June 2006 Archives by author

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

Starting: Thu Jun 1 00:13:13 MST 2006
Ending: Fri Jun 30 23:55:56 MST 2006
Messages: 4178

• [Asterisk-Users] Looking for SIP provider with minimal call setup time   Arnaud
• [Asterisk-Users] Looking for SIP provider with minimal call setuptime   Arnaud
• [Asterisk-Users] Re: GXP-2000 (steer clear)   Francesco Peeters (Asterisk)
• [Asterisk-Users] David Choo/eServices/eSpore is overseas   Francesco Peeters (Asterisk)
• [Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!   Francesco Peeters (Asterisk)
• [Asterisk-Users] Oh oh. Micro$oft just noticed VoIP   Francesco Peeters (Asterisk)
• [Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!   Francesco Peeters (Asterisk)
• [Asterisk-Users] Polycom 601 problems with multiple registrations   Brian Vincent (C)
• [Asterisk-Users] Polycom 601 problems with multiple registrations   Brian Vincent (C)
• [Asterisk-Users] Polycom 601 problems with multiple registrations   Brian Vincent (C)
• [Asterisk-Users] when I press "transfer" -> blind -> 700 . The useris not able to hear what extension the call was parked on   Brian Vincent (C)
• [Asterisk-Users] asterisk-stat display problems   Chris Earle (CBL)
• [Asterisk-Users] asterisk-stat display problems   Chris Earle (CBL)
• [Asterisk-Users] Re: unknown host cvs.digium.com   Matt Riddell (IT)
• [Asterisk-Users] Call-pickup function in Queue application   Matt Riddell (IT)
• [Asterisk-Users] WCTDM-24xxp woes   Matt Riddell (IT)
• [Asterisk-Users] fine-tuning asterisk questions   Matt Riddell (IT)
• [Asterisk-Users] Prices of g729 codec   Matt Riddell (IT)
• [Asterisk-Users] Quad T1 Card   Matt Riddell (IT)
• [Asterisk-Users] Set(CDR(userfield)) Trouble   Matt Riddell (IT)
• [Asterisk-Users] MWI on the PA168V in IAX mode?   Matt Riddell (IT)
• [Asterisk-Users] Many asterisk server behind a redirector?   Matt Riddell (IT)
• [Asterisk-Users] Prices of g729 codec   Chris Mason (Lists)
• [Asterisk-Users] Polycom 601 question   Chris Mason (Lists)
• [Asterisk-Users] Asterisk: T1 hunt group setup   Andrei (MPI)
• [Asterisk-Users] How to redirect an incoming call to an external phone numer   Andrei (MPI)
• [Asterisk-Users] Asterisk: T1 hunt group setup   Andrei (MPI)
• [Asterisk-Users] stuck call on asterisk   Andrei (MPI)
• [Asterisk-Users] skype out   Andrei (MPI)
• [Asterisk-Users] zap calls drop suddenly + tremendous noise when answering a call   Andrei (MPI)
• [Asterisk-Users] Fun with Echo   Andrei (MPI)
• [Asterisk-Users] Any good voip providers lately?   Andrei (MPI)
• [Asterisk-Users] RAGI + Sphinx + Festival   Andrei (MPI)
• [Asterisk-Users] Audio cuts out   Andrei (MPI)
• [Asterisk-Users] Can this config sustain 30 users?   Andrei (MPI)
• [Asterisk-Users] Can this config sustain 30 users?   Andrei (MPI)
• [Asterisk-Users] TDM400P bad echo problem, tried lots of things   Andrei (MPI)
• [Asterisk-Users] best hardphone for Asterisk?   Andrei (MPI)
• [Asterisk-Users] Supporter needed   VoIP Street .com
• [Asterisk-Users] I guess my server capacity is ok   Woodoo People .pGa!
• [Asterisk-Users] Asterisk restarting in a minute   Woodoo People .pGa!
• [Asterisk-Users] dealing with trafication tone   Woodoo People .pGa!
• [Asterisk-Users] connecting asterisk to pstn help   Woodoo People .pGa!
• [Asterisk-Users] X100P fails to initialize   Woodoo People .pGa!
• [Asterisk-Users] X100P fails to initialize   Woodoo People .pGa!
• Fw: [Asterisk-Users] Compiling chan_bluetooth   Woodoo People .pGa!
• [Asterisk-Users] SIP voice recorder   Woodoo People .pGa!
• [Asterisk-Users] asterisk behind cisco pix 506   Woodoo People .pGa!
• [Asterisk-Users] Asterisk on Mini-Box M300   Woodoo People .pGa!
• [Asterisk-Users] Prices of g729 codec   Woodoo People .pGa!
• [Asterisk-Users] ISDN BRI (I.430) over ethernet   Woodoo People .pGa!
• [Asterisk-Users] wav49 size for a 3 minute voicemail   Woodoo People .pGa!
• [Asterisk-Users] ISDN BRI (I.430) over ethernet   Woodoo People .pGa!
• [Asterisk-Users] VGSM Trouble: Kind people, help me please...   Woodoo People .pGa!
• [Asterisk-Users] Detecting gateways which time out   Woodoo People .pGa!
• [Asterisk-Users] RE: VGSM Trouble: Kind people, help me please...   Woodoo People .pGa!
• [Asterisk-Users] Callback Application: Suggestions Please.   Woodoo People .pGa!
• [Asterisk-Users] Receiving faxes and then sending them on   Woodoo People .pGa!
• [Asterisk-Users] asterisk to mobile phone   Woodoo People .pGa!
• [Asterisk-Users] DISA Password Authenntication with Grandstream 488   ITN Info - 11 - 30851536
• [Asterisk-Users] RES: DISA Password Authenntication with Grandstream 488   ITN Info - 11 - 30851536
• [Asterisk-Users] Limiting the size of a Queue   Álvaro Palma
• [Asterisk-Users] FAX + Digium + SpanDSP   Adrià Vidal
• [Asterisk-Users] lspci doesn't show digium card te405p   Bruno de Assumpção Loureiro
• [Asterisk-Users] Using "#include" on zaptel.conf   Denis Galvão - iSolve
• [Asterisk-Users] Using "#include" on zaptel.conf   Denis Galvão - iSolve
• [Asterisk-Users] TE420P/TE415P?   Denis Galvão - iSolve
• [Asterisk-Users] Asterisk - Qsig   Josué Conti
• [Asterisk-Users] X100P fails to initialize   Josué Conti
• [Asterisk-Users] Asterisk - Qsig   Josué Conti
• [Asterisk-Users] Asterisk - Qsig   Josué Conti
• [Asterisk-Users] Asterisk - Qsig   Josué Conti
• [Asterisk-Users] asterisk-1.2.9.1   Josué Conti
• [Asterisk-Users] asterisk-1.2.9.1   Josué Conti
• [Asterisk-Users] - SOLVED - Trouble getting SMS working   Josué Conti
• [Asterisk-Users] - SOLVED - Trouble getting SMS working   Josué Conti
• [Asterisk-Users] Asterisk - Qsig   Josué Conti
• [Asterisk-Users]Asterisk <-IP-> Siemens HiPath 3750   Josué Conti
• [Asterisk-Users] ztdummy   Josué Conti
• [Asterisk-Users] ztdummy   Josué Conti
• [Asterisk-Users] Asterisk-1.0.9 Atxfer   Josué Conti
• [Asterisk-Users] What type of T1 Cards to use for my Asterisk PBX   Josué Conti
• [Asterisk-Users] Asterisk and Qsig   Josué Conti
• [Asterisk-Users] USB handset options for softphones   Josué Conti
• [Asterisk-Users] Asterisk-1.2.9.1 e MOH   Josué Conti
• [Asterisk-Users] Asterisk Users Group - Portugal   Josué Conti
• [Asterisk-Users] Re: Asterisk-1.2.9.1 e MOH   Josué Conti
• [Asterisk-Users] Asterisk-1.2.9.1 with Siemens HiPath 4000   Josué Conti
• [Asterisk-Users] Asterisk Users Group - Portugal   Josué Conti
• [Asterisk-Users] Asterisk and Qsig Protocol   Josué Conti
• [Asterisk-Users] Asterisk x Siemens HiPath 4000   Josué Conti
• [Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!   Josué Conti
• [Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!   Josué Conti
• [Asterisk-Users] This is getting really annoying - re: POSTFIX   Josué Conti
• [Asterisk-Users] Asterisk-1.2.9.1 with Siemens HiPath 4000   Josué Conti
• [Asterisk-Users] Asterisk-1.2.9.1 with Siemens HiPath 4000   Josué Conti
• [Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!   Josué Conti
• [Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!   Josué Conti
• [Asterisk-Users] Re: Asterisk x Siemens HiPath 4000   Josué Conti
• [Asterisk-Users] Re: Asterisk x Siemens HiPath 4000   Josué Conti
• [Asterisk-Users] Re: Asterisk-1.2.9.1 with Siemens HiPath 4000   Josué Conti
• [Asterisk-Users] HDLC Bad FCS (8)   Josué Conti
• [Asterisk-Users] Asterisk x Qsig - messages   Josué Conti
• [Asterisk-Users] Asterisk-1.2.9.1 with QSIG Protocol   Josué Conti
• [Asterisk-Users] Virtual PBX Billing and Management Software   Juan Manuel Coronado Zúñiga
• [Asterisk-Users] asterisk on AMD 64 BIT   Juan Manuel Coronado Zúñiga
• [Asterisk-Users] IP/SS7 gateway on Sun Ultra 20 amd64   Juan Manuel Coronado Zúñiga
• [Asterisk-Users] Page cmd & FOP   Nicolás Gudiño
• [Asterisk-Users] agi, STREAM FILE and SIGHUP   Nicolás Gudiño
• [Asterisk-Users] ASTCC: How to reset periodically all "card in use" flag back?   Nicolás Gudiño
• [Asterisk-Users] ASTCC: How to reset periodically all "card in use" flag back?   Nicolás Gudiño
• [Asterisk-Users] email a message   Filip Drągowski
• [Asterisk-Users] Linksys SPA-941 NAT?   Filip Drągowski
• [Asterisk-Users] /var/log/asterisk/full ?   Filip Drągowski
• [Asterisk-Users] timeout 't'   Filip Drągowski
• [Asterisk-Users] IAX2 Dial command   Filip Drągowski
• [Asterisk-Users] Kernel 2.4 / 2.6 and timer   Filip Drągowski
• [Asterisk-Users] Asterisk trunk cisco 2851   René Enskat [Teamware GmbH]
• [Asterisk-Users] calling between contexts   René Enskat [Teamware GmbH]
• [Asterisk-Users] Re: Bristuffed Asterisk: Hangup problems   Eberhard Müller
• [Asterisk-Users] Re: Bristuffed Asterisk: Hangup problems   Eberhard Müller
• [Asterisk-Users] Prices of g729 codec   Frédéric Marti
• [Asterisk-Users] Prices of g729 codec   Frédéric Marti
• [Asterisk-Users] Prices of g729 codec   Frédéric Marti
• [Asterisk-Users] DUNDi Docs   Frédéric Marti
• [Asterisk-Users] IAX2 channel problems   Jon Schøpzinsky
• SV: [Asterisk-Users] I can hear only one way when I use nokia e-60 withX-lite   Jon Schøpzinsky
• SV: [Asterisk-Users] I can hear only one way when I use nokia e-60withX-lite   Jon Schøpzinsky
• SV: SV: [Asterisk-Users] I can hear only one way when I use nokiae-60withX-lite   Jon Schøpzinsky
• [Asterisk-Users] Using regcontext   Jon Schøpzinsky
• SV: [Asterisk-Users] Using regcontext   Jon Schøpzinsky
• [Asterisk-Users] Call status subscriptions on multiple servers   Jon Schøpzinsky
• SV: [Asterisk-Users] Database file to copy for active sessions.   Jon Schøpzinsky
• SV: [Asterisk-Users] Database file to copy for active sessions.   Jon Schøpzinsky
• SV: [Asterisk-Users] Database file to copy for active sessions.   Jon Schøpzinsky
• SV: [Asterisk-Users] TSP on linux   Jon Schøpzinsky
• SV: [Asterisk-Users] Call status subscriptions on multiple servers   Jon Schøpzinsky
• [Asterisk-Users] IAX2 Vs SIP cpu load   Jon Schøpzinsky
• [Asterisk-Users] DTMF when using g.729   Jon Schøpzinsky
• SV: [Asterisk-Users] DTMF when using g.729   Jon Schøpzinsky
• [Asterisk-Users] SIP Registrations and DUNDi   Jon Schøpzinsky
• [Asterisk-Users] IAX2 Dial command   Jon Schøpzinsky
• SV: [Asterisk-Users] periodic-announce not working   Jon Schøpzinsky
• SV: [Asterisk-Users] Error in config sample for GoToIf?   Jon Schøpzinsky
• [Asterisk-Users] SIP codec preference order ineffective   Patai Tamás
• [Asterisk-Users] SIP codec preference order ineffective   Patai Tamás
• [Asterisk-Users] More Level QueueSystem   Patrick Bök
• [Asterisk-Users] Background + Dial   gaspard.laurent at tiscali.fr
• RE: [Asterisk-Users] Background + Dial   gaspard.laurent at tiscali.fr
• [Asterisk-Users] Voicemail normalization   Colin Anderson
• [Asterisk-Users] Provisional problem with SIP channel   Benoît Mérouze
• [Asterisk-Users] voip to voip bridge   Benoît Mérouze
• [Asterisk-Users] voip to voip bridge   Benoît Mérouze
• [Asterisk-Users] Echo Problem with T411P   Idris AVCI
• [Asterisk-Users] open source sip softphone (Window OS version )   Idris AVCI
• [Asterisk-Users] Asterisk & Cisco 3800   Idris AVCI
• [Asterisk-Users] Echo Problem with T411P   Idris AVCI
• [Asterisk-Users] Echo Problem with T411P   Idris AVCI
• [Asterisk-Users] Help with Audicodes MP-104   Idris AVCI
• [Asterisk-Users] SIP Softphone on Thinclient?   Idris AVCI
• [Asterisk-Users] Transfer call via AMI or dialplan   Idris AVCI
• [Asterisk-Users] Monitor a particular SIP call for training purposes   Idris AVCI
• [Asterisk-Users] Action: Originate PROBLEM   Idris AVCI
• [Asterisk-Users] Queue - Log if caller disconnects   Idris AVCI
• [Asterisk-Users] FW: Issue in configuring TDM400P   Aarthy G - TLS, Chennai.
• [Asterisk-Users] SE Michigan asterisk users group   RE Kushner List Account
• [Asterisk-Users] PHP UnixODBC MS SQl 2000   Mark Ackroyd
• [Asterisk-Users] Sangoma card A101 Card troubles.   Mark Ackroyd
• [Asterisk-Users] Linksys PAP2NA Configuration / Asterisk / Voip consultant wanted   Mark Adams
• [Asterisk-Users] Using Asterisk to better detect hangups when using ATA'S or Analog Gateways'   Mark Adams
• [Asterisk-Users] Voip / AudioCodes MP-108 Help Needed   Mark Adams
• [Asterisk-Users] Centos cause Asterisk crash   Rich Adamson
• [Asterisk-Users] Asterisk: T1 hunt group setup   Rich Adamson
• [Asterisk-Users] SIP voice recorder   Rich Adamson
• [Asterisk-Users] TDM-400 doesn't detect far-end hangup   Rich Adamson
• [Asterisk-Users] WCTDM-24xxp woes   Rich Adamson
• [Asterisk-Users] Quad T1 Card   Rich Adamson
• [Asterisk-Users] Quad T1 Card   Rich Adamson
• [Asterisk-Users] Controlling Cisco 7960 Ringtone from Asterisk   Rich Adamson
• [Asterisk-Users] ISDN master clock issue ?   Rich Adamson
• [Asterisk-Users] ISDN master clock issue ?   Rich Adamson
• [Asterisk-Users] ISDN master clock issue ?   Rich Adamson
• [Asterisk-Users] Re: Linksys PAP2T-NA - call goes through but phone doesn't ring   Rich Adamson
• [Asterisk-Users] PRI & Fax Passthrough   Rich Adamson
• [Asterisk-Users] Fun with Echo   Rich Adamson
• [Asterisk-Users] Fun with Echo   Rich Adamson
• [Asterisk-Users] Broken firewall or brain damaged admin?   Rich Adamson
• [Asterisk-Users] Bullet-proof FXO?   Rich Adamson
• [Asterisk-Users] spa3102 vs spa3000 differences?   Rich Adamson
• [Asterisk-Users] spa3102 vs spa3000 differences?   Rich Adamson
• [Asterisk-Users] TDM-400 and dialplan -- how to ring a SIP extension *before* answering the PSTN line?   Rich Adamson
• [Asterisk-Users] Linksys SRW224P POE Switch   Rich Adamson
• [Asterisk-Users] Zaptel dialing too fast?   Rich Adamson
• [Asterisk-Users] MOS Scores and LCR   Rich Adamson
• [Asterisk-Users] Echo Cancelling VoIP traffic   Rich Adamson
• [Asterisk-Users] Sipura SPA-2000 & Asterisk 1.24 w/incoming calls   Rich Adamson
• [Asterisk-Users] Echo Cancelling VoIP traffic   Rich Adamson
• [Asterisk-Users] Cisco IP Phones - FYI   Rich Adamson
• [Asterisk-Users] Zaptel answering the Line   Rich Adamson
• [Asterisk-Users] DTMF and ivr systems   Rich Adamson
• [Asterisk-Users] need help troubleshooting clipping and garbled VOIP calls   Rich Adamson
• [Asterisk-Users] Digium Hardware Reliability   Rich Adamson
• [Asterisk-Users] Recommended FXO device   Rich Adamson
• [Asterisk-Users] cheapest Cisco Smartnet contract?   Rich Adamson
• [Asterisk-Users] Digium Hardware Reliability   Rich Adamson
• [Asterisk-Users] TrixBox   Adi
• [Asterisk-Users] voiceone?   Neil Adona
• [Asterisk-Users] DTMF feedthru again...   Gabriel Afana
• [Asterisk-Users] How to use a data T-1?   Gabriel Afana
• [Asterisk-Users] Include files in voicemail.conf   Gabriel Afana
• [Asterisk-Users] Include files in voicemail.conf   Gabriel Afana
• [Asterisk-Users] New Member, saying Hi. :)   Lewis Agosta
• [Asterisk-Users] PHP-AGI help   Lewis Agosta
• Re: [Asterisk-Users] Can´t send emails   Lewis Agosta
• [Asterisk-Users] Set(CDR(userfield)) Trouble   Lewis Agosta
• [Asterisk-Users] Set(CDR(userfield)) Trouble   Lewis Agosta
• [Asterisk-Users] long distance ask for pin   Lewis Agosta
• [Asterisk-Users] Digital Receptionist   Lewis Agosta
• [Asterisk-Users] AGI to read MySQL   Lewis Agosta
• [Asterisk-Users] WIFI sip phone   Josep Aguilar
• [Asterisk-Users] Re: Sangoma A101 configuration   Kamran Ahmad
• [Asterisk-Users] Re: Sangoma A101 configuration   Kamran Ahmad
• [Asterisk-Users] Converting .wav to .WAV   Akpome Akpoguma
• [Asterisk-Users] Converting .wav to .WAV   Akpome Akpoguma
• [Asterisk-Users] Optimal Hardware   Akpome Akpoguma
• [Asterisk-Users] VOCAL + Asterisk   Akpome Akpoguma
• [Asterisk-Users] [REPOST] Asterisk Realtime and "Ex-Girlfriend"   Alban
• [Asterisk-Users] Asterisk behind dynamic IP   Tim Alder
• [Asterisk-Users] meetme public   Pablo Allietti
• [Asterisk-Users] Re: meetme public   Pablo Allietti
• [Asterisk-Users] g729 or another   Pablo Allietti
• [Asterisk-Users] Linksys SPA-941 NAT?   Pablo Allietti
• [Asterisk-Users] Re: Linksys SPA-941 NAT?   Pablo Allietti
• [Asterisk-Users] Re: g729 or another   Pablo Allietti
