asterisk-Users] help a poor newbie out with SIP choppy one-way problem   yair hakak
• [Asterisk-Users] Lucent Definity CallerID {Scanned}   htguy
• [Asterisk-Users] Lucent Definity CallerID {Scanned}   htguy
• [Asterisk-Users] Lucent Definity CallerID {Scanned}   htguy
• [Asterisk-Users] Detecting answer supervison from an AGI app   hwstar at rodgers.sdcoxmail.com
• [Asterisk-Users] Detecting answer supervison from an AGI app   hwstar at rodgers.sdcoxmail.com
• [Asterisk-Users] The Smallest Asterisk Server Ever?   listas iPfone
• [Asterisk-Users] compact fxo device   listas iPfone
• [Asterisk-Users] Callerid detection   listas iPfone
• [Asterisk-Users] Callerid detection   listas iPfone
• [Asterisk-Users] WARNING[1142106560]   listas iPfone
• [Asterisk-Users] kernel 2.4.x .... which one?   mihai iancu
• [Asterisk-Users] Large scale e.g. university   info-lists at robertc.de
• [Asterisk-Users] Mark's Asterisk Presentation at Linux-Kongress2003   info-lists at robertc.de
• [Asterisk-Users] Code Hosting...   info-lists at robertc.de
• [Asterisk-Users] Cepstral TTS Code   info-lists at robertc.de
• [Asterisk-Users] Calling SIP   info-lists at robertc.de
• [Asterisk-Users] The Smallest Asterisk Server Ever?   info-lists at robertc.de
• [Asterisk-Users] Callerid & AGI Thougts   info-lists at robertc.de
• [Asterisk-Users] International PSTN dialing   info-lists at robertc.de
• [Asterisk-Users] "Call did not go through"   info-lists at robertc.de
• [Asterisk-Users] EMEA and Chagres Technologies   info-lists at robertc.de
• [Asterisk-Users] SIP extension "busy" when not available ??   info-lists at robertc.de
• [Asterisk-Users] SIP extension "busy" when not available ??   info-lists at robertc.de
• [Asterisk-Users] Need some information   info-lists at robertc.de
• [Asterisk-Users] record application in extensions.conf -- how to stop recording?   info-lists at robertc.de
• [Asterisk-Users] Upgrading asterisk yields broken pipe   jimmy.gatt at zivva.com
• [Asterisk-Users] Asterisk quietly segfaults   jimmy.gatt at zivva.com
• [Asterisk-Users] compile error   jjj3 jjj3
• [Asterisk-Users] compile error   jjj3 jjj3
• [Asterisk-Users] compile error   jjj3 jjj3
• [Asterisk-Users] compile error (still having problems)   jjj3 jjj3
• [Asterisk-Users] compile error (still having problems)   jjj3 jjj3
• [Asterisk-Users] sementation fault with mpg123   john
• [Asterisk-Users] sementation fault with mpg123   john
• [Asterisk-Users] RV: Strange Behaviour with DMZ   jorge
• [Asterisk-Users] Current version of gastman precompiled binary   Lenny Tropiano / asterisk.org Mailing list
• [Asterisk-Users] Pingtel SIPxchange IP PBX goes Open Source...   Lenny Tropiano / asterisk.org Mailing list
• [Asterisk-Users] Gastman doesn't draw lines properly between resources ...   Lenny Tropiano / asterisk.org Mailing list
• [Asterisk-Users] New Zealand users/contractors   matt
• [Asterisk-Users] New Zealand   matt
• [Asterisk-Users] Record conversation   matt
• [Asterisk-Users] Got my DID, getting an error.   matt
• [Asterisk-Users] Woodpeckers   matt
• [Asterisk-Users] IAX Connection - Voicepulse   matt
• [Asterisk-Users] IAX Connection - Voicepulse   matt
• [Asterisk-Users] IAX Connection - Voicepulse   matt
• [Asterisk-Users] IAX Connection - Voicepulse   matt
• [Asterisk-Users] IAX Connection - Voicepulse   matt
• [Asterisk-Users] IAX Connection - Voicepulse   matt
• [Asterisk-Users] Predictive Dialing   matt
• [Asterisk-Users] Re: sementation fault with mpg123   mattf
• [Asterisk-Users] Minor Registration Problem With Polycom Soun dpoint IP 500   mattf
• [Asterisk-Users] Minor Registration Problem With Polycom Soun dpoint IP 500   mattf
• [Asterisk-Users] Asterisk GUI Client - New verison 0.9   mattf
• [Asterisk-Users] Asterisk GUI Client - New verison 0.9   mattf
• [Asterisk-Users] Conference server   mattf
• [Asterisk-Users] System freeze   mattf
• [Asterisk-Users] alternative to mpg123 musiconhold was [Sys tem freeze]   mattf
• [Asterisk-Users] NIC card failure [was: System freeze]   mattf
• [Asterisk-Users] NIC card failure [was: System freeze]   mattf
• [Asterisk-Users] "Stuck" TE410P cards   mattf
• [Asterisk-Users] TE405P and dual Athlon systems   mattf
• [Asterisk-Users] TE405P and dual Athlon systems   mattf