• [Asterisk-Users] Creating Queues on Asterisk server - Sending ingress calls off-net to either PSTN or another VoIP application - thoughts?   Christopher Aloi
• Fwd: FW: [Asterisk-Users] Creating Queues on Asterisk server - Sendingingress calls off-net to either PSTN or another VoIPapplication - thoughts?   Christopher Aloi
• [Asterisk-Users] Asterisk queue log solution?   Christopher Aloi
• [Asterisk-Users] Asterisk queue log solution?   Christopher Aloi
• [Asterisk-Users] Thoughts on building a Voicemail only Asterisk server?   Christopher Aloi
• [Asterisk-Users] Help with incoming SIP routing   Christopher Aloi
• [Asterisk-Users] Help with incoming SIP routing   Christopher Aloi
• [Asterisk-Users] Cisco Gateway   Carlos Alperin
• [Asterisk-Users] Sip t38 gateway tests   Carlos Alperin
• [Asterisk-Users] Cisco Gateway   Carlos Alperin
• [Asterisk-Users] how to hang the zap channel   Carlos Alperin
• [Asterisk-Users] Strange Zaptel issue   Carlos Alperin
• [Asterisk-Users] Sip t38 gateway tests   Carlos Alperin
• [Asterisk-Users] SE Michigan asterisk users group   Carlos Alperin
• [Asterisk-Users] SE Michigan asterisk users group   Carlos Alperin
• [Asterisk-Users] SE Michigan asterisk users group   Carlos Alperin
• [Asterisk-Users] SE Michigan asterisk users group   Carlos Alperin
• [Asterisk-Users] Re: SE Michigan asterisk users group   Carlos Alperin
• [Asterisk-Users] Mail loop?   Carlos Alperin
• [Asterisk-Users] Troncal SIP   Aldo Alexander Leyva Alvarado
• [Asterisk-Users] Polycom Buddies in 1.6.6   Lucas Alvarez
• [Asterisk-Users] asttapi 0.10   Victor Alvarez
• [Asterisk-Users] Polycom-Asterisk hints/presence   Bob Amen
• [Asterisk-Users] Re: Prices of g729 codec   Benny Amorsen
• [Asterisk-Users] Re: Can this config sustain 30 users?   Benny Amorsen
• [Asterisk-Users] Re: Can this config sustain 30 users?   Benny Amorsen
• [Asterisk-Users] Cisco 7936 Conference Phone - SIP or SCCP?   Ryan Amos
• [Asterisk-Users] 7960 help: transferring calls   Ryan Amos
• [Asterisk-Users] Cisco 7905G SIP firmware needed   Ryan Amos
• [Asterisk-Users] HDI remove a key from the Asterisk database with a <null> key, but a value?   Colin Anderson
• [Asterisk-Users] Optimal Hardware   Colin Anderson
• [Asterisk-Users] Asterisk: T1 hunt group setup   Colin Anderson
• [Asterisk-Users] syslog server   Colin Anderson
• [Asterisk-Users] OT: Cellular boosters   Colin Anderson
• [Asterisk-Users] early session audio on zap channel   Colin Anderson
• [Asterisk-Users] [CAVPDiscussion] OT: BT to replace legacy tele com infrastructure with open, standards-based VoIP switches   Colin Anderson
• [Asterisk-Users] Quad T1 Card   Colin Anderson
• [Asterisk-Users] quad t1 / 1U rack server combos   Colin Anderson
• [Asterisk-Users] click to call features on asterisk   Colin Anderson
• [Asterisk-Users] PRI & Fax Passthrough   Colin Anderson
• [Asterisk-Users] Re: Audio problems on Zap & SIP, local netwo rk, not IRQ related?   Colin Anderson
• [Asterisk-Users] Fun with Echo   Colin Anderson
• [Asterisk-Users] IAX DID channels as incoming hunt group?   Colin Anderson
• [Asterisk-Users] IAX DID channels as incoming hunt group?   Colin Anderson
• [Asterisk-Users] TDM-400 and dialplan -- how to ring a SIP ex tension *before* answering the PSTN line?   Colin Anderson
• [Asterisk-Users] IAX2 Vs SIP cpu load   Colin Anderson
• [Asterisk-Users] echo sidetone grandstream and tdm400p   Colin Anderson
• [Asterisk-Users] Can this config sustain 30 users?   Colin Anderson
• [Asterisk-Users] IAX2 Vs SIP cpu load   Colin Anderson
• [Asterisk-Users] Festival RPM?   Colin Anderson
• [Asterisk-Users] IAX2 Vs SIP cpu load   Colin Anderson
• [Asterisk-Users] Can this config sustain 30 users?   Colin Anderson
• [Asterisk-Users] Can this config sustain 30 users?   Colin Anderson
• [Asterisk-Users] Hard drive write cache   Colin Anderson
• [Asterisk-Users] Hard drive write cache   Colin Anderson
• [Asterisk-Users] Hard drive write cache   Colin Anderson
• [Asterisk-Users] 100 lines PBX + system config - repost   Colin Anderson
• [Asterisk-Users] Easiest (best?) linux distribution for dedic atedAsterisk box?   Colin Anderson
• [Asterisk-Users] Best $300 VoIP phone for asterisk?   Colin Anderson
• [Asterisk-Users] Anyone see this?   Colin Anderson
• [Asterisk-Users] EC needed in all-digital situation?   Colin Anderson
• [Asterisk-Users] Gumstix!   Colin Anderson
• [Asterisk-Users] what are the elements of a good asterisk set up?   Colin Anderson
• [Asterisk-Users] IAX2 Vs SIP cpu load   Colin Anderson
• [Asterisk-Users] finding mac addresses   Colin Anderson
• [Asterisk-Users] Re: faxdetect questions - Please HELP!   Colin Anderson
• [Asterisk-Users] Snom 360 Passsword Issue   Colin Anderson
• [Asterisk-Users] Re: faxdetect questions - Please HELP!   Colin Anderson
• [Asterisk-Users] Re: User Loses Ability to Make Outgoing Call s   Colin Anderson
• [Asterisk-Users] Re: faxdetect questions - Please HELP!   Colin Anderson
• [Asterisk-Users] Re: faxdetect questions - Please HELP!   Colin Anderson
• [Asterisk-Users] Out of Office Auto Reply:   Colin Anderson
• [Asterisk-Users] Re: Out of Office Auto Reply:   Colin Anderson
• [Asterisk-Users] troubleshooting echo on speakerphone   Colin Anderson
• [Asterisk-Users] troubleshooting echo on speakerphone   Colin Anderson
• [Asterisk-Users] RE: Voice calls sent to fax extension   Colin Anderson
• [Asterisk-Users] RE: Voice calls sent to fax extension   Colin Anderson
• [Asterisk-Users] RE: Voice calls sent to fax extension   Colin Anderson
• [Asterisk-Users] asterisk to mobile phone   Colin Anderson
• [Asterisk-Users] asterisk to mobile phone   Colin Anderson
• [Asterisk-Users] Most stable Asterisk version   Colin Anderson
• [Asterisk-Users] need help troubleshooting clipping and garbl ed VOIP calls   Colin Anderson
• [Asterisk-Users] Linksys SRW224P POE Switch   Andres
• [Asterisk-Users] Re: Linksys SRW224P POE Switch   Andres
• [Asterisk-Users] shutting down a mysql server renders cdr_mysqldead and asterisk nolonger makes or receives calls   Andres
• [Asterisk-Users] Cisco router and "488 Not acceptable here" messages   Andres
• [Asterisk-Users] SPA941 and Echo   Andres
• [Asterisk-Users] SPA941 and Echo   Andres
• [Asterisk-Users] SPA941 and Echo   Andres
• [Asterisk-Users] Wanted: CISCO 186 ATAs   Cory Andrews
• [Asterisk-Users] Prices of g729 codec   Cory Andrews
• [Asterisk-Users] Customer's voice not compatible with service?   Cory Andrews
• [Asterisk-Users] Re: Zork and Asterisk   Cory Andrews
• [Asterisk-Users] CISCO POE   Cory Andrews
• [Asterisk-Users] Analog Line "Static" and Low Volume   Cory Andrews
• [Asterisk-Users] Linksys SPA-941 NAT?   Cory Andrews
• [Asterisk-Users] Linksys SRW224P POE Switch   Cory Andrews
• [Asterisk-Users] MBX Servers?   Cory Andrews
• [Asterisk-Users] Single T1 card with Echo Cancellation toworkwithDell?   Cory Andrews
• [Asterisk-Users] Best $300 VoIP phone for asterisk?   Cory Andrews
• [Asterisk-Users] Anyone using VoIP WiFi phones?   Cory Andrews
• [Asterisk-Users] Anyone using VoIP WiFi phones?   Cory Andrews
• [Asterisk-Users] USB handset options for softphones   Cory Andrews
• [Asterisk-Users] Suggested Phone   Cory Andrews
• [Asterisk-Users] Digium Hardware Reliability   Cory Andrews
• [Asterisk-Users] Digium Hardware Reliability   Cory Andrews
• [Asterisk-Users] h323 with asterisk problem   Thameem Ansari
• [Asterisk-Users] h323 with asterisk problem   Thameem Ansari
• [Asterisk-Users] h323 with asterisk problem   Thameem Ansari
• [Asterisk-Users] h323 with asterisk problem   Thameem Ansari
• [Asterisk-Users] h323 with asterisk problem   Thameem Ansari
• [Asterisk-Users] h323 with asterisk problem   Thameem Ansari
• [Asterisk-Users] Auto dialer   Thameem Ansari
• [Asterisk-Users] Re: Can I enter an extension to dial while voicemail is playing?   Peter Antonacci
• [Asterisk-Users] best hardphone for Asterisk?   Peter Antonacci
• [Asterisk-Users] Multiple processes   Lee Archer
• [Asterisk-Users] Duplicate asterisk processes   Lee Archer
• [Asterisk-Users] Compiling mpg123 under 1.2.9.1 / Ubuntu 6.06   Lee Archer
• [Asterisk-Users] How many TE405 ...   Ard
• [Asterisk-Users] I guess my server capacity is ok   Goke Aruna
• [Asterisk-Users] Attended call transfer with GXP-2000   Lacy Moore - Aspendora
• [Asterisk-Users] No CID on ZAP   Lacy Moore - Aspendora
• [Asterisk-Users] Cell gateway for T-Mobile US??   Lacy Moore - Aspendora
• [Asterisk-Users] Cisco 7960 BLA   Lacy Moore - Aspendora
• [Asterisk-Users] Linksys SRW224P POE Switch   Lacy Moore - Aspendora
• [Asterisk-Users] Auto-pickup cisco phones   Lacy Moore - Aspendora
• [Asterisk-Users] Cisco 7936 Conference Phone - SIP or SCCP?   Lacy Moore - Aspendora
• [Asterisk-Users] Best $300 VoIP phone for asterisk?   Lacy Moore - Aspendora
• [Asterisk-Users] Linksys SRW224P POE Switch   Lacy Moore - Aspendora
• [Asterisk-Users] Problem trying to SayDigits when an invalid extension is dialed   Lacy Moore - Aspendora
• [Asterisk-Users] Using HINT with Cisco 7960/SIP   Lacy Moore - Aspendora
• [Asterisk-Users] Which phones are good, or at least acceptable, for home and office   Lacy Moore - Aspendora
• [Asterisk-Users] Which phones are good, or at least acceptable, for home and office   Lacy Moore - Aspendora
• [Asterisk-Users] 7960 help: transferring calls   Lacy Moore - Aspendora
• [Asterisk-Users] SIP, Microsoft RTC, and Originate problem   Asterisk
• [Asterisk-Users] SIP, Microsoft RTC, and Originate problem   Asterisk
• [Asterisk-Users] SIP, Microsoft RTC, and Originate problem   Asterisk
• [Asterisk-Users] Transfer call via AMI or dialplan   Asterisk
• [Asterisk-Users] Strange problem with MusicOnHold - works outgoing - works with extension - but not incoming!   Yoja Asterisk
• [Asterisk-Users] How can I use my regular phones with Asterisk running on my Linksys WRT54G router?   Jonathan Attwood
• [Asterisk-Users] How can I use my regular phones with Asterisk running on my Linksys WRT54G router?   Jonathan Attwood
• [Asterisk-Users] No incoming sip calls   Jonathan Attwood
• [Asterisk-Users] Can I get caller id passed to a phone connected to a Supura 2100?   Jonathan Attwood
• [Asterisk-Users] NCS patch   Giedrius Augys
• [Asterisk-Users] CDRTool   Giedrius Augys
• [Asterisk-Users] CDRTool   Giedrius Augys
• [Asterisk-Users] AudioCodes MP-124   Giedrius Augys
• [Asterisk-Users] Change g729 payload   Dan Austin
• [Asterisk-Users] fax   JD Austin
• [Asterisk-Users] FW: TTS from MySQL   Walid Azab
• [Asterisk-Users] TTS to read from Database   Walid Azab
• [Asterisk-Users] AGI to read MySQL   Walid Azab
• [Asterisk-Users] MWI not working   Walid Azab
• [Asterisk-Users] AGI to read MySQL   Walid Azab
• [Asterisk-Users] IVR Applications   Walid Azab
• [Asterisk-Users] hanging up call after launching a script, script should continue independently   Christian B
• [Asterisk-Users] app_queue and Real roundrobin   Michiel van Baak
• [Asterisk-Users] Re: XML to monitor queues on Cisco display ?   Michiel van Baak
• [Asterisk-Users] Config Revision Control   Michiel van Baak
• [Asterisk-Users] Config Revision Control   Michiel van Baak
• [Asterisk-Users] Config Revision Control   Michiel van Baak
• [Asterisk-Users] Change in dial command behaviour between 1.2.7.1 and 1.2.8?   Michiel van Baak
• [Asterisk-Users] syslog server   Michiel van Baak
• [Asterisk-Users] Web UI - Best practices?   Michiel van Baak
• [Asterisk-Users] Voicemail with NFS   Michiel van Baak
• [Asterisk-Users] Bristuff-0.3.0-PRE-1q and & florz patch compile trouble   Michiel van Baak
• [Asterisk-Users] Anyone using VoIP WiFi phones?   Michiel van Baak
• [Asterisk-Users] zapata.conf: recent changes?   Michiel van Baak
• [Asterisk-Users] periodic-announce not working   Michiel van Baak
• SV: [Asterisk-Users] periodic-announce not working   Michiel van Baak
• [Asterisk-Users] Realtime problem   Michiel van Baak
• SV: [Asterisk-Users] periodic-announce not working   Michiel van Baak
• SV: [Asterisk-Users] periodic-announce not working   Michiel van Baak
• SV: [Asterisk-Users] periodic-announce not working   Michiel van Baak
• [Asterisk-Users] asterisk sip listening port   Michiel van Baak
• [Asterisk-Users] This is getting really annoying - re: POSTFIX   Michiel van Baak
• [Asterisk-Users] Call length limitation   Michiel van Baak
• [Asterisk-Users] Realtime Voicemail   Michiel van Baak
• [Asterisk-Users] Realtime Voicemail   Michiel van Baak
• [Asterisk-Users] getting agentID and DNID help   Michiel van Baak
• [Asterisk-Users] Trixbox maunual configuration   Michiel van Baak
• [Asterisk-Users] Remote employees using Polycom 501 lose ability to receive incoming calls after few minutes.   Michiel van Baak
• [Asterisk-Users] AEL #include   Jason Bachman
• [Asterisk-Users] asterisk behind cisco pix 506   Jason Bachman
• [Asterisk-Users] asterisk behind cisco pix 506   Jason Bachman
• [Asterisk-Users] asterisk behind cisco pix 506   Jason Bachman
• [Asterisk-Users] asterisk behind cisco pix 506   Jason Bachman
• [Asterisk-Users] Local vs. toll Dial Plan   Jason Bachman
• [Asterisk-Users] Broken firewall or brain damaged admin?   Jason Bachman
• [Asterisk-Users] grandstream GXV-3000   Rod Bacon
• [Asterisk-Users] AGI and Video   Rod Bacon
• [Asterisk-Users] Chanspy Jitter?   Wes Baehr
• [Asterisk-Users] Random Zap Channel Drops to SIP   Wes Baehr
• [Asterisk-Users] Hitting * in a queue call hangs up?   Wes Baehr
• [Asterisk-Users] Prices of g729 codec   Chris Bagnall
• [Asterisk-Users] Queues with really short timeouts   Chris Bagnall
• [Asterisk-Users] Call accounting where calls cross charge zones (code fragment request)   Chris Bagnall
• [Asterisk-Users] 7960 help: transferring calls   Chris Bagnall
• [Asterisk-Users] How to set overlap dial timeout in bristuff zaptel?   Rostislav Bagrov
• [Asterisk-Users] PABX Setup   Boris Bakchiev
• [Asterisk-Users] Hard drive write cache   Boris Bakchiev
• [Asterisk-Users] Hard drive write cache   Boris Bakchiev
• [Asterisk-Users] Will 200KB/s drive access be OK for voicemailstorage?   Boris Bakchiev
• [Asterisk-Users] FAX + Digium + SpanDSP   Boris Bakchiev
• [Asterisk-Users] TE420P/TE415P?   Boris Bakchiev
• [Asterisk-Users] Vonage and FXO   Padmanaban Balasubramaniam
• [Asterisk-Users] Vonage and FXO   Padmanaban Balasubramaniam
• [Asterisk-Users] HELP!!!! Weird TDM2406E unable to bridge all outgoing calls.   Time Bandit
• [Asterisk-Users] Running a poll server with asterisk   Time Bandit
• [Asterisk-Users] TDM-400 and dialplan -- how to ring a SIP extension *before* answering the PSTN line?   Time Bandit
• [Asterisk-Users] TDM-400 and dialplan -- how to ring a SIP extension *before* answering the PSTN line?   Time Bandit
• [Asterisk-Users] Executing a Function from AGI   Time Bandit
• [Asterisk-Users] database copy in asterisk   Time Bandit
• [Asterisk-Users] SIP or IAX client written in C   Time Bandit
• [Asterisk-Users] Re: DELL PowerEdge 2850 and TE4110P and TE110P   Remco Barendse
• [Asterisk-Users] TDM11B FXS port stops working after a reload?   Remco Barendse
• [Asterisk-Users] Example config files for Snom mass updating?   Remco Barendse
• [Asterisk-Users] Change in dial command behaviour between 1.2.7.1 and 1.2.8?   Remco Barendse
• [Asterisk-Users] Change in dial command behaviour between 1.2.7.1 and 1.2.8?   Remco Barendse
• [Asterisk-Users] Change in dial command behaviour between 1.2.7.1 and 1.2.8?   Remco Barendse
• [Asterisk-Users] Change in dial command behaviour between 1.2.7.1 and 1.2.8?   Remco Barendse
• [Asterisk-Users] Change in dial command behaviour between 1.2.7.1 and 1.2.8?   Remco Barendse
• [Asterisk-Users] PRI & Fax Passthrough   Remco Barendse
• [Asterisk-Users] /var/log/asterisk/full ?   Remco Barendse
• [Asterisk-Users] Bristuff-0.3.0-PRE-1q and & florz patch compile trouble   Remco Barendse
• [Asterisk-Users] Bristuff-0.3.0-PRE-1q and & florz patch compile trouble   Remco Barendse