• [Asterisk-Users] max asterisk load   mattf
• [Asterisk-Users] Record communication   mattf
• [Asterisk-Users] Registering Polycom IP 500 with Asterisk [re vised]   mattf
• [Asterisk-Users] About Grandstream ATA-286 and ring voltage   mattf
• [Asterisk-Users] Processor load spikes   mattf
• [Asterisk-Users] Processor load spikes   mattf
• [Asterisk-Users] Processor load spikes   mattf
• [Asterisk-Users] Simulating the "lighted line in use" type of phone   mattf
• [Asterisk-Users] Simulating the "lighted line in use" type of phone   mattf
• [Asterisk-Users] Core dump crash   mattf
• [Asterisk-Users] Core dump crash   mattf
• [Asterisk-Users] Pingtel Phones?   mgraves at mstvp.com
• [Asterisk-Users] Web based UA   mgraves at mstvp.com
• [Asterisk-Users] Termination in Phoenix   ml at neoninternet.com
• [Asterisk-Users] G.729, show command or log to confirm it's using the G.729 codec.   mvickers at real.com
• [Asterisk-Users] How do I provide redundancy and reliability w/ Asterisk?   phollenback at pobox.com
• [Asterisk-Users] How do I provide redundancy and reliability w/ Asterisk?   phollenback at pobox.com
• [Asterisk-Users] Playing announcement to called user prior toConfirmation   reseaux
• [Asterisk-Users] Queue Modified ACD for Asterisk 0.7.2   reseaux
• [Asterisk-Users] Dual Xeon   reseaux
• [Asterisk-Users] MWI false light activity - msg0000.txt   rjrae
• [Asterisk-Users] Big Install examples please   rjrae
• [Asterisk-Users] Message Not Delivered   romanp at mantraent.com
• [Asterisk-Users] diax softphone   Peer Oliver schmidt
• [Asterisk-Users] diax softphone   Peer Oliver schmidt
• [Asterisk-Users] IAX2 (was: diax softphone )   Peer Oliver schmidt
• [Asterisk-Users] no extension in callerid of outgoing calls ...   Peer Oliver schmidt
• [Asterisk-Users] asterisk and fax over ip - concept   Peer Oliver schmidt
• [Asterisk-Users] OT: SNOM and TAPI   Peer Oliver schmidt
• [Asterisk-Users] OT: SNOM and TAPI   Peer Oliver schmidt
• [Asterisk-Users] OT: SNOM and TAPI   Peer Oliver schmidt
• [Asterisk-Users] Understanding AgentCallbackLogin   Peer Oliver schmidt
• [Asterisk-Users] Comments on Voice Quality IP Hard Phones   Peer Oliver schmidt
• [Asterisk-Users] MGCP media gateway   spkao
• [Asterisk-Users] OT: Cisco 7940 Smartnet in the UK   stan
• [Asterisk-Users] DSL (DMT) goes down when X100 plugged in   stan
• [Asterisk-Users] SIP phones with dual ethernet.   starmail at prybil.se
• [Asterisk-Users] SIP phones with dual ethernet.   starmail at prybil.se
• [Asterisk-Users] Re: Asterisk-Users digest, Vol 1 #2711 - 15 msgs   tad
• [Asterisk-Users] System freeze   tan at yointernet.com
• [Asterisk-Users] Callerid & AGI Thougts   todd at tlsolutions.net
• [Asterisk-Users] Smallest server continued...   toms at xmission.com
• [Asterisk-Users] More external call control   toms at xmission.com
• [Asterisk-Users] Make a phone dial remotely?   toms at xmission.com
• [Asterisk-Users] DevLite problem with ztcfg   toms at xmission.com
• [Asterisk-Users] Re: DSL (DMT) goes down when X100 plugged in   toms at xmission.com
• [Asterisk-Users] help *** newbie   jorge verastegui
• [Asterisk-Users] Are there any list moderators?   woody+asterisk at solutionsfirst.com.au
• [Asterisk-Users] Asterisk with a laptop with built-in Intel 537 modem   woody+asterisk at solutionsfirst.com.au
• [Asterisk-Users] Pictures of new multiport FXO/FXS from digum   woody+asterisk at solutionsfirst.com.au
• [Asterisk-Users] Pictures of new multiport FXO/FXS from digum   woody+asterisk at solutionsfirst.com.au
• [Asterisk-Users] Audiocodes FXO - loop channels   woody+asterisk at solutionsfirst.com.au
• [Asterisk-Users] simple test setup   woody+asterisk at solutionsfirst.com.au
• [Asterisk-Users] PCMCIA   woody+asterisk at solutionsfirst.com.au
• [Asterisk-Users] Motherboard and fxo suggestion.   woody+asterisk at solutionsfirst.com.au
• [Asterisk-Users] Transferring Incoming Calls Twice   woody+asterisk at solutionsfirst.com.au
• [Asterisk-Users] x100p   yaboo
• [Asterisk-Users] nortel i2004 ip phone   yaboo
• [Asterisk-Users] registration   yaoyunhong
• [Asterisk-Users] Pls help for Musiconhold   young
• [Asterisk-Users] Pls help for Musiconhold   young
• [Asterisk-Users] I need patch for musiconhold-multiful format   Ç㿵