• [Asterisk-Users] bristuff chan_zap.c zt_pri_error line errors?   Remco Barendse
• [Asterisk-Users] best hardphone for Asterisk?   Iain Barker
• [Asterisk-Users] GXP-2000   Erick Baum
• [Asterisk-Users] Re: GXP-2000 (steer clear)   Erick Baum
• [Asterisk-Users] GXP-2000   Erick Baum
• [Asterisk-Users] voip to voip bridge   Erick Baum
• [Asterisk-Users] Sangoma card A101 Card troubles.   Benjamin J. Bawkon
• [Asterisk-Users] Include Text file in Dial Plan   Forrest Beck
• [Asterisk-Users] Suggested Phone   Forrest Beck
• [Asterisk-Users] WebPhone   Forrest Beck
• [Asterisk-Users] AGI: Dial and Recording my own CDR   Peter Beckman
• [Asterisk-Users] AGI: Dial and Recording my own CDR   Peter Beckman
• [Asterisk-Users] DTMF Tones not coming in clear   Peter Beckman
• [Asterisk-Users] Asterisk exit on startup   Nathan Bell
• [Asterisk-Users] Asterisk exit on startup   Nathan Bell
• [Asterisk-Users] T1 passthrough/middleman   Nathan Bell
• [Asterisk-Users] T1 passthrough/middleman   Nathan Bell
• [Asterisk-Users] Quad T1 Card   Nic Bellamy
• [Asterisk-Users] unknown host cvs.digium.com   Andrea Bencini
• [Asterisk-Users] RSA Signature (key ***) failed   Michele Bendazzoli
• [Asterisk-Users] RSA Signature (key ***) failed   Michele Bendazzoli
• [Asterisk-Users] SkypeOUT proxy   Dovid Bender
• [Asterisk-Users] free sun boxes   Dovid Bender
• [Asterisk-Users] Broadvoice - Last Straw!   Dovid Bender
• [Asterisk-Users] Meetme Dumping Call's   Dovid Bender
• [Asterisk-Users] Meetme   Dovid Bender
• [Asterisk-Users] transferring calls from ekiga to asterisk   don Paolo Benvenuto
• [Asterisk-Users] transferring calls from ekiga to asterisk   don Paolo Benvenuto
• [Asterisk-Users] transferring calls from ekiga to asterisk   don Paolo Benvenuto
• [Asterisk-Users] Changing standard Voicemail behavior   Jan Berggren
• [Asterisk-Users] revisit to legacy PBX and CID over PRI   BerkHolz, Steven
• [Asterisk-Users] Cell gateway for T-Mobile US??   BerkHolz, Steven
• [Asterisk-Users] Re: CallerID name inbound from PRI   BerkHolz, Steven
• [Asterisk-Users] calleridname.agi patch to only overwrite name if it is missing   BerkHolz, Steven
• [Asterisk-Users] Backup Question?   BerkHolz, Steven
• [Asterisk-Users] me, voip.trxtel.com and early media   BerkHolz, Steven
• [Asterisk-Users] SE Michigan asterisk users group   BerkHolz, Steven
• [Asterisk-Users] metermaid patch   BerkHolz, Steven
• [Asterisk-Users] Asterisk: T1 hunt group setup   John Bigelow
• [Asterisk-Users] "Reserving" a conference room   Eric Bishop
• [Asterisk-Users] "sip show inuse" is useless!   Eric Bishop
• [Asterisk-Users] "sip show inuse" is useless!   Eric Bishop
• [Asterisk-Users] Grandstream BT101/102 lost register with asterisk ?   Gareth Blades
• [Asterisk-Users] Playback welcome message while phones ring, please help   Gareth Blades
• [Asterisk-Users] GXP-2000   Gareth Blades
• [Asterisk-Users] GXP-2000 MultiPurpose Keys   Gareth Blades
• [Asterisk-Users] GXP-2000 MultiPurpose Keys   Gareth Blades
• [Asterisk-Users] PRI Broke on 1.2.9.1?   Gareth Blades
• [Asterisk-Users] GXP-2000 Audio Quality   Gareth Blades
• [Asterisk-Users] GXP-2000 1.1.0.13 Issues   Gareth Blades
• [Asterisk-Users] GXP-2000 and Configdownload via TFTP   Gareth Blades
• [Asterisk-Users] GXP-2000 addressbook   Gareth Blades
• [Asterisk-Users] IAX FXS.. Any experience with...   Gareth Blades
• [Asterisk-Users] x100p buying advice   Gareth Blades
• [Asterisk-Users] GXP-2000 MultiPurpose Keys   Phil Blundell
• [Asterisk-Users] Signaling and media   Kevin Bockman
• [Asterisk-Users] Thoughts on building a Voicemail only Asteriskserver?   William Boehlke
• [Asterisk-Users] SIP voice recorder   Leo Ann Boon
• [Asterisk-Users] Small form factor system w/PCI slot   Leo Ann Boon
• [Asterisk-Users] h323 with asterisk problem   Leo Ann Boon
• [Asterisk-Users] TBCT - Two B-Channel Transfer   Leo Ann Boon
• [Asterisk-Users] Asterisk & Eyebeam chat function   Leo Ann Boon
• [Asterisk-Users] open source sip softphone (Window OS version )   Leo Ann Boon
• [Asterisk-Users] TigerJet PCI PPG FXO Card   Leo Ann Boon
• [Asterisk-Users] WRTG54GS Capacity   Leo Ann Boon
• [Asterisk-Users] DTMF when using g.729   Leo Ann Boon
• [Asterisk-Users] Soekris net4801 and IAXy dhcp issue   Leo Ann Boon
• [Asterisk-Users] zaptel.conf settings for Singtel ISDN-2   Leo Ann Boon
• [Asterisk-Users] Config Revision Control   Casey Boone
• [Asterisk-Users] finding mac addresses   Casey Boone
• [Asterisk-Users] TDM PCI Master Abort   Stephen Bosch
• [Asterisk-Users] TDM PCI Master Abort   Stephen Bosch
• [Asterisk-Users] TDM PCI Master Abort   Stephen Bosch
• [Asterisk-Users] Configuring Polycom 501 IP phones via the console   Stephen Bosch
• [Asterisk-Users] Configuring Polycom 501 IP phones via the console   Stephen Bosch
• [Asterisk-Users] Configuring Polycom 501 IP phones via the console   Stephen Bosch
• [Asterisk-Users] TDM-400 doesn't detect far-end hangup   Stephen Bosch
• [Asterisk-Users] IAX DID channels as incoming hunt group?   Stephen Bosch
• [Asterisk-Users] IAX DID channels as incoming hunt group?   Stephen Bosch
• [Asterisk-Users] IAX DID channels as incoming hunt group?   Stephen Bosch
• [Asterisk-Users] TDM-400 and dialplan -- how to ring a SIP extension *before* answering the PSTN line?   Stephen Bosch
• [Asterisk-Users] IAX DID channels as incoming hunt group?   Stephen Bosch
• [Asterisk-Users] TDM-400 and dialplan -- how to ring a SIP ex tension *before* answering the PSTN line?   Stephen Bosch
• [Asterisk-Users] IAX DID channels as incoming hunt group?   Stephen Bosch
• [Asterisk-Users] TDM-400 and dialplan -- how to ring a SIP extension *before* answering the PSTN line?   Stephen Bosch
• [Asterisk-Users] TTS engine query   Stephen Bosch
• [Asterisk-Users] Unable to connect to Asterisk? (simple[?] question)   Peter Bowyer
• [Asterisk-Users] conference   Peter Bowyer
• [Asterisk-Users] Unable to connect to Asterisk? (simple[?] question)   Peter Bowyer
• [Asterisk-Users] MWI not working   Peter Bowyer
• [Asterisk-Users] Caller ID Matching in extensions.conf   Peter Bowyer
• [Asterisk-Users] Mail loop?   Peter Bowyer
• [Asterisk-Users] Reception softphone suggestions?   Crazy Boy
• [Asterisk-Users] What is Echo?   Crazy Boy
• [Asterisk-Users] What is Echo?   Crazy Boy
• [Asterisk-Users] How much bandwidth needed?   Crazy Boy
• [Asterisk-Users] How much bandwidth needed?   Crazy Boy
• [Asterisk-Users] best hardphone for Asterisk?   Crazy Boy
• [Asterisk-Users] PRI Issue - Calls being rejected with unacceptable channel   Andy Brezinsky
• [Asterisk-Users] pix 501   John Bridgewater
• [Asterisk-Users] Dial(ZAP with t option for call transfer via *2)   Bruno.Voigt at ic3s.de
• [Asterisk-Users] Sip error messages   Neil Bullock
• [Asterisk-Users] Nokia N80 and asterisk?   Nick Burch
• [Asterisk-Users] Custom Extension halting execution upon caller hanging up   Alexander Burke
• [Asterisk-Users] Custom extension halting execution upon caller hanging up   Alexander Burke
• [Asterisk-Users] Is anybody using XEN in conjunction with Asteriskand/or Openser?   Ryan Burke
• [Asterisk-Users] Integrating Asterisk   Dakota Burns
• [Asterisk-Users] Bullet-proof System   Dakota Burns
• [Asterisk-Users] Recommended Web Interface   Dakota Burns
• [Asterisk-Users] Integrating Asterisk   Dakota Burns
• [Asterisk-Users] Single T1 card with Echo Cancellation to work withDell?   Shane Burrell
• [Asterisk-Users] X100P fails to initialize   Lachek Butalek
• [Asterisk-Users] MWI on the PA168V in IAX mode?   Lachek Butalek
• [Asterisk-Users] MWI on the PA168V in IAX mode?   Lachek Butalek
• [Asterisk-Users] FreePBX 2.1.0: Manually rewriting extensions_additional.conf   Lachek Butalek
• [Asterisk-Users] FreePBX 2.1.0: Manually rewriting extensions_additional.conf   Lachek Butalek
• [Asterisk-Users] Easiest (best?) linux distribution for dedicated Asterisk box?   Lachek Butalek
• [Asterisk-Users] Trying to find good VOIP provider.   Lachek Butalek
• [Asterisk-Users] Playback welcome message while phones ring, please help   Tommaso Calosi
• [Asterisk-Users] Playback welcome message while phones ring, please help   Tommaso Calosi
• [Asterisk-Users] SPA-941 Disable call waiting or Disable Call waiting via asterisk   Tommaso Calosi
• [Asterisk-Users] SPA-941 Disable call waiting or Disable Call waiting via asterisk   Tommaso Calosi
• [Asterisk-Users] Call limit function on sip channel to external pop   Tommaso Calosi
• [Asterisk-Users] Snom 360 Passsword Issue   Tommaso Calosi
• [Asterisk-Users] New Digium Card b410p   Tommaso Calosi
• [Asterisk-Users] Bearer capabilities on PRI   Greg Camp
• [Asterisk-Users] Meetme versus app_conference   Brian Capouch
• [Asterisk-Users] New York Times article on VoIP Hacker   Brian Capouch
• [Asterisk-Users] how to identify agi crash cause   Brian Capouch
• [Asterisk-Users] RE: IAX Passing Variables   Brian Capouch
• [Asterisk-Users] Broken firewall or brain damaged admin?   Brian Capouch
• [Asterisk-Users] Surprise!!! New sound files auto-downloaded to my system   Brian Capouch
• [Asterisk-Users] Gumstix!   Brian Capouch
• [Asterisk-Users] Voicemail with NFS   Brian Capouch
• [Asterisk-Users] Voicemail with NFS   Brian Capouch
• [Asterisk-Users] Voicemail with NFS   Brian Capouch
• [Asterisk-Users] Will 200KB/s drive access be OK for voicemailstorage?   Brian Capouch
• [Asterisk-Users] Voicemail with NFS   Brian Capouch
• [Asterisk-Users] What ever happened to the LTAPI, the Linux Telephony API?   Brian Capouch
• [Asterisk-Users] Anyone using VoIP WiFi phones?   Brian Capouch
• [Asterisk-Users] syntax error   Brian Capouch
• [Asterisk-Users] Don't use CDRTool From AG-projescts   Brian Capouch
• [Asterisk-Users] Upgrading old version of Asteriak - changes   Brian Capouch
• [Asterisk-Users] Oh oh. Micro$oft just noticed VoIP   Brian Capouch
• [Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!   Brian Capouch
• [Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!   Brian Capouch
• [Asterisk-Users] Error in config sample for GoToIf?   Brian Capouch
• [Asterisk-Users] Error in config sample for GoToIf?   Brian Capouch
• SV: [Asterisk-Users] Error in config sample for GoToIf?   Brian Capouch
• [Asterisk-Users] Re: Digium Hardware Reliability   Brian Capouch
• [Asterisk-Users] ASTCC: How to reset periodically all "card in use" flag back?   JP Carballo
• [Asterisk-Users] ASTCC: How to reset periodically all "card in use" flag back?   JP Carballo
• [Asterisk-Users] ASTCC: How to reset periodically all "card in use" flag back?   JP Carballo
• [Asterisk-Users] ASTCC: How to reset periodically all "card in use" flag back?   JP Carballo
• [Asterisk-Users] ASTCC: customer wants 100 accounts   JP Carballo
• [Asterisk-Users] ASTCC: customer wants 100 accounts   JP Carballo
• [Asterisk-Users] ASTCC: customer wants 100 accounts   JP Carballo
• [Asterisk-Users] Bearer capabilities on PRI   Who Carez?
• [Asterisk-Users] Bearer capabilities on PRI [LOOKING FOR PRI expert to resolve the issue - for hire]   Who Carez?
• [Asterisk-Users] Voicemail records nonsense, but record() works (??)   Strom Carlson
• [Asterisk-Users] Problem with VoicemailMain   Ricardo Carvalho
• [Asterisk-Users] Asterisk keeps running after hungup untill I press #   Ricardo Carvalho
• [Asterisk-Users] Asking for phone number to dial   Anthony Cennami
• [Asterisk-Users] Asking for phone number to dial   Anthony Cennami
• [Asterisk-Users] Asking for phone number to dial   Anthony Cennami
• [Asterisk-Users] Include Text file in Dial Plan   Anthony Cennami
• [Asterisk-Users] sip to h323 gateway ...   Cesc
• [Asterisk-Users] open source sip softphone (Window OS version )   Cesc
• [Asterisk-Users] sip to h323 gateway ...   Cesc
• [Asterisk-Users] sip to h323 gateway ...   Cesc
• [Asterisk-Users] sip to h323 ... direct RTP?   Cesc
• [Asterisk-Users] sip to h323 ... direct RTP?   Cesc
• [Asterisk-Users] Conferencing with multiple servers   Cesc
• [Asterisk-Users] Asterisk h323   Cesc
• [Asterisk-Users] H.323 soft phone known to be run with asterisk.   Cesc
• [Asterisk-Users] ADSL modem, TDM400P, zaptel and not hanging up   Nick Chalk
• [Asterisk-Users] ADSL modem, TDM400P, zaptel and not hanging up   Nick Chalk
• [Asterisk-Users] Asterisk-Addons compile problem (cdr_addon_mysql.c)   Ken Chan
• [Asterisk-Users] direct a call to a busy channel   Krishna Sumanth Chava
• [Asterisk-Users] MFC/R2 for Voice and Data   Carlos Chavez
• [Asterisk-Users] How many TE405 ...   Carlos Chavez
• [Asterisk-Users] Block access to number@domain.com   Carlos Chavez
• [Asterisk-Users] Best settings for Unicall and Fax   Carlos Chavez
• [Asterisk-Users] voice mail   Khaled Chehab
• [Asterisk-Users] weather   Khaled Chehab
• [Asterisk-Users] FW: voice mail   Khaled Chehab
• [Asterisk-Users] directory   Khaled Chehab
• [Asterisk-Users] extensions problem   Khaled Chehab
• [Asterisk-Users] key pads   Khaled Chehab
• [Asterisk-Users] fax   Khaled Chehab
• [Asterisk-Users] conference   Khaled Chehab
• [Asterisk-Users] conference   Khaled Chehab
• [Asterisk-Users] FW: conference   Khaled Chehab
• [Asterisk-Users] conference   Khaled Chehab
• [Asterisk-Users] asterisk auto conference   Khaled Chehab
• [Asterisk-Users] asterisk auto conference   Khaled Chehab
• [Asterisk-Users] asterisk auto conference   Khaled Chehab
• [Asterisk-Users] directory   Khaled Chehab
• [Asterisk-Users] Update   Khaled Chehab
• [Asterisk-Users] Digital Receptionist   Khaled Chehab
• [Asterisk-Users] asterisk auto conference   Khaled Chehab
• [Asterisk-Users] Asterisk h323   Khaled Chehab
• [Asterisk-Users] database space   Khaled Chehab
• [Asterisk-Users] Asterisk h323   Khaled Chehab
• [Asterisk-Users] database space   Khaled Chehab
• [Asterisk-Users] billing   Khaled Chehab
• [Asterisk-Users] fax   Khaled Chehab
• [Asterisk-Users] asterisk sip listening port   Khaled Chehab
• [Asterisk-Users] database space   Khaled Chehab
• [Asterisk-Users] voicemail number of recorded messages   Khaled Chehab
• [Asterisk-Users] voicemail number of recorded messages   Khaled Chehab
• [Asterisk-Users] a2billing   Khaled Chehab
• [Asterisk-Users] (no subject)   Khaled Chehab
• [Asterisk-Users] Voicemail   Khaled Chehab
• [Asterisk-Users] directory   Khaled Chehab
• [Asterisk-Users] Please Help - Polycom IP 601 Buddy Watch problems   Ben Chennat
• [Asterisk-Users] '500 Internal Server' Error on SIP NOTIFY   Ben Chennat
• [Asterisk-Users] Canreinvite   Neil Cherry
• [Asterisk-Users] F3000 registering to asterisk   Neil Cherry
• [Asterisk-Users] F3000 registering to asterisk   Neil Cherry
• [Asterisk-Users] Wanted: CISCO 186 ATAs   James Ching
• [Asterisk-Users] audio streaming points different with VRRP   Bob Chiodini
• [Asterisk-Users] very slow network from GXP-2000 switch port   Bob Chiodini
• [Asterisk-Users] OLD PA system.   Bob Chiodini
• [Asterisk-Users] OLD PA system.   Bob Chiodini
• [Asterisk-Users] OLD PA system.   Bob Chiodini
• [Asterisk-Users] This is getting really annoying - re: POSTFIX   Bob Chiodini
• [Asterisk-Users] David Choo/eServices/eSpore is overseas   David Choo
• [Asterisk-Users] Asterisk with Sipbroker calling / routing problem   Mathieu Chouquet-Stringer
• [Asterisk-Users] Supporter needed   Soren Christensen
• [Asterisk-Users] Fun with Echo   Dr. Michael J. Chudobiak
• [Asterisk-Users] Snom 360 doesn't register after reboot   Dr. Michael J. Chudobiak
• [Asterisk-Users] Snom 360 with Firmware 6.1?   Dr. Michael J. Chudobiak
• [Asterisk-Users] call quality statistics?   Dr. Michael J. Chudobiak
• [Asterisk-Users] Remote employees using Polycom 501 lose ability to receive incoming calls after few minutes.   Dr. Michael J. Chudobiak
• [Asterisk-Users] Remote employees using Polycom 501 lose ability to receive incoming calls after few minutes.   Dr. Michael J. Chudobiak
• [Asterisk-Users] sound quality problem on mISDN   Piotr Chytla
• [Asterisk-Users] sound quality problem on mISDN   Piotr Chytla
• [Asterisk-Users] Chanspy Jitter?   Simone Cittadini
• [Asterisk-Users] IAX2 channel problems   Simone Cittadini
• [Asterisk-Users] increase the volume ?   Simone Cittadini
• [Asterisk-Users] Can I use an onboard modem?   pieter Claassen
• [Asterisk-Users] Page cmd & FOP   Mike Clark
• [Asterisk-Users] open source sip softphone (Window OS version )   Mike Clark
• [Asterisk-Users] Plainvoip problem.   Henry J. Cobb
• [Asterisk-Users] Plainvoip problem.   Henry J. Cobb
• [Asterisk-Users] 100 lines + system config   Henry J. Cobb
• [Asterisk-Users] Plainvoip problem.   Henry J. Cobb
• [Asterisk-Users] How to use a data T-1?   Henry J. Cobb
• [Asterisk-Users] TigerJet PCI PPG FXO Card   Mark Coccimiglio
• [Asterisk-Users] SIP voice recorder   richard Coco
• [Asterisk-Users] SIP voice recorder   richard Coco
• [Asterisk-Users] Asterisk-1.2.9.1 with Siemens HiPath 4000   richard Coco
• [Asterisk-Users] Asterisk-1.2.9.1 with Siemens HiPath 4000   richard Coco
• [Asterisk-Users] Asterisk-1.2.9.1 with Siemens HiPath 4000   richard Coco
• [Asterisk-Users] siemens pbx and asterisk   richard Coco
• [Asterisk-Users] Re: siemens pbx and asterisk   richard Coco
• [Asterisk-Users] Re: Asterisk-1.2.9.1 with Siemens HiPath 4000   richard Coco
• [Asterisk-Users] X100P fails to initialize   Tzafrir Cohen
• [Asterisk-Users] How to make this into a Macro?   Tzafrir Cohen
• [Asterisk-Users] Can´t send emails   Tzafrir Cohen
• [Asterisk-Users] Prices of g729 codec   Tzafrir Cohen
• [Asterisk-Users] Local vs. toll Dial Plan   Tzafrir Cohen
• [Asterisk-Users] Local vs. toll Dial Plan   Tzafrir Cohen
• [Asterisk-Users] Local vs. toll Dial Plan   Tzafrir Cohen
• [Asterisk-Users] Can I use an onboard modem?   Tzafrir Cohen
• [Asterisk-Users] Compiling QuaBri cards   Tzafrir Cohen
• [Asterisk-Users] Asterisk exit on startup   Tzafrir Cohen
• [Asterisk-Users] Compiling QuaBri cards   Tzafrir Cohen
• [Asterisk-Users] Re: syslog server   Tzafrir Cohen
• [Asterisk-Users] app_flite.so   Tzafrir Cohen
• [Asterisk-Users] asterisk-1.2.9 / res_snmp.so   Tzafrir Cohen
• [Asterisk-Users] Quad T1 Card   Tzafrir Cohen
• [Asterisk-Users] QuadBri card   Tzafrir Cohen
• [Asterisk-Users] HELP!!!! Weird TDM2406E unable to bridge all outgoing calls.   Tzafrir Cohen
• [Asterisk-Users] QuadBri card   Tzafrir Cohen
• [Asterisk-Users] FreePBX 2.1.0: Manually rewriting extensions_additional.conf   Tzafrir Cohen
• [Asterisk-Users] how to identify agi crash cause   Tzafrir Cohen
• [Asterisk-Users] h323 with asterisk problem   Tzafrir Cohen
• [Asterisk-Users] Prices of g729 codec   Tzafrir Cohen
• [Asterisk-Users] h323 with asterisk problem   Tzafrir Cohen
• [Asterisk-Users] Disabling debug output   Tzafrir Cohen
• [Asterisk-Users] increase the volume ?   Tzafrir Cohen
• [Asterisk-Users] Xorcom Rapid   Tzafrir Cohen
• [Asterisk-Users] Xorcom Rapid   Tzafrir Cohen
• [Asterisk-Users] Database file to copy for active sessions.   Tzafrir Cohen
• [Asterisk-Users] Using "#include" on zaptel.conf   Tzafrir Cohen
• [Asterisk-Users] Using "#include" on zaptel.conf   Tzafrir Cohen
• [Asterisk-Users] RE: IAX Passing Variables   Tzafrir Cohen
• [Asterisk-Users] Compiling mpg123 under 1.2.9.1 / Ubuntu 6.06   Tzafrir Cohen
• [Asterisk-Users] Xorcom Rapid   Tzafrir Cohen
• [Asterisk-Users] Xorcom Rapid   Tzafrir Cohen
• [Asterisk-Users] Web UI - Best practices?   Tzafrir Cohen
• [Asterisk-Users] kiax - iax2 softphone   Tzafrir Cohen
• [Asterisk-Users] Easiest (best?) linux distribution for dedicatedAsterisk box?   Tzafrir Cohen
• [Asterisk-Users] Easiest (best?) linux distribution for dedicatedAsterisk box?   Tzafrir Cohen
• [Asterisk-Users] Web UI - Best practices?   Tzafrir Cohen
• [Asterisk-Users] TigerJet PCI PPG FXO Card   Tzafrir Cohen
• [Asterisk-Users] Sangoma driver and zaptel   Tzafrir Cohen
• [Asterisk-Users] Backup Question?   Tzafrir Cohen
• [Asterisk-Users] open source sip softphone (Window OS version )   Tzafrir Cohen
• [Asterisk-Users] Hard drive write cache   Tzafrir Cohen
• [Asterisk-Users] how to hang the zap channel   Tzafrir Cohen
• [Asterisk-Users] Anyone see this?   Tzafrir Cohen
• [Asterisk-Users] Zaptel HZ Warning   Tzafrir Cohen
• [Asterisk-Users] Voicemail with NFS   Tzafrir Cohen
• [Asterisk-Users] Voicemail with NFS   Tzafrir Cohen
• [Asterisk-Users] GXP-2000 Audio Quality   Tzafrir Cohen
• [Asterisk-Users] Voicemail with NFS   Tzafrir Cohen
• [Asterisk-Users] Music On Hold troubleshooting   Tzafrir Cohen
• [Asterisk-Users] Bristuff-0.3.0-PRE-1q and & florz patch compile trouble   Tzafrir Cohen
• [Asterisk-Users] finding mac addresses   Tzafrir Cohen
• [Asterisk-Users] finding mac addresses   Tzafrir Cohen
• [Asterisk-Users] software to do sip stress tests   Tzafrir Cohen
• [Asterisk-Users] finding mac addresses   Tzafrir Cohen
• [Asterisk-Users] Asterisk 1.07 crash under Debian Sarge   Tzafrir Cohen
• [Asterisk-Users] Bristuff-0.3.0-PRE-1q and & florz patch compile trouble   Tzafrir Cohen
• [Asterisk-Users] Include files in voicemail.conf   Tzafrir Cohen
• [Asterisk-Users] zapata.conf: recent changes?   Tzafrir Cohen
• [Asterisk-Users] zapata.conf: recent changes?   Tzafrir Cohen
• [Asterisk-Users] How to configure asterisk to emulate FXO signaling ?   Tzafrir Cohen
• [Asterisk-Users] How to configure asterisk to emulate FXO signaling ?   Tzafrir Cohen
• [Asterisk-Users] How to configure asterisk to emulate FXO signaling ?   Tzafrir Cohen
• [Asterisk-Users] Zaptel answering the Line   Tzafrir Cohen
• [Asterisk-Users] Zaptel answering the Line   Tzafrir Cohen
• [Asterisk-Users] Gizmo and Asterisk analysis   Tzafrir Cohen
• [Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!   Tzafrir Cohen
• [Asterisk-Users] using variable   Tzafrir Cohen
• [Asterisk-Users] Mail loop?   Tzafrir Cohen
• [Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!   Tzafrir Cohen
• [Asterisk-Users] Problems compiling Zaptel   Tzafrir Cohen
• [Asterisk-Users] Problems compiling Zaptel   Tzafrir Cohen
• [Asterisk-Users] Ztdummy and Debian on Intel Macmini   Tzafrir Cohen
• [Asterisk-Users] WebPhone   Tzafrir Cohen
• [Asterisk-Users] asterisk shutdown   Tzafrir Cohen
• [Asterisk-Users] Ztdummy and Debian on Intel Macmini   Tzafrir Cohen
• [Asterisk-Users] asterisk shutdown   Tzafrir Cohen
• [Asterisk-Users] Re: Two FXO: How to dial a number when a RING comes in?   John D. Coleman
• [Asterisk-Users] OT but relevant to SMS   Dean Collins
• [Asterisk-Users] Voicemail to Email on Blackberry   Dean Collins
• [Asterisk-Users] Vega 50 10 FXO   Dean Collins
• [Asterisk-Users] Packet8 and Asterisk, do they play nice?   Dean Collins
• [Asterisk-Users] Out of Office Auto Reply:   Dean Collins
• [Asterisk-Users] Microsoft unified communications   Dean Collins
• [Asterisk-Users] Caller ID Matching in extensions.conf   Kevin Collins
• [Asterisk-Users] AEL #include   Michael Collins
• [Asterisk-Users] Asterisk: T1 hunt group setup   Michael Collins
• [Asterisk-Users] AEL #include (Labels and Goto app)   Michael Collins
• [Asterisk-Users] A@H / Trixbox Question   Michael Collins
• [Asterisk-Users] Quad T1 Card   Michael Collins
• [Asterisk-Users] Quad T1 Card   Michael Collins
• [Asterisk-Users] Quad T1 Card   Michael Collins
• [Asterisk-Users] /var/log/asterisk/full ?   Michael Collins
• [Asterisk-Users] AMD Machine Detect   Michael Collins
• [Asterisk-Users] freepbx centos 4 install script?   Michael Collins
• [Asterisk-Users] freepbx centos 4 install script?   Michael Collins
• [Asterisk-Users] DUNDi Not Able to HandleComplexFailoverSituations   Michael Collins
• [Asterisk-Users] RE: IAX Passing Variables   Joshua Colp
• [Asterisk-Users] AEL2   Joshua Colp
• [Asterisk-Users] Quad T1 Card   Joshua Colp
• [Asterisk-Users] Quad T1 Card   Joshua Colp
• [Asterisk-Users] Compiling SVN Trunk   Joshua Colp
• [Asterisk-Users] Compiling SVN Trunk   Joshua Colp
• [Asterisk-Users] IAX DID channels as incoming hunt group?   Joshua Colp
• [Asterisk-Users] IAX DID channels as incoming hunt group?   Joshua Colp
• [Asterisk-Users] Showing Current Calls   Joshua Colp
• [Asterisk-Users] Cisco Gateway   Juan Jose Comellas
• [Asterisk-Users] Cisco Gateway   Juan Jose Comellas
• [Asterisk-Users] New version of NVBackgroundDetect:   Tim Connolly
• [Asterisk-Users] Using HINT with Cisco 7960/SIP   Tim Connolly
• [Asterisk-Users] Sangoma A104D is dropping DTMF digits, during IVR   Andre Courchesne - Consultant
• [Asterisk-Users] Re: New Digium Card b410p   David Cook
• [Asterisk-Users] Polycom-Asterisk hints/presence   Sean Cook
• [Asterisk-Users] Polycom-Asterisk hints/presence   Sean Cook
• [Asterisk-Users] How to make this into a Macro?   Sean Cook
• [Asterisk-Users] Campusing two Asterisk boxes?   Sean Cook
• [Asterisk-Users] syslog server   Sean Cook
• [Asterisk-Users] Re: syslog server   Sean Cook
• [Asterisk-Users] Quad T1 Card   Sean Cook
• [Asterisk-Users] Quad T1 Card   Sean Cook
• [Asterisk-Users] Polycom Configuration   Sean Cook
• [Asterisk-Users] Polycom Configuration   Sean Cook
• [Asterisk-Users] Single T1 card with Echo CancellationtoworkwithDell?   Sean Cook
• [Asterisk-Users] FAX + Digium + SpanDSP   Sean Cook
• [Asterisk-Users] Polycom Buddies in 1.6.6   Sean Cook
• [Asterisk-Users] Dell PowerEdge 1650   Sean Cook
• [Asterisk-Users] FXO for PSTN   Sean Cook
• [Asterisk-Users] point to point T hookup?   Sean Cook
• [Asterisk-Users] point to point T hookup?   Sean Cook
• [Asterisk-Users] point to point T hookup?   Sean Cook
• [Asterisk-Users] (no subject)   Sean Cook
• [Asterisk-Users] Digium Hardware Reliability   Sean Cook
• [Asterisk-Users] Auto answer an IAXY how   Sean Cook
• [Asterisk-Users] Latest SVN with downloaded sounds.   Dave Cotton
• [Asterisk-Users] Latest SVN with downloaded sounds. Update   Dave Cotton
• [Asterisk-Users] dial pattern   Dave Cotton
• [Asterisk-Users] Update   Dave Cotton
• [Asterisk-Users] Anyone see this?   Dave Cotton
• [Asterisk-Users] disabling modules - how?   Dave Cotton
• [Asterisk-Users] Don't use CDRTool From AG-projescts   Dave Cotton
• [Asterisk-Users] best hardphone for Asterisk?   Dave Cotton
• [Asterisk-Users] Tribox - Unistim9.4 Makefile   Ian Cowley
• [Asterisk-Users] best hardphone for Asterisk?   Jonathan k. Creasy
• [Asterisk-Users] Transcoding g.711 -> g.729   Matthew Crocker
• [Asterisk-Users] Hard drive write cache   Matthew Crocker
• [Asterisk-Users] PSTN outgoing DTMF vs. transfer Problem   Doug Crompton
• [Asterisk-Users] PSTN outgoing DTMF vs. transfer Problem   Doug Crompton
• [Asterisk-Users] transfer & other features   Doug Crompton
• [Asterisk-Users] Allowing multiple exchanges   Doug Crompton
• [Asterisk-Users] Local vs. toll Dial Plan   Doug Crompton
• [Asterisk-Users] Local vs. toll Dial Plan   Doug Crompton
• [Asterisk-Users] Local vs. toll Dial Plan   Doug Crompton
• [Asterisk-Users] Local vs. toll Dial Plan   Doug Crompton
• [Asterisk-Users] Compile install error.   Doug Crompton
• [Asterisk-Users] Help - DTMF feedthru   Doug Crompton
• [Asterisk-Users] Local vs. toll Dial Plan   Doug Crompton
• [Asterisk-Users] Compile install error.   Doug Crompton
• [Asterisk-Users] DTMF feedthru again...   Doug Crompton
• [Asterisk-Users] DTMF feedthru again...   Doug Crompton
• [Asterisk-Users] DTMF feedthru again...   Doug Crompton
• [Asterisk-Users] DTMF feedthru again...   Doug Crompton
• [Asterisk-Users] DTMF feedthru again...   Doug Crompton
• [Asterisk-Users] DTMF feedthru again...   Doug Crompton
• [Asterisk-Users] DTMF feedthru again...   Doug Crompton
• [Asterisk-Users] DTMF feedthru again...   Doug Crompton
• [Asterisk-Users] DTMF feedthru again...   Doug Crompton
• [Asterisk-Users] DTMF feedthru again...   Doug Crompton
• [Asterisk-Users] DTMF feedthru again...   Doug Crompton
• [Asterisk-Users] DTMF feedthru again...   Doug Crompton
• [Asterisk-Users] Notice Question   Doug Crompton
• [Asterisk-Users] What does RELAXDTMF do?   Doug Crompton
• [Asterisk-Users] What does RELAXDTMF do?   Doug Crompton
• [Asterisk-Users] AEL2   Doug Crompton
• [Asterisk-Users] What does RELAXDTMF do?   Doug Crompton
• [Asterisk-Users] AEL2   Doug Crompton
• [Asterisk-Users] Dial Plan rules   Doug Crompton
• [Asterisk-Users] hook flash call transfer   Doug Crompton
• [Asterisk-Users] TTS engine query   Doug Crompton
• [Asterisk-Users] spa3102 vs spa3000 differences?   Doug Crompton
• [Asterisk-Users] TTS engine query   Doug Crompton
• [Asterisk-Users] FW: TTS from MySQL   Doug Crompton
• [Asterisk-Users] ztdummy   Doug Crompton
• [Asterisk-Users] ztdummy   Doug Crompton
• [Asterisk-Users] DTMF Talk off   Doug Crompton
• [Asterisk-Users] DTMF Talk off   Doug Crompton
• [Asterisk-Users] DTMF Talk off   Doug Crompton
• [Asterisk-Users] DTMF Talk off   Doug Crompton
• [Asterisk-Users] DTMF Talk off   Doug Crompton
• [Asterisk-Users] DTMF Talk off   Doug Crompton
• [Asterisk-Users] DTMF Talk off   Doug Crompton
• [Asterisk-Users] best hardphone for Asterisk?   Doug Crompton
• [Asterisk-Users] best hardphone for Asterisk?   Doug Crompton
• [Asterisk-Users] best hardphone for Asterisk?   Doug Crompton
• [Asterisk-Users] In-bound faxing working ~1/3 of time.   Ken D'Ambrosio
• [Asterisk-Users] Reception softphone suggestions?   Ken D'Ambrosio
• [Asterisk-Users] How to find out which line in extensions.conf?   Ken D'Ambrosio
• [Asterisk-Users] isdn-data over iax   DRi at b-w-computer.de
• [Asterisk-Users] isdn-data over iax   DRi at b-w-computer.de
• [Asterisk-Users] H323 to SIP connection problem   Daren J. Howell DTCommunication
• [Asterisk-Users] password on radius authentication   Glenn Dalgliesh
• [Asterisk-Users] AEL2 and CID   Aaron Daniel
• [Asterisk-Users] Converting Voicemail wav to mp3   Aaron Daniel
• [Asterisk-Users] Converting Voicemail wav to mp3   Aaron Daniel
• [Asterisk-Users] Converting Voicemail wav to mp3   Aaron Daniel
• [Asterisk-Users] Config Revision Control   Aaron Daniel
• [Asterisk-Users] Config Revision Control   Aaron Daniel
• [Asterisk-Users] NFS and voicemail   Aaron Daniel
• [Asterisk-Users] Config Revision Control   Aaron Daniel
• [Asterisk-Users] Config Revision Control   Aaron Daniel
• [Asterisk-Users] NFS and voicemail   Aaron Daniel
• [Asterisk-Users] Busy Signals after hangup   Aaron Daniel
• [Asterisk-Users] Config Revision Control   Aaron Daniel
• [Asterisk-Users] Unlock / install of Cisco 7940 IP Phone ?   Aaron Daniel
• AW: [Asterisk-Users] Unlock / install of Cisco 7940 IP Phone ?   Aaron Daniel
• [Asterisk-Users] DUNDi Docs   Aaron Daniel
• [Asterisk-Users] DUNDi Users   Aaron Daniel
• [Asterisk-Users] DUNDi Not Able to Handle Complex Failover Situations   Aaron Daniel
• [Asterisk-Users] Anyone see this?   Aaron Daniel
• [Asterisk-Users] DUNDi Not Able to Handle Complex FailoverSituations   Aaron Daniel
• [Asterisk-Users] DUNDi Not Able to Handle Complex FailoverSituations   Aaron Daniel
• [Asterisk-Users] DUNDi Not Able to Handle ComplexFailoverSituations   Aaron Daniel
• [Asterisk-Users] DUNDi Not Able to HandleComplexFailoverSituations   Aaron Daniel
• [Asterisk-Users] DUNDi Not Able to HandleComplexFailoverSituations   Aaron Daniel
• [Asterisk-Users] asterisk load balance   Aaron Daniel
• [Asterisk-Users] asterisk load balance   Aaron Daniel
• [Asterisk-Users] Voicemail with NFS   Aaron Daniel
• [Asterisk-Users] Voicemail volume adjustment   Aaron Daniel
• [Asterisk-Users] Voicemail volume adjustment   Aaron Daniel
• [Asterisk-Users] Realtime Voicemail   Aaron Daniel
• [Asterisk-Users] Realtime SIP Registrations   Aaron Daniel
• [Asterisk-Users] Realtime patch   Aaron Daniel
• [Asterisk-Users] Realtime SIP Registrations   Aaron Daniel
• [Asterisk-Users] Realtime patch   Aaron Daniel
• [Asterisk-Users] Realtime SIP Registrations   Aaron Daniel
• [Asterisk-Users] username in Real-time changes all the time   Aaron Daniel
• [Asterisk-Users] Dial Macro timeout fails   David
• [Asterisk-Users] Asterisk Users Group   David Evennou (Data Masters, Inc.)
• [Asterisk-Users] Caller-ID Info with Voice Mail -- Can it display to the phone?   Paul Davidson
• [Asterisk-Users] Problems compiling Zaptel   Mark Davies
• [Asterisk-Users] Problems compiling Zaptel   Mark Davies
• [Asterisk-Users] ISDN call-progress IE in SETUP frames   Stephen Davies
• [Asterisk-Users] DUNDi Not Able to Handle Complex FailoverSituations   Stephen Davies
• [Asterisk-Users] Delay on calls   Steve Davies
• [Asterisk-Users] dial pattern   Steve Davies
• [Asterisk-Users] Fun with Echo   Steve Davies
• [Asterisk-Users] Fun with Echo   Steve Davies
• [Asterisk-Users] Fun with Echo   Steve Davies
• [Asterisk-Users] Dial Plan rules   Steve Davies
• [Asterisk-Users] spa3102 vs spa3000 differences?   Steve Davies
• [Asterisk-Users] Single T1 card with Echo Cancellation toworkwithDell?   Steve Davies
• [Asterisk-Users] Echo Problem with T411P   Steve Davies
• [Asterisk-Users] Echo Problem with T411P   Steve Davies
• [Asterisk-Users] Echo Problem with T411P   Steve Davies
• [Asterisk-Users] Echo Problem with T411P   Steve Davies
• [Asterisk-Users] Bristuff-0.3.0-PRE-1q and & florz patch compile trouble   Steve Davies
• [Asterisk-Users] Anyone using VoIP WiFi phones?   Steve Davies
• [Asterisk-Users] [ISSUE] Unable to divert external calls.   Steve Davies
• [Asterisk-Users] Sangoma card A101 Card troubles.   Steve Davies
• [Asterisk-Users] h323 with asterisk problem   Daye
• [Asterisk-Users] Asterisk + Linksys PAP2-NA / Call Clearing   Shane DeRidder
• [Asterisk-Users] What does RELAXDTMF do?   Peter J Dean
• [Asterisk-Users] What does RELAXDTMF do?   Peter J Dean
• [Asterisk-Users] [ISSUE] Unable to divert external calls.   Peter J Dean
• [Asterisk-Users] [WORKAROUND] Unable to divert external calls.   Peter J Dean
• [Asterisk-Users] Cell gateway for T-Mobile US??   Rusty Dekema
• [Asterisk-Users] SE Michigan asterisk users group   Rusty Dekema
• [Asterisk-Users] Two FXO: How to dial a number when a RING comes in?   Vincent Delporte
• [Asterisk-Users] Re: Two FXO: How to dial a number when a RING comes in?   Vincent Delporte
• [Asterisk-Users] Re: Two FXO: How to dial a number when a RING comes in?   Vincent Delporte
• [Asterisk-Users] Re: Two FXO: How to dial a number when a RING comes in?   Vincent Delporte
• [Asterisk-Users] Re: Two FXO: How to dial a number when a RING comes in?   Vincent Delporte
• [Asterisk-Users] Re: Two FXO: How to dial a number when a RING comes in?   Vincent Delporte
• [Asterisk-Users] Sangoma A200 Caller ID in UK   Mike Dent
• [Asterisk-Users] DELL PowerEdge 2850 and TE4110P and TE110P   Derek
• [Asterisk-Users] Asterisk not waiting for E&M Wink (I think)   Derek
• [Asterisk-Users] PHP UnixODBC MS SQl 2000   Derek
• [Asterisk-Users] Phone recommendations?   Derek
• [Asterisk-Users] No "ringing" being played to remote caller?   Derek
• [Asterisk-Users] Voicemail with NFS   Mike Diehl
• [Asterisk-Users] Voicemail with NFS   Mike Diehl
• [Asterisk-Users] Voicemail with NFS   Mike Diehl
• [Asterisk-Users] Zork and Asterisk   Simon P. Ditner
• [Asterisk-Users] SIP Presence   Viggiani Domenico
• [Asterisk-Users] Asking for phone number to dial   Don
• [Asterisk-Users] Asking for phone number to dial   Don
• [Asterisk-Users] Asking for phone number to dial   Don
• [Asterisk-Users] DTMF feedthru again...   Doug
• [Asterisk-Users] FXO registration and VegaStream   Peter Doyle
• [Asterisk-Users] FXO registration and VegaStream   Peter Doyle
• [Asterisk-Users] FXO registration and VegaStream   Peter Doyle
• [Asterisk-Users] Duplicate CDRs   Mark Drayton
• [Asterisk-Users] Duplicate CDRs   Mark Drayton
• [Asterisk-Users] SIP auth failed "wrong pw" but pw is correct   Mark Drayton
• [Asterisk-Users] Sharing experiences   Christophe Ngo Van Duc
• [Asterisk-Users] G729 Code   Christophe Ngo Van Duc
• [Asterisk-Users] Best GPL Gui?   Paul Duffy
• [Asterisk-Users] Bullet-proof FXO?   Paul Dugas
• [Asterisk-Users] ITSP in Atlanta?   Paul Dugas
• [Asterisk-Users] Time Based Goto Ifs Act Strange?   Corporate IT Solutions - Michael Dunne
• [Asterisk-Users] GXP 2000 - BLF and Hold/Hangup Answering   Corporate IT Solutions - Michael Dunne
• [Asterisk-Users] Suggested Phone   Corporate IT Solutions - Michael Dunne
• [Asterisk-Users] Sangoma A104D is dropping DTMF digits during IVR   Jay Dutt
• [Asterisk-Users] SIP Channel hangup problem with re-INVITE enabled - ugrent   Hoa Thai Duy
• [Asterisk-Users] SIP Multi Call Generation   Hoa Thai Duy
• [Asterisk-Users] SIP Channel hangup problem with re-INVITE enabled- ugrent   Hoa Thai Duy
• [Asterisk-Users] Background + Dial   Hoa Thai Duy
• [Asterisk-Users] SIP reinvite still does not occour   Hoa Thai Duy
• [Asterisk-Users] sip to h323 ... direct RTP?   Johansson Olle E
• [Asterisk-Users] disabling modules - how?   Johansson Olle E
• [Asterisk-Users] show register users   Johansson Olle E
• [Asterisk-Users] Signaling and media   Johansson Olle E
• [Asterisk-Users] SBC/ATT Supertrunk configuration   Steve Edwards
• [Asterisk-Users] Caller ID problems   Dan Elder
• [Asterisk-Users] Caller ID issue solved (for now)   Dan Elder
• [Asterisk-Users] Random Zap Channel Drops to SIP   Dan Elder
• [Asterisk-Users] Fun with Echo   Dan Elder
• [Asterisk-Users] Calls keep ringing after being picked up   Dan Elder
• [Asterisk-Users] RE: Calls keep ringing after being picked up   Dan Elder
• [Asterisk-Users] Dropped calls continued   Dan Elder
• [Asterisk-Users] RE: Dropped calls continued   Dan Elder
• [Asterisk-Users] Modifying Voicemail menus?   Dan Elder
• [Asterisk-Users] RE: Asterisk-Users Digest, Vol 23, Issue 182   Dan Elder
• [Asterisk-Users] Connection issues   Scott England
• [Asterisk-Users] Asterisk & wengophone   Pasqualotto Enrico
• [Asterisk-Users] SIPCALLID, but which callid?   Erik
• [Asterisk-Users] Polycom-Asterisk hints/presence   Damon Estep
• [Asterisk-Users] Polycom-Asterisk hints/presence   Damon Estep
• [Asterisk-Users] Polycom-Asterisk hints/presence   Damon Estep
• [Asterisk-Users] Polycom-Asterisk hints/presence   Damon Estep
• [Asterisk-Users] Polycom-Asterisk hints/presence   Damon Estep
• [Asterisk-Users] Polycom-Asterisk hints/presence   Damon Estep
• [Asterisk-Users] Polycom-Asterisk hints/presence   Damon Estep
• [Asterisk-Users] Polycom-Asterisk hints/presence   Damon Estep
• [Asterisk-Users] Polycom-Asterisk hints/presence   Damon Estep
• [Asterisk-Users] Polycom-Asterisk hints/presence   Damon Estep
• [Asterisk-Users] Polycom-Asterisk hints/presence   Damon Estep
• [Asterisk-Users] addons trunk make error   Damon Estep
• [Asterisk-Users] addons trunk make error   Damon Estep
• [Asterisk-Users] Polycom-Asterisk hints/presence   Damon Estep
• [Asterisk-Users] ringback tone or signal on the phone somehow?   Damon Estep
• [Asterisk-Users] Polycom Configuration   Damon Estep
• [Asterisk-Users] long distance ask for pin   Damon Estep
• [Asterisk-Users] get value from DB directly   Damon Estep
• [Asterisk-Users] TDM Fax Problems   Damon Estep
• [Asterisk-Users] Sipura SPA-3000 vs Linksys SPA3000   Damon Estep
• [Asterisk-Users] get value from DB directly   Damon Estep
• [Asterisk-Users] get value from DB directly   Damon Estep
• [Asterisk-Users] MOH too loud   Damon Estep
• [Asterisk-Users] Can this config sustain 30 users?   Damon Estep
• [Asterisk-Users] Asterisk as Wholesale   Damon Estep
• [Asterisk-Users] spa3102 vs spa3000 differences?   Damon Estep
• [Asterisk-Users] 100 lines PBX + system config - repost   Damon Estep
• [Asterisk-Users] astdb entry in sip.conf   C F
• [Asterisk-Users] Asterisk: T1 hunt group setup   C F
• [Asterisk-Users] ADIT 600 <=> Asterisk Help   C F
• [Asterisk-Users] Size limitations of extensions.conf   C F
• [Asterisk-Users] SIP Trunking   C F
• [Asterisk-Users] Local vs. toll Dial Plan   C F
• [Asterisk-Users] Directory problem   C F
• [Asterisk-Users] Directory problem   C F
• [Asterisk-Users] Directory problem   C F
• [Asterisk-Users] Directory problem   C F
• [Asterisk-Users] revisit to legacy PBX and CID over PRI   C F
• [Asterisk-Users] Bullet-proof FXO?   C F
• [Asterisk-Users] revisit to legacy PBX and CID over PRI   C F
• [Asterisk-Users] What is Echo?   C F
• [Asterisk-Users] Auto-pickup cisco phones   C F
• [Asterisk-Users] directory   C F
• [Asterisk-Users] Voicemail with NFS   C F
• [Asterisk-Users] Canreinvite   C F
• [Asterisk-Users] T1 delivered via Copper   C F
• [Asterisk-Users] Asterisk 1.07 crash under Debian Sarge   C F
• [Asterisk-Users] chat with asterisk   C F
• [Asterisk-Users] TE420P/TE415P?   C F
• [Asterisk-Users] Showing Current Calls   C F
• [Asterisk-Users] 1.2.9.1 SIP/Local/Queue behaviours weird   C F
• [Asterisk-Users] One way Audio   C F
• [Asterisk-Users] Issue with using dialing PBX digits after call is connected   C F
• [Asterisk-Users] recommended telephones   C F
• [Asterisk-Users] pickup problem   Fabio
• [Asterisk-Users] Low volume/ audio problems on TDM400 card   Fabio
• [Asterisk-Users] (no subject)   Fabio
• [Asterisk-Users] TDM400P bad echo problem, tried lots of things   Fabio
• [Asterisk-Users] STNU spport   Chen Fan
• [Asterisk-Users] STNU spport   Chen Fan
• [Asterisk-Users] STNU spport   Chen Fan
• [Asterisk-Users] PHP-AGI help   Jon Farmer
• [Asterisk-Users] How to retrieve voicemail   Jon Farmer
• [Asterisk-Users] IAX2 and dialin   Matthias Fechner
• [Asterisk-Users] Problem with IAX2 dialin with portunity   Matthias Fechner
• [Asterisk-Users] Asterisk 1.2.8   Matthias Fechner
• [Asterisk-Users] Asterisk 1.2.8   Matthias Fechner
• [Asterisk-Users] Problem with IAX2 dialin with portunity   Matthias Fechner
• [Asterisk-Users] Problem with IAX2 dialin with portunity   Matthias Fechner
• [Asterisk-Users] Re: GXP-2000 (steer clear)   Matthias Fechner
• [Asterisk-Users] Re: GXP-2000 (steer clear)   Matthias Fechner
• [Asterisk-Users] GXP-2000   Matthias Fechner
• [Asterisk-Users] GXP-2000 and Configdownload via TFTP   Matthias Fechner
• [Asterisk-Users] GXP-2000 and Configdownload via TFTP   Matthias Fechner
• [Asterisk-Users] GXP-2000 and Configdownload via TFTP   Matthias Fechner
• [Asterisk-Users] GXP-2000 addressbook   Matthias Fechner
• [Asterisk-Users] GXP-2000 addressbook   Matthias Fechner
• [Asterisk-Users] Playing sound before dialing   Matthias Fechner
• [Asterisk-Users] best hardphone for Asterisk?   Matthias Fechner
• [Asterisk-Users] Playing sound before dialing   Matthias Fechner
• [Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!   Matthias Fechner
• [Asterisk-Users] Polycom-Asterisk hints/presence   Mike Fedyk
• [Asterisk-Users] addons trunk make error   Mike Fedyk
• [Asterisk-Users] Polycom-Asterisk hints/presence   Mike Fedyk
• Adding Asterisk between existing phone system and PSTN Re: [Asterisk-Users] Integrating Asterisk   Mike Fedyk
• [Asterisk-Users] Prices of g729 codec   Mike Fedyk
• [Asterisk-Users] Looking for postpaid quality A-Z termination   Mike Fedyk
• [Asterisk-Users] a new asterisk version   Mike Fedyk
• [Asterisk-Users] GXP-2000   Mike Fedyk
• [Asterisk-Users] GXP-2000   Mike Fedyk
• [Asterisk-Users] GXP-2000   Mike Fedyk
• [Asterisk-Users] Good ATAs from companies other than Sipura/Linksys?   Mike Fedyk
• [Asterisk-Users] GXP-2000   Mike Fedyk
• [Asterisk-Users] GXP-2000   Mike Fedyk
• [Asterisk-Users] GXP-2000   Mike Fedyk
• [Asterisk-Users] Quad T1 Card   Mike Fedyk
• [Asterisk-Users] Quad T1 Card   Mike Fedyk
• [Asterisk-Users] Fun with Echo   Mike Fedyk
• [Asterisk-Users] Reorganizing menus in Polycom 301? Was: [asterisk-biz] New Polycom SoundPoint Series IP-430   Mike Fedyk
• [Asterisk-Users] FXO registration and VegaStream   Mike Fedyk
• [Asterisk-Users] spa3102 vs spa3000 differences?   Mike Fedyk
• [Asterisk-Users] grandstream GXV-3000   Mike Fedyk
• [Asterisk-Users] Hard drive write cache   Mike Fedyk
• [Asterisk-Users] Can this config sustain 30 users?   Mike Fedyk
• [Asterisk-Users] Can this config sustain 30 users?   Mike Fedyk
• [Asterisk-Users] Can this config sustain 30 users?   Mike Fedyk
• [Asterisk-Users] Snom high SIP ping time   Mike Fedyk
• [Asterisk-Users] Are zttest results relevant on a system with no telephony hardware?   Mike Fedyk
• [Asterisk-Users] Can this config sustain 30 users?   Mike Fedyk
• [Asterisk-Users] Are zttest results relevant on a system with no telephony hardware?   Mike Fedyk
• [Asterisk-Users] Linksys SRW224P POE Switch   Mike Fedyk
• [Asterisk-Users] Easiest (best?) linux distribution for dedicated Asterisk box?   Mike Fedyk
• [Asterisk-Users] IAX2 Vs SIP cpu load   Mike Fedyk
• [Asterisk-Users] Easiest (best?) linux distribution for dedicatedAsterisk box?   Mike Fedyk
• [Asterisk-Users] Comfort noise support incomplete in Asterisk (RFC 3389)   Mike Fedyk
• [Asterisk-Users] SPA941 and Echo   Mike Fedyk
• [Asterisk-Users] SPA941 and Echo   Mike Fedyk
• [Asterisk-Users] delay in MeetMe   Mike Fedyk
• [Asterisk-Users] open source sip softphone (Window OS version )   Mike Fedyk
• [Asterisk-Users] Web UI - Best practices?   Mike Fedyk
• [Asterisk-Users] GXP-2000 addressbook   Mike Fedyk
• [Asterisk-Users] Echo Problem with T411P   Mike Fedyk
• [Asterisk-Users] EC needed in all-digital situation?   Mike Fedyk
• [Asterisk-Users] Echo Problem with T411P   Mike Fedyk
• [Asterisk-Users] Best $300 VoIP phone for asterisk?   Mike Fedyk
• [Asterisk-Users] EC needed in all-digital situation?   Mike Fedyk
• [Asterisk-Users] Best $300 VoIP phone for asterisk?   Mike Fedyk
• [Asterisk-Users] Best $300 VoIP phone for asterisk?   Mike Fedyk
• [Asterisk-Users] What ever happened to the LTAPI, the Linux Telephony API?   Mike Fedyk
• [Asterisk-Users] free sun boxes   Mike Fedyk
• [Asterisk-Users] finding mac addresses   Mike Fedyk
• [Asterisk-Users] Echo Problem with T411P   Mike Fedyk
• [Asterisk-Users] DTMF Talk off   Mike Fedyk
• [Asterisk-Users] GXP-2000   Mike Fedyk
• [Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!   Mike Fedyk
• [Asterisk-Users] Mail loop?   Mike Fedyk
• [Asterisk-Users] Wiki Voip Phone reviews   Mike Fedyk
• [Asterisk-Users] Recommended FXO device   Mike Fedyk
• Octasic for TDM2400P and TDM400P? was: [Asterisk-Users] TE420P/TE415P?   Mike Fedyk
• [Asterisk-Users] USB handset options for softphones   Steve Feinstein
• [Asterisk-Users] Unicall Protocol Failure   Martinez Felix
• [Asterisk-Users] Unicall Protocol Failure   Martinez Felix
• [Asterisk-Users] Compiling zaptel on FC5   J.J. Feminella
• [Asterisk-Users] Integrating H.323 gateways with Asterisk?   J.J. Feminella
• [Asterisk-Users] Playing sounds from the CLI   J.J. Feminella
• [Asterisk-Users] Voicemail with NFS   Bruce Ferrell
• [Asterisk-Users] Prices of g729 codec   Brian C. Fertig
• [Asterisk-Users] Prices of g729 codec   Brian C. Fertig
• [Asterisk-Users] Prices of g729 codec   Brian C. Fertig
• [Asterisk-Users] Quad T1 Card   Brian C. Fertig
• [Asterisk-Users] Block access to number@domain.com   Brian C. Fertig
• [Asterisk-Users] T1 passthrough/middleman   Brian C. Fertig
• [Asterisk-Users] Cell gateway for T-Mobile US??   Brian C. Fertig
• [Asterisk-Users] Trying to find good VOIP provider.   Brian C. Fertig
• [Asterisk-Users] Asterisk 1.2.9 cli "-x" doesn't flush?   Bryan Field-Elliot
• [Asterisk-Users] SIP qualify time - best practices?   Bryan Field-Elliot
• [Asterisk-Users] Grandstream BT100 lockup after attended transfer on 1.2.8 and 1.2.9.1   Leonardo Gomes Figueira
• [Asterisk-Users] Agent channel X SIP Transfer on 1.2.9.1   Leonardo Gomes Figueira
• [Asterisk-Users] JIAX status   Rubens Zupelli Filho
• [Asterisk-Users] JIAX status   Rubens Zupelli Filho
• [Asterisk-Users] JIAX status   Rubens Zupelli Filho
• [Asterisk-Users] ADIT 600 <=> Asterisk Help   Bart Fisher
• [Asterisk-Users] How many TE405 ...   Bart Fisher
• [Asterisk-Users] T1 + E&M   Bart Fisher
• [Asterisk-Users] E&M + Dial tone   Bart Fisher
• [Asterisk-Users] Dial Tone + E&M   Bart Fisher
• [Asterisk-Users] Receiving faxes and then sending them on   Frederik Fix
• [Asterisk-Users] Anyone using VoIP WiFi phones?   Barry Flanagan
• [Asterisk-Users] astdb entry in sip.conf   Kevin P. Fleming
• [Asterisk-Users] astdb entry in sip.conf   Kevin P. Fleming
• [Asterisk-Users] Re: G729, voicemail, no codec_g729   Kevin P. Fleming
• [Asterisk-Users] G729, voicemail, no codec_g729   Kevin P. Fleming
• [Asterisk-Users] app_queue and Real roundrobin   Kevin P. Fleming
• [Asterisk-Users] G729, voicemail, no codec_g729   Kevin P. Fleming
• [Asterisk-Users] app_queue and Real roundrobin   Kevin P. Fleming
• [Asterisk-Users] sip channel monitoring   Kevin P. Fleming
• [Asterisk-Users] G729 + Native (files) MOH   Kevin P. Fleming
• [Asterisk-Users] NFS and voicemail   Kevin P. Fleming
• [Asterisk-Users] Problems and questions with setting up a Feature Group D trunk to a Nortel DMS-10 switch   Kevin P. Fleming
• [Asterisk-Users] Problems and questions with setting up a Feature Group D trunk to a Nortel DMS-10 switch   Kevin P. Fleming
• [Asterisk-Users] lspci doesn't show digium card te405p   Kevin P. Fleming
• [Asterisk-Users] Problems and questions with setting up a Feature Group D trunk to a Nortel DMS-10 switch   Kevin P. Fleming
• [Asterisk-Users] Prices of g729 codec   Kevin P. Fleming
• [Asterisk-Users] Prices of g729 codec   Kevin P. Fleming
• [Asterisk-Users] Config Revision Control   Kevin P. Fleming
• [Asterisk-Users] Asterisk + PRI Card -> Nortel BCM   Kevin P. Fleming
• [Asterisk-Users] Asterisk + PRI Card -> Nortel BCM   Kevin P. Fleming
• [Asterisk-Users] Busy Signals after hangup   Kevin P. Fleming
• [Asterisk-Users] Asterisk 1.2.8 - WARNING[28769]: rtp.c:500 ast_rtp_read: Forcing Marker bit, because SSRC has changed   Kevin P. Fleming
• [Asterisk-Users] Meetme versus app_conference   Kevin P. Fleming
• [Asterisk-Users] Meetme versus app_conference   Kevin P. Fleming
• [Asterisk-Users] Inconsistency with ANI and channel callerid   Kevin P. Fleming
• [Asterisk-Users] reinvite   Kevin P. Fleming
• [Asterisk-Users] Asterisk and SATA Raid 1   Kevin P. Fleming
• [Asterisk-Users] Duplicate CDRs   Kevin P. Fleming
• [Asterisk-Users] fine-tuning asterisk questions   Kevin P. Fleming
• [Asterisk-Users] reinvite   Kevin P. Fleming
• [Asterisk-Users] fine-tuning asterisk questions   Kevin P. Fleming
• [Asterisk-Users] Duplicate CDRs   Kevin P. Fleming
• [Asterisk-Users] Configuring behaviour of flash hook   Kevin P. Fleming
• [Asterisk-Users] Prices of g729 codec   Kevin P. Fleming
• [Asterisk-Users] Prices of g729 codec   Kevin P. Fleming
• [Asterisk-Users] Re: Prices of g729 codec   Kevin P. Fleming
• [Asterisk-Users] Compile install error.   Kevin P. Fleming
• [Asterisk-Users] show channel issue with 1.2.9   Kevin P. Fleming
• [Asterisk-Users] Change in dial command behaviour between 1.2.7.1 and 1.2.8?   Kevin P. Fleming
• [Asterisk-Users] Prices of g729 codec   Kevin P. Fleming
• [Asterisk-Users] SIP One-way audio: == Forcing Marker bit, because SSRC has changed - trxtel.com   Kevin P. Fleming
• [Asterisk-Users] IAX Passing Variables   Kevin P. Fleming
• [Asterisk-Users] Transcoding g.711 -> g.729   Kevin P. Fleming
• [Asterisk-Users] Directory problem   Kevin P. Fleming
• [Asterisk-Users] DTMF feedthru again...   Kevin P. Fleming
• [Asterisk-Users] Prices of g729 codec   Kevin P. Fleming
• [Asterisk-Users] Quad T1 Card   Kevin P. Fleming
• [Asterisk-Users] Quad T1 Card   Kevin P. Fleming
• [Asterisk-Users] Quad T1 Card   Kevin P. Fleming
• [Asterisk-Users] Latest SVN with downloaded sounds. Update   Kevin P. Fleming
• [Asterisk-Users] Native Music On Hold Volume LOUD! How to adjust?   Kevin P. Fleming
• [Asterisk-Users] Quad T1 Card   Kevin P. Fleming
• [Asterisk-Users] Quad T1 Card   Kevin P. Fleming
• [Asterisk-Users] bug? asterisk -rx "show dialplan default"   Kevin P. Fleming
• [Asterisk-Users] revisit to legacy PBX and CID over PRI   Kevin P. Fleming
• [Asterisk-Users] Astricon alive and well   Kevin P. Fleming
• [Asterisk-Users] Quad T1 Card   Kevin P. Fleming
• [Asterisk-Users] Asterisk + Zimbra when?   Kevin P. Fleming
• [Asterisk-Users] Call status subscriptions on multiple servers   Kevin P. Fleming
• [Asterisk-Users] H.264 and Motorola Ojo   Kevin P. Fleming
• SV: [Asterisk-Users] Call status subscriptions on multiple servers   Kevin P. Fleming
• [Asterisk-Users] Stupid question zaptel-1.2.6 vs. svn/trunk   Kevin P. Fleming
• [Asterisk-Users] Using "#include" on zaptel.conf   Kevin P. Fleming
• [Asterisk-Users] Compiling SVN Trunk   Kevin P. Fleming
• [Asterisk-Users] Using "#include" on zaptel.conf   Kevin P. Fleming
• [Asterisk-Users] Hitting * in a queue call hangs up?   Kevin P. Fleming
• [Asterisk-Users] IAX DID channels as incoming hunt group?   Kevin P. Fleming
• [Asterisk-Users] Queues and macros and agents   Kevin P. Fleming
• [Asterisk-Users] how to hang the zap channel   Kevin P. Fleming
• [Asterisk-Users] Realtime queue_members and penalties nost escalating (clue anyone?)   Kevin P. Fleming
• [Asterisk-Users] AddQueueMember and Local channels   Kevin P. Fleming
• [Asterisk-Users] No incoming sip calls   Kevin P. Fleming
• [Asterisk-Users] Queues and local channels   Kevin P. Fleming
• [Asterisk-Users] show queue ... Invalid   Kevin P. Fleming
• [Asterisk-Users] Echo Problem with T411P   Kevin P. Fleming
• [Asterisk-Users] Polycom Buddies in 1.6.6   Kevin P. Fleming
• [Asterisk-Users] sip to h323 ... direct RTP?   Kevin P. Fleming
• [Asterisk-Users] TE420P/TE415P?   Kevin P. Fleming
• [Asterisk-Users] Re: fail to make call   Kevin P. Fleming
• [Asterisk-Users] show queue ... Invalid   Kevin P. Fleming
• [Asterisk-Users] sip to h323 ... direct RTP?   Kevin P. Fleming
• [Asterisk-Users] Re: fail to make call   Kevin P. Fleming
• [Asterisk-Users] Re: fail to make call   Kevin P. Fleming
• [Asterisk-Users] 1.2.9.1 crashed today   Kevin P. Fleming
• [Asterisk-Users] SPA-2002 call HANGUP. May be a SIP bug.   Kevin P. Fleming
• [Asterisk-Users] Asterisk 1.2.7/9.1 mp3 volume is good, wav file of same volume are too loud!   Kevin P. Fleming
• [Asterisk-Users] TE420P/TE415P?   Kevin P. Fleming
• [Asterisk-Users] Can Asterisk Send a TEL URI INVITE?   Kevin P. Fleming
• [Asterisk-Users] Meetme versus app_conference   Matt Florell
• [Asterisk-Users] Meetme versus app_conference   Matt Florell
• [Asterisk-Users] Mixing meetme conferences   Matt Florell
• [Asterisk-Users] Prices of g729 codec   Matt Florell
• [Asterisk-Users] Mixing meetme conferences   Matt Florell
• [Asterisk-Users] IAX Passing Variables   Matt Florell
• [Asterisk-Users] Quad T1 Card   Matt Florell
• [Asterisk-Users] [HELP] - Sound cutting and dropping out - 1.2.7.1/Sangoma/PowerEdge 2850   Matt Florell
• [Asterisk-Users] logrotate and logger reload   Matt Florell
• [Asterisk-Users] Distributed ACD Queues   Matt Florell
• [Asterisk-Users] Re: [asterisk-dev] Bridging two existing calls (MeetMe, Sip Reinvite)   Matt Florell
• [Asterisk-Users] Re: [asterisk-dev] Bridging two existing calls (MeetMe, Sip Reinvite)   Matt Florell
• [Asterisk-Users] New VICIDIAL astGUIclient Release: 1.1.12   Matt Florell
• [Asterisk-Users] Meetme max users   Matt Florell
• [Asterisk-Users] Meetme max users   Matt Florell
• [Asterisk-Users] Meetme max users   Matt Florell
• [Asterisk-Users] Meetme max users   Matt Florell
• [Asterisk-Users] Meetme + Sangoma issue?   Matt Florell
• [Asterisk-Users] HELP!!!! Weird TDM2406E unable to bridge all outgoing calls.   El Flynn
• [Asterisk-Users] Voiicemail NFS Cutting Out   El Flynn
• [Asterisk-Users] analog call progress - can I use backgrounddetect   El Flynn
• [Asterisk-Users] massive screetch and echo from Treo 700w   El Flynn
• [Asterisk-Users] Call length limitation   El Flynn
• [Asterisk-Users] voicemail number of recorded messages   El Flynn
• [Asterisk-Users] Echo Cancellation   El Flynn
• [Asterisk-Users] Help with incoming SIP routing   El Flynn
• [Asterisk-Users] Sangoma A200 hangup detection   El Flynn
• [Asterisk-Users] Digium Hardware Reliability   El Flynn
• [Asterisk-Users] recording all calls patch through asterisk   El Flynn
• [Asterisk-Users] Voicemail   El Flynn
• [Asterisk-Users] WIFI sip phone   Alessio Focardi
• [Asterisk-Users] Call Queue NOT using RoundRobin ?!?   Alessio Focardi
• [Asterisk-Users] Call Queue NOT using RoundRobin ?!?   Alessio Focardi
• [Asterisk-Users] Mail loop?   Cory Forsyth
• [Asterisk-Users] asterisk -> my cell phone's voicemail sound problems   Cory Forsyth
• [Asterisk-Users] SpanDSP and analog Digium channels (TDM400P)   Raul Fragoso
• [Asterisk-Users] Two FXO Astralis X101P cards in older PC?   Fred
• [Asterisk-Users] Fun with Echo   Matthew Fredrickson
• [Asterisk-Users] Fun with Echo   Matthew Fredrickson
• [Asterisk-Users] TDM400P bad echo problem, tried lots of things   Matthew Fredrickson
• [Asterisk-Users] TE420P/TE415P?   Matthew Fredrickson
• [Asterisk-Users] Weird Can-Reinvite problem   Jim Freeze
• [Asterisk-Users] ExternalIVR vs AGI   Jim Freeze
• [Asterisk-Users] Addon-ooh323 install problem   Chaim Fried
• [Asterisk-Users] Cisco 7905G SIP firmware needed   Andrea Frigo
• [Asterisk-Users] Realtime queue_members and penalties nost escalating (clue anyone?)   Danny Froberg
• [Asterisk-Users] Realtime queue_members and penalties nost escalating (clue anyone?)   Danny Froberg
• [Asterisk-Users] Problems and questions with setting up a Feature Group D trunk to a Nortel DMS-10 switch   John Fulton
• [Asterisk-Users] Problems and questions with setting up a Feature Group D trunk to a Nortel DMS-10 switch   John Fulton
• [Asterisk-Users] Problems and questions with setting up a Feature Group D trunk to a Nortel DMS-10 switch   John Fulton
• RESOLVED Re: [Asterisk-Users] Problems and questions with setting up a Feature Group D trunk to a Nortel DMS-10 switch   John Fulton
• [Asterisk-Users] Why are sip-channels too lagged?   Arik Raffael Funke
• [Asterisk-Users] Re: No ring tone on outgoing calls   Arik Raffael Funke
• [Asterisk-Users] Re: Forcing Marker bit   Andrew Furey
• [Asterisk-Users] Comfort noise support incomplete in Asterisk (RFC 3389)   Andrew Furey
• [Asterisk-Users] Sip bug...problem seem to be fixed in trunk. How do I find the patch for 1.2   Doug G
• [Asterisk-Users] Sip bug: Problem seem to be fixed in trunk. How do I find a patch for 1.2?   Doug G
• [Asterisk-Users] Realtime SIP Registrations   Doug G
• [Asterisk-Users] Realtime SIP Registrations   Doug G
• [Asterisk-Users] Voiicemail NFS Cutting Out   BILL GITONGA
• [Asterisk-Users] Voice Mail NFS Problem-Cutting Out   BILL GITONGA
• [Asterisk-Users] OPENSER / SER and Asterisk   BILL GITONGA
• [Asterisk-Users] Voice Mail or MWI notify as a (windows) tray icon   Mike Gach
• [Asterisk-Users] Upgrading old version of Asteriak - changes   George Gardiner
• [Asterisk-Users] SIP Jitter buffer. What version of Asterisk PLEASE?   James Gardiner
• [Asterisk-Users] GXP-2000   Kerry Garrison
• [Asterisk-Users] Voicemail to Email on Blackberry   Kerry Garrison
• [Asterisk-Users] open source sip softphone (Window OS version )   Kerry Garrison
• [Asterisk-Users] Polycom-Asterisk hints/presence   Douglas Garstang
• [Asterisk-Users] AEL #include   Douglas Garstang
• [Asterisk-Users] AEL #include   Douglas Garstang
• [Asterisk-Users] Converting Voicemail wav to mp3   Douglas Garstang
• [Asterisk-Users] AEL #include   Douglas Garstang
• [Asterisk-Users] Converting Voicemail wav to mp3   Douglas Garstang
• [Asterisk-Users] Polycom-Asterisk hints/presence   Douglas Garstang
• [Asterisk-Users] Config Revision Control   Douglas Garstang
• [Asterisk-Users] Config Revision Control   Douglas Garstang
• [Asterisk-Users] Config Revision Control   Douglas Garstang
• [Asterisk-Users] Config Revision Control   Douglas Garstang
• [Asterisk-Users] Config Revision Control   Douglas Garstang
• [Asterisk-Users] Config Revision Control   Douglas Garstang
• [Asterisk-Users] Config Revision Control   Douglas Garstang
• [Asterisk-Users] Config Revision Control   Douglas Garstang
• [Asterisk-Users] Config Revision Control   Douglas Garstang
• [Asterisk-Users] Config Revision Control   Douglas Garstang
• [Asterisk-Users] Config Revision Control   Douglas Garstang
• [Asterisk-Users] NFS and voicemail   Douglas Garstang
• [Asterisk-Users] Asterisk chroot   Douglas Garstang
• [Asterisk-Users] RE: Size limitations of extensions.conf   Douglas Garstang
• [Asterisk-Users] Config Revision Control   Douglas Garstang
• [Asterisk-Users] Asterisk chroot   Douglas Garstang
• [Asterisk-Users] Config Revision Control   Douglas Garstang
• [Asterisk-Users] Polycom SIP 1.6.6   Douglas Garstang
• [Asterisk-Users] IAX Passing Variables   Douglas Garstang
• [Asterisk-Users] IAX Passing Variables   Douglas Garstang
• [Asterisk-Users] Transcoding g.711 -> g.729   Douglas Garstang
• [Asterisk-Users] Polycom SIP 1.6.6   Douglas Garstang
• [Asterisk-Users] how to identify agi crash cause   Douglas Garstang
• [Asterisk-Users] RE: IAX Passing Variables   Douglas Garstang
• [Asterisk-Users] RE: IAX Passing Variables   Douglas Garstang
• [Asterisk-Users] Database file to copy for active sessions.   Douglas Garstang
• [Asterisk-Users] Polycom subscriptions   Douglas Garstang
• [Asterisk-Users] AGI Stderr   Douglas Garstang
• [Asterisk-Users] AGI Stderr   Douglas Garstang
• [Asterisk-Users] Asterisk & Eyebeam chat function   Douglas Garstang
• [Asterisk-Users] Asterisk & Eyebeam chat function   Douglas Garstang
• [Asterisk-Users] extensions.conf   Douglas Garstang
• [Asterisk-Users] Polycom Queues   Douglas Garstang
• [Asterisk-Users] OPENSER / SER and Asterisk   Douglas Garstang
• [Asterisk-Users] OPENSER / SER and Asterisk   Douglas Garstang
• [Asterisk-Users] OPENSER / SER and Asterisk   Douglas Garstang
• [Asterisk-Users] OPENSER / SER and Asterisk   Douglas Garstang
• [Asterisk-Users] DUNDi Docs   Douglas Garstang
• [Asterisk-Users] DUNDi Users   Douglas Garstang
• [Asterisk-Users] DUNDi Docs   Douglas Garstang
• [Asterisk-Users] DUNDi Docs   Douglas Garstang
• [Asterisk-Users] DUNDi Users   Douglas Garstang
• [Asterisk-Users] DUNDi Users   Douglas Garstang
• [Asterisk-Users] Determining if extension exists   Douglas Garstang
• [Asterisk-Users] RE: Determining if extension exists   Douglas Garstang
• [Asterisk-Users] DUNDi Not Able to Handle Complex Failover Situations   Douglas Garstang
• [Asterisk-Users] DUNDi Not Able to Handle Complex FailoverSituations   Douglas Garstang
• [Asterisk-Users] DUNDi Not Able to Handle Complex FailoverSituations   Douglas Garstang
• [Asterisk-Users] DUNDi Not Able to Handle Complex FailoverSituations   Douglas Garstang
• [Asterisk-Users] Asterisk Realtime and SIP Registration   Douglas Garstang
• [Asterisk-Users] Distributed ACD Queues   Douglas Garstang
• [Asterisk-Users] ACD Distributed Scenario....   Douglas Garstang
• [Asterisk-Users] DUNDi Not Able to Handle ComplexFailoverSituations   Douglas Garstang
• [Asterisk-Users] DUNDi Not Able to Handle ComplexFailoverSituations   Douglas Garstang
• [Asterisk-Users] DUNDi Not Able to HandleComplexFailoverSituations   Douglas Garstang
• [Asterisk-Users] DUNDi Not Able to Handle ComplexFailoverSituations   Douglas Garstang
• [Asterisk-Users] DUNDILOOKUP and DundiLookup()   Douglas Garstang
• [Asterisk-Users] DUNDi Not Able to Handle ComplexFailoverSituations   Douglas Garstang
• [Asterisk-Users] DUNDi Not Able to HandleComplexFailoverSituations   Douglas Garstang
• [Asterisk-Users] DUNDi Not Able to HandleComplexFailoverSituations   Douglas Garstang
• [Asterisk-Users] DUNDi Not Able to HandleComplexFailoverSituations   Douglas Garstang
• [Asterisk-Users] Executing a Function from AGI   Douglas Garstang
• [Asterisk-Users] Executing a Function from AGI   Douglas Garstang
• [Asterisk-Users] Executing a Function from AGI   Douglas Garstang
• [Asterisk-Users] Zaptel HZ Warning   Douglas Garstang
• [Asterisk-Users] Executing a Function from AGI   Douglas Garstang
• [Asterisk-Users] Re: Executing a Function from AGI   Douglas Garstang
• [Asterisk-Users] Voicemail with NFS   Douglas Garstang
• [Asterisk-Users] Voicemail with NFS   Douglas Garstang
• [Asterisk-Users] Voicemail with NFS   Douglas Garstang
• [Asterisk-Users] Voicemail with NFS   Douglas Garstang
• [Asterisk-Users] Voicemail with NFS   Douglas Garstang
• [Asterisk-Users] Voicemail with NFS   Douglas Garstang
• [Asterisk-Users] Voicemail with NFS   Douglas Garstang
• [Asterisk-Users] Voicemail with NFS   Douglas Garstang
• [Asterisk-Users] asterisk load balance   Douglas Garstang
• [Asterisk-Users] Voicemail with NFS   Douglas Garstang
• [Asterisk-Users] Voicemail with NFS   Douglas Garstang
• [Asterisk-Users] Voicemail with NFS   Douglas Garstang
• [Asterisk-Users] asterisk load balance   Douglas Garstang
• [Asterisk-Users] Voicemail with NFS   Douglas Garstang
• [Asterisk-Users] Polycom Buddies in 1.6.6   Douglas Garstang
• [Asterisk-Users] Conferencing with multiple servers   Douglas Garstang
• [Asterisk-Users] asterisk load balance   Douglas Garstang
• [Asterisk-Users] MySQL Realtime Voicemail Connection Lost   Douglas Garstang
• [Asterisk-Users] Showing Current Calls   Douglas Garstang
• [Asterisk-Users] Showing Current Calls   Douglas Garstang
• [Asterisk-Users] Caller ID Matching in extensions.conf   Douglas Garstang
• [Asterisk-Users] Caller ID Matching in extensions.conf   Douglas Garstang
• [Asterisk-Users] Asterisk Startups   Douglas Garstang
• [Asterisk-Users] Asterisk Startups   Douglas Garstang
• [Asterisk-Users] News: Asterisk VOIP Jobs Site - Revision 3.0 up!   Douglas Garstang
• [Asterisk-Users] Asterisk Startups   Douglas Garstang
• [Asterisk-Users] '500 Internal Server' Error on SIP NOTIFY   Douglas Garstang
• [Asterisk-Users] '500 Internal Server' Error on SIP NOTIFY   Douglas Garstang
• [Asterisk-Users] AGI script can not print out error message toconsole   Douglas Garstang
• [Asterisk-Users] Realtime Voicemail   Douglas Garstang
• [Asterisk-Users] Realtime Voicemail   Douglas Garstang
• [Asterisk-Users] Realtime Voicemail   Douglas Garstang
• [Asterisk-Users] Realtime Voicemail   Douglas Garstang
• [Asterisk-Users] Polycom Buddies in 1.6.6   Douglas Garstang
• [Asterisk-Users] Realtime Voicemail Broken?   Douglas Garstang
• [Asterisk-Users] DUNDi Not Able to HandleComplexFailoverSituations   Douglas Garstang
• [Asterisk-Users] Standard Sound Files Distortion   Douglas Garstang
• [Asterisk-Users] Standard Sound Files Distortion   Douglas Garstang
• [Asterisk-Users] Echo Cancellation   Douglas Garstang
• [Asterisk-Users] '500 Internal Server' Error on SIP NOTIFY   Douglas Garstang
• [Asterisk-Users] Echo Cancellation   Douglas Garstang
• [Asterisk-Users] Realtime: how to use column setvar?   Douglas Garstang
• [Asterisk-Users] Realtime SIP Registrations   Douglas Garstang
• [Asterisk-Users] Realtime SIP Registrations   Douglas Garstang
• [Asterisk-Users] Realtime SIP Registrations   Douglas Garstang
• [Asterisk-Users] Asterisk -x option in 1.2.9.1   Douglas Garstang
• [Asterisk-Users] Auto NOTIFY   Douglas Garstang
• [Asterisk-Users] Auto NOTIFY   Douglas Garstang
• [Asterisk-Users] Voicemail records nonsense, but record() works (??)   Stefan-Michael. Guenther (in-put GbR)
• [Asterisk-Users] Bug in Voicemail ??   Stefan-Michael. Guenther (in-put GbR)
• [Asterisk-Users] Bug in Voicemail ??   Stefan-Michael. Guenther (in-put GbR)
• [Asterisk-Users] Voicemail records nonsense, but record() works (??)   Stefan-Michael. Guenther (in-put GbR)
• [Asterisk-Users] Voicemail records nonsense, but record() works (??)   Stefan-Michael. Guenther (in-put GbR)
• [Asterisk-Users] 2 or more ISDN cards: which comes first ??   Stefan-Michael. Guenther (in-put GbR)
• [Asterisk-Users] SNOM Softphone on windows 2000   Stefan-Michael. Guenther (in-put GbR)
• [Asterisk-Users] 2 or more ISDN cards: which comes first ??   Stefan-Michael. Guenther (in-put GbR)
• [Asterisk-Users] Out of Office Auto Reply:   Doug Geary
• [Asterisk-Users] analog call progress - can I use backgrounddetect   Jerry Geis
• [Asterisk-Users] linksys WIP300 and SMS text messaging   Jerry Geis
• [Asterisk-Users] suggestions for Wireless phone that receives text messages   Jerry Geis
• [Asterisk-Users] uniden uip 200 phones lockup but rare - anyone seen this   Jerry Geis
• [Asterisk-Users] dlink wifi dph-540 and text messaging   Jerry Geis
• [Asterisk-Users] Auto answer an IAXY how   Jerry Geis
• [Asterisk-Users] Auto answer an IAXY how   Jerry Geis
• [Asterisk-Users] best hardphone for Asterisk?   Michael George
• [Asterisk-Users] SE Michigan asterisk users group   Michael George
• [Asterisk-Users] DTMF and ivr systems   Michael George
• [Asterisk-Users] Problem with simple incoming calls   Bill Gibbs
• [Asterisk-Users] Voicemail to Email on Blackberry   Bill Gibbs
• [Asterisk-Users] Polycom Intercom - almost there   Bill Gibbs
• [Asterisk-Users] Polycom 601 problems with multiple registrations   Bill Gibbs
• [Asterisk-Users] Polycom Intercom - almost there   Bill Gibbs
• [Asterisk-Users] Polycom Intercom - almost there   Bill Gibbs
• [Asterisk-Users] Voice calls sent to fax extension   Bill Gibbs
• [Asterisk-Users] Small Asterisk Weather / Cepstral Howto   Matt Gibson
• [Asterisk-Users] weather   Matt Gibson
• [Asterisk-Users] News: Asterisk VOIP Jobs Site - Revision 3.0 up!   Matt Gibson
• [Asterisk-Users] JIAX status   Scott Gifford
• [Asterisk-Users] JIAX status   Scott Gifford
• [Asterisk-Users] JIAX status   Scott Gifford
• [Asterisk-Users] Help with JIAXClient   Scott Gifford
• [Asterisk-Users] SIP Softphone on Thinclient?   Jean-Denis Girard
• [Asterisk-Users] SIP Softphone on Thinclient?   Jean-Denis Girard
• [Asterisk-Users] SIP Softphone on Thinclient?   Jean-Denis Girard
• [Asterisk-Users] Out of Office Auto Reply:   Gireesh.Hariharasubramani at ustri.com
• [Asterisk-Users] Antek EGW-804 e *   Stefano Giuffredi
• [Asterisk-Users] disconnect with mute   Will Glass-Husain
• [Asterisk-Users] Snom high SIP ping time   Steve Glaus
• [Asterisk-Users] Snom high SIP ping time   Steve Glaus
• [Asterisk-Users] Snom high SIP ping time   Steve Glaus
• [Asterisk-Users] Cisco 7960 BLA   Steve Glaus
• [Asterisk-Users] Multiple Sound Folders Support for Same Language (Syntax)   Max Glucksmann
• [Asterisk-Users] Multiple Sound Folder Support for Same Language Syntax   Max Glucksmann
• [Asterisk-Users] Multiple Sound Files Folders - Spanish Syntax. AstCC AGI. 1/3   Max Glucksmann
• [Asterisk-Users] Multiple Sound Files Folders - Spanish Syntax. AstCC AGI. 2/3   Max Glucksmann
• [Asterisk-Users] Multiple Sound Files Folders - Spanish Syntax. AstCC AGI. 3/3   Max Glucksmann
• [Asterisk-Users] Large Asterisk System   Charles R. Gomes
• [Asterisk-Users] Re: What happens if the soekris hardware is defective upon arrival? The Cortex Systems way.   Jonathan Gonzalez
• [Asterisk-Users] QSIG   Giordano Grandis
• [Asterisk-Users] Sip re-invite   Giordano Grandis
• [Asterisk-Users] asterisk compiling   Giordano Grandis
• [Asterisk-Users] Compiling asterisk   Giordano Grandis
• [Asterisk-Users] DID in Houston 713?   Michael Graves
• [Asterisk-Users] DID in Houston 713?   Michael Graves
• [Asterisk-Users] skype out   Michael Graves
• Fwd: [Asterisk-Users] Prices of g729 codec   Michael Graves
• [Asterisk-Users] Best $300 VoIP phone for asterisk?   Michael Graves
• [Asterisk-Users] Best $300 VoIP phone for asterisk?   Michael Graves
• [Asterisk-Users] Best $300 VoIP phone for asterisk?   Michael Graves
• [Asterisk-Users] Best $300 VoIP phone for asterisk?   Michael Graves
• [Asterisk-Users] Best $300 VoIP phone for asterisk?   Michael Graves
• [Asterisk-Users] Nuvio SIP config   Michael Graves
• [Asterisk-Users] Which phones are good, or at least acceptable, for home and office   Michael Graves
• [Asterisk-Users] Re: Which phones are good, or at least acceptable, for home and office   Michael Graves
• [Asterisk-Users] Anyone using VoIP WiFi phones?   Michael Graves
• [Asterisk-Users] USB handset options for softphones   Michael Graves
• [Asterisk-Users] USB handset options for softphones   Michael Graves
• [Asterisk-Users] Anyone using VoIP WiFi phones?   Michael Graves
• [Asterisk-Users] USB handset options for softphones   Michael Graves
• [Asterisk-Users] best hardphone for Asterisk?   Michael Graves
• [Asterisk-Users] Remote employees using Polycom 501 lose ability to receive incoming calls after few minutes.   Michael Graves
• [Asterisk-Users] Callstatus on bridge IAX2 <-> ZAPTEL is always "answer" even if the call fails   Mouss Greg
• [Asterisk-Users] Callstatus on bridge IAX2 <-> ZAPTEL is always "answer" even if the call fails   Mouss Greg
• [Asterisk-Users] Change g729 payload   Attilla De Groot
• [Asterisk-Users] Change g729 payload   Attilla De Groot
• [Asterisk-Users] Change g729 payload   Attilla De Groot
• [Asterisk-Users] Monitor application and e-mailing attachment   Attilla De Groot
• [Asterisk-Users] How to make this into a Macro?   Attilla De Groot
• [Asterisk-Users] Monitor application and e-mailing attachment   Attilla De Groot
• [Asterisk-Users] Monitor application and e-mailing attachment   Attilla De Groot
• [Asterisk-Users] Monitor application and e-mailing attachment   Attilla De Groot
• [Asterisk-Users] Call-pickup function in Queue application   Attilla De Groot
• [Asterisk-Users] Call-pickup function in Queue application   Attilla De Groot
• [Asterisk-Users] Asterisk & Eyebeam chat function   Attilla De Groot
• [Asterisk-Users] Asterisk & Eyebeam chat function   Attilla De Groot
• [Asterisk-Users] chan_capi-cm-0.6 and incoming calls problem   Esteban Guana-Jarrin
• [Asterisk-Users] chan_capi-cm-0.6 and incoming calls problem   Esteban Guana-Jarrin
• [Asterisk-Users] chan_capi-cm-0.6 and incoming calls problem   Esteban Guana-Jarrin
• [Asterisk-Users] chan_capi-cm-0.6 and incoming calls problem   Esteban Guana-Jarrin
• [Asterisk-Users] chan-capi and dtmf   Esteban Guana-Jarrin
• [Asterisk-Users] Prices of g729 codec   Sahil Gupta
• [Asterisk-Users] ASTCC Developer   Sahil Gupta
• Fwd: [Asterisk-Users] Prices of g729 codec   Sahil Gupta
• [Asterisk-Users] PABX Setup   Sahil Gupta
• [Asterisk-Users] PABX Setup   Sahil Gupta
• [Asterisk-Users] Receiving faxes and then sending them on   Craig Guy
• [Asterisk-Users] faxdetect questions - Please HELP!   Craig Guy
• [Asterisk-Users] pbx_spool - outgoing qcall failure upon call progress   Mark Hagler
• [Asterisk-Users] asterisk sip listening port   Yair Hakak
• [Asterisk-Users] SIP Trunking   Steven Haldeman
• [Asterisk-Users] SIP Trunking   Steven Haldeman
• [Asterisk-Users] SIP Trunking   Steven Haldeman
• [Asterisk-Users] ISDN BRI (I.430) over ethernet   Paul Hales
• [Asterisk-Users] PRI & Fax Passthrough   Paul Hales
• [Asterisk-Users] Phone recommendations?   Paul Hales
• [Asterisk-Users] ISDN BRI (I.430) over ethernet   Paul Hales
• [Asterisk-Users] Zultys WIP2 WIFI phone   Paul Hales
• [Asterisk-Users] Hard drive write cache   Paul Hales
• [Asterisk-Users] grandstream GXV-3000   Paul Hales
• [Asterisk-Users] Where's the Fiber   Paul Hales
• [Asterisk-Users] IAX FXS.. Any experience with...   Paul Hales
• [Asterisk-Users] Fun with Echo -- Follow up   Paul Hales
• [Asterisk-Users] IAX FXS.. Any experience with...   Paul Hales
• [Asterisk-Users] Asterisk Startups   Paul Hales
• [Asterisk-Users] Trixbox maunual configuration   Paul Hales
• [Asterisk-Users] Re: siemens pbx and asterisk   Michael Hamann
• [Asterisk-Users] IAX multiport ATA   Mike Hammett
• [Asterisk-Users] Snom high SIP ping time   Mike Hammett
• [Asterisk-Users] RE: Snom high SIP ping time   Mike Hammett
• [Asterisk-Users] app_flite.so   Russell Handorf
• [Asterisk-Users] sangoma card test   Steve Hanselman
• [Asterisk-Users] ICLID or CNAM calling name and number through a cisco isdn gateway   Hanseman, Todd
• [Asterisk-Users] Vega 50 10 FXO   HaoXu
• [Asterisk-Users] choppy audio sip <-> capi   James Harper
• [Asterisk-Users] misdn and dtmf problem resolved   James Harper
• [Asterisk-Users] chan_capi-cm-0.6 and incoming calls problem   James Harper
• [Asterisk-Users] ISDN BRI (I.430) over ethernet   James Harper
• [Asterisk-Users] ISDN BRI (I.430) over ethernet   James Harper
• [Asterisk-Users] ISDN BRI (I.430) over ethernet   James Harper
• [Asterisk-Users] ISDN BRI (I.430) over ethernet   James Harper
• [Asterisk-Users] Trouble getting SMS working   James Harper
• [Asterisk-Users] Trouble getting SMS working   James Harper
• [Asterisk-Users] Trouble getting SMS working   James Harper
• [Asterisk-Users] record until silence, playback, repeat   James Harper
• [Asterisk-Users] Question setting up a "bat phone" extension.   James Harper
• [Asterisk-Users] SIP quality monitoring   James Harper
• [Asterisk-Users] Question setting up a "bat phone" extension.   James Harper
• [Asterisk-Users] Question setting up a "bat phone" extension.   James Harper
• [Asterisk-Users] Cisco router and "488 Not acceptable here" messages   James Harper
• [Asterisk-Users] Cisco router and "488 Not acceptable here" messages   James Harper
• [Asterisk-Users] Cisco router and "488 Not acceptable here"messages   James Harper
• [Asterisk-Users] ISDN and DVO   James Harper
• [Asterisk-Users] SOLVED - Cisco router and "488 Not acceptable here" messages   James Harper
• [Asterisk-Users] - SOLVED - Trouble getting SMS working   James Harper
• [Asterisk-Users] Fun with Echo   James Harper
• [Asterisk-Users] ztdummy   James Harper
• [Asterisk-Users] Gumstix!   James Harper
• [Asterisk-Users] reinvite, DISA, and switching codec's.   James Harper
• [Asterisk-Users] Where's the Fiber   James Harper
• [Asterisk-Users] ISDN BRI NetJet   James Harper
• [Asterisk-Users] Re: ISDN BRI NetJet   James Harper
• [Asterisk-Users] finding mac addresses   James Harper
• [Asterisk-Users] troubleshooting echo on speakerphone   James Harper
• [Asterisk-Users] Is anybody using XEN in conjunction with Asteriskand/or Openser?   James Harper
• [Asterisk-Users] Very bad quality with AVM Fritz!card PCI andchan_capi   James Harper
• [Asterisk-Users] Very bad quality with AVM Fritz!card PCIandchan_capi   James Harper
• [Asterisk-Users] Very bad quality with AVMFritz!cardPCIandchan_capi   James Harper
• [Asterisk-Users] Anyway to customize ring tones on aastra phones?   William Harrison
• [Asterisk-Users] problem - DSL line and Digium card   Eric Hartley
• [Asterisk-Users] problem - DSL line and Digium card   Eric Hartley
• [Asterisk-Users] Asterisk on Mini-Box M300   Darrick Hartman
• [Asterisk-Users] Directory - First Name/Last Name - How to use both? a@h?   Darrick Hartman
• [Asterisk-Users] need help troubleshooting clipping and garbledVOIP calls   James Hawks
• [Asterisk-Users] Switchtype   James Hawks
• [Asterisk-Users] SE Michigan asterisk users group   Tom Hayden
• [Asterisk-Users] F3000 registering to asterisk   Paul Hayes
• [Asterisk-Users] Avaya 4624 Ip phone   Henk
• [Asterisk-Users] Avaya 4624 Ip phone   Henk
• [Asterisk-Users] Very bad quality with AVM Fritz!cardPCIandchan_capi   Henk
• [Asterisk-Users] Avaya 4610sw SIP setup problem   Henk
• [Asterisk-Users] skype out   Paul Hewlett
• [Asterisk-Users] Size limitations of extensions.conf   Paul Hewlett
• [Asterisk-Users] transfer & other features   Paul Hewlett
• [Asterisk-Users] IAX2 Vs SIP cpu load   Paul Hewlett
• [Asterisk-Users] IAX2 Vs SIP cpu load   Paul Hewlett
• [Asterisk-Users] DTMF when using g.729   Paul Hewlett
• [Asterisk-Users] Bristuff-0.3.0-PRE-1q and & florz patch compile trouble   Paul Hewlett
• [Asterisk-Users] Re: Can I enter an extension to dial while voicemail is playing?   Paul Hewlett
• [Asterisk-Users] zapata.conf: recent changes?   Paul Hewlett
• [Asterisk-Users] PCI or MiniPCI Hardware DSP for G.729, G.723.1 and/or GSM   Paul Hewlett
• [Asterisk-Users] Change g729 payload   Jean-Michel Hiver
• [Asterisk-Users] Looking for postpaid quality A-Z termination   Jean-Michel Hiver
• [Asterisk-Users] Looking for postpaid quality A-Z termination   Jean-Michel Hiver
• [Asterisk-Users] Detecting gateways which time out   Jean-Michel Hiver
• [Asterisk-Users] Callback Application: Suggestions Please.   Jean-Michel Hiver
• [Asterisk-Users] VOCAL + Asterisk   Jean-Michel Hiver
• [Asterisk-Users] extensions.conf   Jean-Michel Hiver
• [Asterisk-Users] OPENSER / SER and Asterisk   Jean-Michel Hiver
• [Asterisk-Users] OPENSER / SER and Asterisk   Jean-Michel Hiver
• [Asterisk-Users] ODBC cdr tearing my hair out   Jean-Michel Hiver
• [Asterisk-Users] T1 Copper or T1 Fiber Line   Jean-Michel Hiver
• [Asterisk-Users] Echo Cancelling VoIP traffic   Jean-Michel Hiver
• [Asterisk-Users] AGI: Dial and Recording my own CDR   Jean-Michel Hiver
• [Asterisk-Users] IAX FXS.. Any experience with...   Jean-Michel Hiver
• [Asterisk-Users] AGI: Dial and Recording my own CDR   Jean-Michel Hiver
• [Asterisk-Users] Don't use CDRTool From AG-projescts   Jean-Michel Hiver
• [Asterisk-Users] Signaling and media   Jean-Michel Hiver
• [Asterisk-Users] Signaling and media   Jean-Michel Hiver
• [Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!   Jean-Michel Hiver
• [Asterisk-Users] Most stable Asterisk version   Jean-Michel Hiver
• [Asterisk-Users] Echo Cancellation   Jean-Michel Hiver
• [Asterisk-Users] Echo Cancellation   Jean-Michel Hiver
• [Asterisk-Users] Limiting a group of phones available channels   Jean-Michel Hiver
• [Asterisk-Users] GXP-2000   Shaun Hofer
• [Asterisk-Users] OPENSER / SER and Asterisk   Shaun Hofer
• [Asterisk-Users] DTMF feedthru again...   Nick Hoffman
• [Asterisk-Users] Disabling debug output   Nick Hoffman
• [Asterisk-Users] Debian Sarge or CentOS4.3   Nick Hoffman
• [Asterisk-Users] Asterisk and SER   Nick Hoffman
• [Asterisk-Users] No incoming sip calls   Russell Horn
• [Asterisk-Users] No incoming sip calls   Russell Horn
• [Asterisk-Users] No incoming sip calls   Russell Horn
• [Asterisk-Users] Voicemail beep doesn't end   Russell Horn
• [Asterisk-Users] Asterisk and SATA Raid 1   Tomer Horn
• [Asterisk-Users] Prices of g729 codec   Lee Howard
• [Asterisk-Users] Prices of g729 codec   Lee Howard
• [Asterisk-Users] Prices of g729 codec   Lee Howard
• [Asterisk-Users] Prices of g729 codec   Lee Howard
• [Asterisk-Users] Prices of g729 codec   Lee Howard
• [Asterisk-Users] Prices of g729 codec   Lee Howard
• [Asterisk-Users] A few questions on a conversion to *   Lee Howard
• [Asterisk-Users] Re: faxdetect questions - Please HELP!   Lee Howard
• [Asterisk-Users] Re: faxdetect questions - Please HELP!   Lee Howard
• [Asterisk-Users] Re: faxdetect questions - Please HELP!   Lee Howard
• [Asterisk-Users] RE: Voice calls sent to fax extension   Lee Howard
• [Asterisk-Users] Asterisk ACD with Polycom IP501   Dean at INKnBITs
• [Asterisk-Users] Asterisk ACD with Polycom IP501   Dean at INKnBITs
• [Asterisk-Users] Asterisk ACD with Polycom IP501   Dean at INKnBITs
• [Asterisk-Users] Asterisk ACD with Polycom IP501   Dean at INKnBITs
• [Asterisk-Users] Suggested Phone   Dean at INKnBITs
• [Asterisk-Users] Asterisk ACD with Polycom IP501   Dean at INKnBITs
• [Asterisk-Users] Asterisk ACD Polycom - Please help   Dean at INKnBITs
• [Asterisk-Users] Ordered my first phones :)   George A. Roberts IV
• [Asterisk-Users] DID in Houston 713?   George A. Roberts IV
• [Asterisk-Users] DID in Houston 713?   George A. Roberts IV
• [Asterisk-Users] Odd Asterisk Stress Test Results   Michael Iedema
• [Asterisk-Users] Odd Asterisk Stress Test Results   Michael Iedema
• [Asterisk-Users] How can I use features without enabling 'call parking'?   Koen Van Impe
• [Asterisk-Users] FW: asterisk and nortel meredian option 11c   Koen Van Impe
• [Asterisk-Users] Music On Hold not working with new 1.2.7.1 install   Koen Van Impe
• [Asterisk-Users] Compiling mpg123 under 1.2.9.1 / Ubuntu 6.06   Koen Van Impe
• [Asterisk-Users] Receiving faxes and then sending them on   Koen Van Impe
• [Asterisk-Users] d & e options in meetme()   Koen Van Impe
• [Asterisk-Users] gsm file   Giorgio Incantalupo
• [Asterisk-Users] voicemail suddenly exits on DTMF: a bug?   Giorgio Incantalupo
• [Asterisk-Users] beronet BNS40 led blinking: not working or not connected?   Giorgio Incantalupo
• [Asterisk-Users] beronet BNS40 led blinking: not working or not connected?   Giorgio Incantalupo
• [Asterisk-Users] Re: Two FXO: How to dial a number when a RINGcomes in?   Ioan Indreias
• [Asterisk-Users] Forcing Marker bit   Ira
• [Asterisk-Users] DID from Latvia?   Ira
• [Asterisk-Users] is '9' needed for "outside" numbers   Ira
• [Asterisk-Users] Local vs. toll Dial Plan   Ira
• [Asterisk-Users] Local vs. toll Dial Plan   Ira
• [Asterisk-Users] Local vs. toll Dial Plan   Ira
• [Asterisk-Users] Fun with Echo   Ira
• [Asterisk-Users] Best $300 VoIP phone for asterisk?   Ira
• [Asterisk-Users] Voicemail with NFS   Ira
• [Asterisk-Users] Call limit function on sip channel to external pop   Ira
• [Asterisk-Users] FXO for PSTN   Ira
• [Asterisk-Users] Telsey CPV   Morten Isaksen
• [Asterisk-Users] Problem with callerid in sip to isdn gateway   Morten Isaksen
• [Asterisk-Users] Problem with callerid in sip to isdn gateway   Morten Isaksen
• [Asterisk-Users] Re: Sangoma A101 configuration   Nabeel Jafferali
• [Asterisk-Users] GXP-2000   Nabeel Jafferali
• [Asterisk-Users] spa3102 vs spa3000 differences?   Nabeel Jafferali
• [Asterisk-Users] Anyone using VoIP WiFi phones?   Nabeel Jafferali
• [Asterisk-Users] me, voip.trxtel.com and early media   Nabeel Jafferali
• [Asterisk-Users] Polycom Buddies in 1.6.6   Nabeel Jafferali
• [Asterisk-Users] No Sounds   Bernhard Janetzki
• [Asterisk-Users] No Sounds   Bernhard Janetzki
• [Asterisk-Users] Unicall local_unblocking_expired error   Frederic Jean
• [Asterisk-Users] AGI Stderr   Frederic Jean
• [Asterisk-Users] AGI Stderr   Frederic Jean
• [Asterisk-Users] AGI to read MySQL   Frederic Jean
• [Asterisk-Users] French prompts for calling-card app ?   Frederic Jean
• [Asterisk-Users] Help Asterisk crashes   Fredrik Emil Jensen
• [Asterisk-Users] Help Asterisk crashes   Fredrik Emil Jensen
• [Asterisk-Users] Polycom subscriptions   Kai-Uwe Jensen
• [Asterisk-Users] Asterisk - Qsig   Pavel Jezek
• [Asterisk-Users] IAX jitter / clocking problem   Pavel Jezek
• [Asterisk-Users] IAX jitter / clocking problem   Pavel Jezek
• [Asterisk-Users] SOLVED: IAX jitter / clocking problem   Pavel Jezek
• [Asterisk-Users] Queues and hangup caller on Agent hangup   Johann
• [Asterisk-Users] Odd SIP error message   Johann
• [Asterisk-Users] What to do on a national celebration day? Test, test, test!   Olle E Johansson
• [Asterisk-Users] Astricon No More...   Olle E Johansson
• [Asterisk-Users] Using regcontext   Olle E Johansson
• [Asterisk-Users] Sip transfer, Sip on hold   Olle E Johansson
• [Asterisk-Users] who is the mantainer ....   Olle E Johansson
• [Asterisk-Users] Video phones probem   Olle E Johansson
• [Asterisk-Users] Working with Asterisk and SIP? Register for the Asterisk SIP Master class!   Olle E Johansson
• [Asterisk-Users] Bug in asterisk "static" realtime?   Olle E Johansson
• [Asterisk-Users] Video phones probem   Olle E Johansson
• [Asterisk-Users] Signaling and media   Olle E Johansson
• [Asterisk-Users] Asterisk, mISDN and a Fritz card   Chris Jones
• [Asterisk-Users] Asterisk, mISDN and a Fritz card -- kernel crashes   Chris Jones
• [Asterisk-Users] Asterisk,mISDN and a Fritz card -- kernel   Chris Jones
• [Asterisk-Users] Need help with two-stage ringing macro   Daryl Jones
• [Asterisk-Users] Asterisk: T1 hunt group setup   Jerry Jones
• [Asterisk-Users] OLD PA system.   Jerry Jones
• [Asterisk-Users] OLD PA system.   Jerry Jones
• [Asterisk-Users] How can I use my regular phones with Asterisk running on my Linksys WRT54G router?   Jerry Jones
• [Asterisk-Users] How can I use my regular phones with Asterisk running on my Linksys WRT54G router?   Jerry Jones
• [Asterisk-Users] Polycom Buddies in 1.6.6   Jerry Jones
• [Asterisk-Users] Polycom 601 problems with multiple registrations   Jerry Jones
• [Asterisk-Users] Asterisk 1.2.7.1 with Polycom 501 on SIP -> Conf Calling   Jerry Jones
• [Asterisk-Users] point to point T hookup?   Jerry Jones
• [Asterisk-Users] Asterisk 1.2.7.1 with Polycom 501 on SIP -> ConfCalling   Jerry Jones
• [Asterisk-Users] Asterisk 1.2.7.1 with Polycom 501 on SIP -> Conf Calling   Jerry Jones
• [Asterisk-Users] DTMF and ivr systems   Jerry Jones
• [Asterisk-Users] Asterisk + PRI Card -> Nortel BCM   Mr. Jones
• [Asterisk-Users] Asterisk + PRI Card -> Nortel BCM   Mr. Jones
• [Asterisk-Users] DTMF and DISA   Mr. Jones
• [Asterisk-Users] DTMF and DISA   Mr. Jones
• [Asterisk-Users] Vonage and FXO   Steve Jones
• [Asterisk-Users] Dial Plan rules   Steve Jones
• [Asterisk-Users] Linksys SRW224P POE Switch   Steve Jones
• [Asterisk-Users] DTMF in the middle of a call   Steve Jones
• [Asterisk-Users] Echo and crackle   Steve Jones
• [Asterisk-Users] Which phones are good, or at least acceptable, for home and office   Steve Jones
• [Asterisk-Users] How to use a data T-1?   Steve Jones
• [Asterisk-Users] How to use a data T-1?   Steve Jones
• [Asterisk-Users] Which phones are good, or at least acceptable, for home and office   Steve Jones
• [Asterisk-Users] How to use a data T-1?   Steve Jones
• [Asterisk-Users] How to use a data T-1?   Steve Jones
• [Asterisk-Users] IAX FXS.. Any experience with...   Steve Jones
• [Asterisk-Users] cheapest Cisco Smartnet contract?   Steve Jones
• [Asterisk-Users] Anyone see this?   Senad Jordanovic
• [Asterisk-Users] Anyone see this?   Senad Jordanovic
• [Asterisk-Users] TE420P/TE415P?   Senad Jordanovic
• [Asterisk-Users] freevoip.gedameurope.com - dial out   Joseph
• [Asterisk-Users] 5.8GHz phone and DTMF   Joseph
• [Asterisk-Users] Openion on Sipura SPA-2100   John Joseph
• [Asterisk-Users] I can hear only one way when I use nokia e-60 with X-lite   John Joseph
• SV: [Asterisk-Users] I can hear only one way when I use nokia e-60 withX-lite   John Joseph
• [Asterisk-Users] Nokia N80 and asterisk?   John Joseph
• [Asterisk-Users] Nokai E60 and E61 , working fine with Asterisk , with new access points   John Joseph
• [Asterisk-Users] Re: Nokai E60 and E61 , working fine with Asterisk , with new access points   John Joseph
• [Asterisk-Users] Optimal Hardware   Martin Joseph
• [Asterisk-Users] Looking for very basic example   Martin Joseph
• [Asterisk-Users] IAX multiport ATA   Martin Joseph
• [Asterisk-Users] OT recommend an IAX phone or IAX softphone+USB handset?   Martin Joseph
• [Asterisk-Users] Integrating Asterisk   Martin Joseph
• [Asterisk-Users] Integrating Asterisk   Martin Joseph
• [Asterisk-Users] PSTN outgoing DTMF vs. transfer Problem   Martin Joseph
• [Asterisk-Users] Prices of g729 codec   Martin Joseph
• [Asterisk-Users] Looking for postpaid quality A-Z termination   Martin Joseph
• [Asterisk-Users] Prices of g729 codec   Martin Joseph
• [Asterisk-Users] DTMF feedthru again...   Martin Joseph
• [Asterisk-Users] Notice Question   Martin Joseph
• SV: [Asterisk-Users] I can hear only one way when I use nokia e-60withX-lite   Martin Joseph
• [Asterisk-Users] Analog Line "Static" and Low Volume   Martin Joseph
• [Asterisk-Users] SIP to SIP connection problem   Martin Joseph
• [Asterisk-Users] Good ATAs from companies other than Sipura/Linksys?   Martin Joseph
• [Asterisk-Users] Anyone with GSM488 experience?   Martin Joseph
• [Asterisk-Users] Re: SIP to SIP connection problem   Martin Joseph
• [Asterisk-Users] What does RELAXDTMF do?   Martin Joseph
• [Asterisk-Users] What does RELAXDTMF do?   Martin Joseph
• SV: [Asterisk-Users] I can hear only one way when I use nokiae-60withX-lite   Martin Joseph
• [Asterisk-Users] RE: IAX Passing Variables   Martin Joseph
• [Asterisk-Users] increase the volume ?   Martin Joseph
• [Asterisk-Users] What does RELAXDTMF do?   Martin Joseph
• [Asterisk-Users] increase the volume ?   Martin Joseph
• [Asterisk-Users] Nokia N80 and asterisk?   Martin Joseph
• [Asterisk-Users] Nokai E60 and E61 , working fine with Asterisk , with new access points   Martin Joseph
• [Asterisk-Users] Cisco router and "488 Not acceptable here" messages   Martin Joseph
• [Asterisk-Users] What is Echo?   Martin Joseph
• [Asterisk-Users] Sipura SPA2100 ringing without phone   Martin Joseph
• [Asterisk-Users] Voiicemail NFS Cutting Out   Martin Joseph
• [Asterisk-Users] echo sidetone grandstream and tdm400p   Martin Joseph
• [Asterisk-Users] Can this config sustain 30 users?   Martin Joseph
• [Asterisk-Users] OPENSER / SER and Asterisk   Martin Joseph
• [Asterisk-Users] ztdummy   Martin Joseph
• [Asterisk-Users] How much bandwidth needed?   Martin Joseph
• [Asterisk-Users] SIP codec preference order ineffective   Martin Joseph
• [Asterisk-Users] Best $300 VoIP phone for asterisk?   Martin Joseph
• [Asterisk-Users] Best $300 VoIP phone for asterisk?   Martin Joseph
• [Asterisk-Users] Anyone see this?   Martin Joseph
• [Asterisk-Users] Echo Cancelling VoIP traffic   Martin Joseph
• [Asterisk-Users] Looking for SIP provider with minimal call setuptime   Martin Joseph
• [Asterisk-Users] massive screetch and echo from Treo 700w   Martin Joseph
• [Asterisk-Users] IAX FXS.. Any experience with...   Martin Joseph
• [Asterisk-Users] IAX FXS.. Any experience with...   Martin Joseph
• [Asterisk-Users] User Loses Ability to Make Outgoing Calls   Martin Joseph
• [Asterisk-Users] Fun with Echo -- Follow up   Martin Joseph
• [Asterisk-Users] IAX FXS.. Any experience with...   Martin Joseph
• [Asterisk-Users] Anyone using VoIP WiFi phones?   Martin Joseph
• [Asterisk-Users] USB handset options for softphones   Martin Joseph
• [Asterisk-Users] USB handset options for softphones   Martin Joseph
• [Asterisk-Users] syntax error   Martin Joseph
• [Asterisk-Users] Out of Office Auto Reply:   Martin Joseph
• [Asterisk-Users] problem - DSL line and Digium card   Martin Joseph
• [Asterisk-Users] Upgrading old version of Asteriak - changes   Martin Joseph
• [Asterisk-Users] Caller ID info for DID calls?   Martin Joseph
• [Asterisk-Users] TDMoE question   Martin Joseph
• [Asterisk-Users] DUNDi Not Able to Handle ComplexFailoverSituations   Martin Joseph
• [Asterisk-Users] Signaling and media   Martin Joseph
• [Asterisk-Users] This is getting really annoying - re: POSTFIX   Martin Joseph
• [Asterisk-Users] best hardphone for Asterisk?   Martin Joseph
• [Asterisk-Users] STUN?   Martin Joseph
• [Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!   Martin Joseph
• Re: [Asterisk-Users] Why I can´t upload my wav file?   Martin Joseph
• [Asterisk-Users] Addon-ooh323 install problem   Martin Joseph
• [Asterisk-Users] Most stable Asterisk version   Martin Joseph
• Re: [Asterisk-Users] In other words (Why I can´t upload my wav file?)   Martin Joseph
• [Asterisk-Users] Ztdummy and Debian on Intel Macmini   Martin Joseph
• [Asterisk-Users] need help troubleshooting clipping and garbled VOIP calls   Martin Joseph
• [Asterisk-Users] iax2 group pickup   Bartosz Jozwiak
• [Asterisk-Users] Announcement : A2Billing V1.2.1 released today   Areski K
• [Asterisk-Users] Polycom IP-601 Microbrowser encountered HTTP error 406   KC
• [Asterisk-Users] reinvite   Osama Kamal
• [Asterisk-Users] reinvite   Osama Kamal
• [Asterisk-Users] syslog server   Osama Kamal
• [Asterisk-Users] transcoding problem   Osama Kamal
• [Asterisk-Users] Quantum Voice Asterisk?   Kenny Kant
• [Asterisk-Users] Pickup zap issue   Fredrik von Kantzow
• [Asterisk-Users] asterisk-backports.org   Roy Sigurd Karlsbakk
• [Asterisk-Users] asterisk-backports.org   Roy Sigurd Karlsbakk
• [Asterisk-Users] Gizmo and Asterisk analysis   Roy Sigurd Karlsbakk
• [Asterisk-Users] Transcoding g.711 -> g.729   Mats Karlsson
• [Asterisk-Users] Re: Nokai E60 and E61 , working fine with Asterisk , with new access points   Mats Karlsson
• [Asterisk-Users] JAMAICA DID'S - 1-876   Mats Karlsson
• [Asterisk-Users] Asterisk server   Mats Karlsson
• [Asterisk-Users] Backup Question?   Mats Karlsson
• [Asterisk-Users] H.264 and Motorola Ojo   Nicholas Kathmann
• [Asterisk-Users] Hard drive write cache   Nicholas Kathmann
• [Asterisk-Users] Hard drive write cache   Nicholas Kathmann
• [Asterisk-Users] Linksys SRW224P POE Switch   Nicholas Kathmann
• [Asterisk-Users] quadBRI in bri_net mode - t3 timer expired   Sebastian Kayser
• [Asterisk-Users] Bad call quality using a certain channel.   Shawn Kelley
• [Asterisk-Users] Question about the SET(CALLERID(all)) Function   Shawn Kelley
• [Asterisk-Users] What is the --> priexclusive <-- setting for in zapata.conf?   Shawn Kelley
• [Asterisk-Users] RE: Remote employees using Polycom 501 lose ability to receive incoming calls after few minutes.   Greg Kennedy
• [Asterisk-Users] UK Male English Voices   Steve Kennedy
• [Asterisk-Users] [asterisk-dev] UK Male English Voices   Steve Kennedy
• [Asterisk-Users] Re: Zork and Asterisk   Steve Kennedy
• [Asterisk-Users] [CAVPDiscussion] OT: BT to replace legacy tele com infrastructure with open, standards-based VoIP switches   Steve Kennedy
• [Asterisk-Users] South Africa DIDs   Steve Kennedy
• [Asterisk-Users] South Africa DIDs   Steve Kennedy
• [Asterisk-Users] UK English Sounds   Steve Kennedy
• [Asterisk-Users] Mail loop?   Steve Kennedy
• [Asterisk-Users] Upgrading asterisk   Thomas Kenyon
• [Asterisk-Users] Upgrading asterisk   Thomas Kenyon
• [Asterisk-Users] IAX multiport ATA   Thomas Kenyon
• [Asterisk-Users] Any ideas why I can't dial this SIP phone (sometimes)?   Thomas Kenyon
• [Asterisk-Users] MWI lost after migration   Thomas Kenyon
• [Asterisk-Users] Wanted: CISCO 186 ATAs   Thomas Kenyon
• [Asterisk-Users] GXP-2000   Thomas Kenyon
• [Asterisk-Users] Good ATAs from companies other than Sipura/Linksys?   Thomas Kenyon
• [Asterisk-Users] Re: GXP-2000 (steer clear)   Thomas Kenyon
• [Asterisk-Users] MWI on the PA168V in IAX mode?   Thomas Kenyon
• [Asterisk-Users] query   Thomas Kenyon
• [Asterisk-Users] hangup extension   Thomas Kenyon
• [Asterisk-Users] hangup extension   Thomas Kenyon
• [Asterisk-Users] ADSL modem, TDM400P, zaptel and not hanging up   Thomas Kenyon
• [Asterisk-Users] ADSL modem, TDM400P, zaptel and not hanging up   Thomas Kenyon
• [Asterisk-Users] Question setting up a "bat phone" extension.   Thomas Kenyon
• [Asterisk-Users] OLD PA system.   Thomas Kenyon
• [Asterisk-Users] Question setting up a "bat phone" extension.   Thomas Kenyon
• [Asterisk-Users] asterisk-1.2.9.1   Thomas Kenyon
• [Asterisk-Users] OLD PA system.   Thomas Kenyon
• [Asterisk-Users] OLD PA system.   Thomas Kenyon
• [Asterisk-Users] OLD PA system.   Thomas Kenyon
• [Asterisk-Users] use AT320 international call   Thomas Kenyon
• [Asterisk-Users] use AT320 international call   Thomas Kenyon
• [Asterisk-Users] Upgrading asterisk   Thomas Kenyon
• [Asterisk-Users] Codec negotiation   Thomas Kenyon
• [Asterisk-Users] 1.2.9.1 crashed today   Thomas Kenyon
• [Asterisk-Users] iax2 registration problems   Thomas Kenyon
• [Asterisk-Users] Zaptel answering the Line   Thomas Kenyon
• [Asterisk-Users] Zaptel answering the Line   Thomas Kenyon
• [Asterisk-Users] Zaptel answering the Line   Thomas Kenyon
• [Asterisk-Users] Zaptel answering the Line   Thomas Kenyon
• [Asterisk-Users] Zaptel answering the Line   Thomas Kenyon
• [Asterisk-Users] need help troubleshooting clipping and garbled VOIP calls   Thomas Kenyon
• [Asterisk-Users] need help troubleshooting clipping and garbled VOIP calls   Thomas Kenyon
• [Asterisk-Users] OH323 issue on AT320 Phones   Thomas Kenyon
• [Asterisk-Users] IAX2 Jitterbuffer and trunking   Thomas Kenyon
• [Asterisk-Users] Does anyone know what this means?   Thomas Kenyon
• [Asterisk-Users] Realtime monitor of a channel   Ronan de Kermadec
• [Asterisk-Users] Sipura SPA2100 ringing without phone   Mindaugas Kezys
• [Asterisk-Users] Calling same queue member all the time   Mindaugas Kezys
• [Asterisk-Users] Calling same queue member all the time   Mindaugas Kezys
• [Asterisk-Users] G729, voicemail, no codec_g729   Kristian Kielhofner
• [Asterisk-Users] G729, voicemail, no codec_g729   Kristian Kielhofner
• [Asterisk-Users] G729, voicemail, no codec_g729   Kristian Kielhofner
• [Asterisk-Users] G729 + Native (files) MOH   Kristian Kielhofner
• [Asterisk-Users] Converting Voicemail wav to mp3   Kristian Kielhofner
• [Asterisk-Users] G729 + Native (files) MOH   Kristian Kielhofner
• [Asterisk-Users] asterisk-1.2.9 is not stable   Kristian Kielhofner
• [Asterisk-Users] Native Music On Hold Volume LOUD! How to adjust?   Kristian Kielhofner
• [Asterisk-Users] GXP-2000   Kristian Kielhofner
• [Asterisk-Users] Linksys SRW224P POE Switch   Kristian Kielhofner
• [Asterisk-Users] WRTG54GS Capacity   Kristian Kielhofner
• [Asterisk-Users] Will 200KB/s drive access be OK for voicemailstorage?   Kristian Kielhofner
• [Asterisk-Users] GXP-2000 Audio Quality   Kristian Kielhofner
• [Asterisk-Users] Polycom Intercom - almost there   Kristian Kielhofner
• [Asterisk-Users] Asterisk Native Sound Distortion (ulaw)   Kristian Kielhofner
• [Asterisk-Users] Asterisk Follow Me   Kevin Kiely
• [Asterisk-Users] WCTDM-24xxp woes   Barry King
• [Asterisk-Users] WCTDM-24xxp woes - Patch made   Barry King
• [Asterisk-Users] Bridging two existing calls (MeetMe, Sip Reinvite)   Matt King
• [Asterisk-Users] Re: DTMF Talk off   Matt King
• [Asterisk-Users] shutting down a mysql server renders cdr_mysql dead and asterisk nolonger makes or receives calls   Andrew Kirch
• [Asterisk-Users] shutting down a mysql server renders cdr_mysqldead and asterisk nolonger makes or receives calls   Andrew Kirch
• [Asterisk-Users] Re: Out of Office Auto Reply:   Andrew Kirch
• [Asterisk-Users] WCTDM-24xxp woes   Andrew D Kirch
• [Asterisk-Users] WCTDM-24xxp woes   Andrew D Kirch
• [Asterisk-Users] OLD PA system.   Andrew D Kirch
• [Asterisk-Users] Prices of g729 codec   Ben Klang
• [Asterisk-Users] kiax - iax2 softphone   Ben Klang
• [Asterisk-Users] How can I use my regular phones with Asterisk running on my Linksys WRT54G router?   John Klimek
• [Asterisk-Users] How can I use my regular phones with Asterisk running on my Linksys WRT54G router?   John Klimek
• [Asterisk-Users] Unable to connect to Asterisk? (simple[?] question)   John Klimek
• [Asterisk-Users] Unable to connect to Asterisk? (simple[?] question)   John Klimek
• [Asterisk-Users] Do I need to store voicemail locally?   John Klimek
• [Asterisk-Users] Will 200KB/s drive access be OK for voicemail storage?   John Klimek
• [Asterisk-Users] Easiest (best?) linux distribution for dedicated Asterisk box?   John Klimek
• [Asterisk-Users] Incoming PSTN calls not routing to Asterisk? (using Sipura 3000)   John Klimek
• [Asterisk-Users] Can I enter an extension to dial while voicemail is playing?   John Klimek
• [Asterisk-Users] Re: Can I enter an extension to dial while voicemail is playing?   John Klimek
• [Asterisk-Users] Asterisk --> BV: Incoming does not work....   John Klimek
• [Asterisk-Users] Asterisk --> BV: Incoming does not work....   John Klimek
• [Asterisk-Users] Asterisk --> BV: Incoming does not work....   John Klimek
• [Asterisk-Users] Asterisk --> BV: Incoming does not work....   John Klimek
• [Asterisk-Users] Re: Can I enter an extension to dial while voicemail is playing?   John Klimek
• [Asterisk-Users] Re: Can I enter an extension to dial while voicemail is playing?   John Klimek
• [Asterisk-Users] Re: Can I enter an extension to dial whilevoicemail is playing?   John Klimek
• [Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!   John Klimek
• [Asterisk-Users] Getting at SIP error with SIP_HEADER() ?   Philipp von Klitzing
• [Asterisk-Users] Inconsistency with ANI and channel callerid   Gil Kloepfer
• [Asterisk-Users] logrotate and logger reload   Bob Knight
• [Asterisk-Users] free sun boxes   Bob Knight
• [Asterisk-Users] free sun boxes   Bob Knight
• [Asterisk-Users] free sun boxes   Bob Knight
• [Asterisk-Users] DTMF and DISA   Dave Koberstein
• [Asterisk-Users] VGSM Trouble: Kind people, help me please...   Tigran Kocharyan
• [Asterisk-Users] RE: VGSM Trouble: Kind people, help me please...   Tigran Kocharyan
• [Asterisk-Users] Callback Application: Suggestions Please.   Tigran Kocharyan
• [Asterisk-Users] Callback Application: Suggestions Please.   Tigran Kocharyan
• [Asterisk-Users] Callback Application: Suggestions Please.   Tigran Kocharyan
• [Asterisk-Users] sip to h323 gateway ...   Tigran Kocharyan
• [Asterisk-Users] H.323 soft phone known to be run with asterisk.   Tigran Kocharyan
• [Asterisk-Users] DTMF Detection Problems on VGSM channel   Tigran Kocharyan
• [Asterisk-Users] Playing sound before dialing   Tigran Kocharyan
• [Asterisk-Users] DTMF Detection: Where it happens actually?   Tigran Kocharyan
• [Asterisk-Users] Prices of g729 codec   Andrew Kohlsmith
• [Asterisk-Users] Prices of g729 codec   Andrew Kohlsmith
• [Asterisk-Users] Busy Signals after hangup   Andrew Kohlsmith
• [Asterisk-Users] Config Revision Control   Andrew Kohlsmith
• [Asterisk-Users] Prices of g729 codec   Andrew Kohlsmith
• [Asterisk-Users] Prices of g729 codec   Andrew Kohlsmith
• [Asterisk-Users] Prices of g729 codec   Andrew Kohlsmith
• [Asterisk-Users] Prices of g729 codec   Andrew Kohlsmith
• [Asterisk-Users] Prices of g729 codec   Andrew Kohlsmith
• [Asterisk-Users] Prices of g729 codec   Andrew Kohlsmith
• [Asterisk-Users] Prices of g729 codec   Andrew Kohlsmith
• [Asterisk-Users] Prices of g729 codec   Andrew Kohlsmith
• [Asterisk-Users] Prices of g729 codec   Andrew Kohlsmith
• [Asterisk-Users] How many TE405 ...   Andrew Kohlsmith
• [Asterisk-Users] Quad T1 Card   Andrew Kohlsmith
• [Asterisk-Users] PRI & Fax Passthrough   Andrew Kohlsmith
• [Asterisk-Users] Fun with Echo   Andrew Kohlsmith
• [Asterisk-Users] Fun with Echo   Andrew Kohlsmith
• [Asterisk-Users] Hard drive write cache   Andrew Kohlsmith
• [Asterisk-Users] EC needed in all-digital situation?   Andrew Kohlsmith
• [Asterisk-Users] Best $300 VoIP phone for asterisk?   Andrew Kohlsmith
• [Asterisk-Users] Best $300 VoIP phone for asterisk?   Andrew Kohlsmith
• [Asterisk-Users] EC needed in all-digital situation?   Andrew Kohlsmith
• [Asterisk-Users] WRTG54GS Capacity   Andrew Kohlsmith
• [Asterisk-Users] T1 Copper or T1 Fiber Line   Andrew Kohlsmith
• [Asterisk-Users] Best $300 VoIP phone for asterisk?   Andrew Kohlsmith
• [Asterisk-Users] Best $300 VoIP phone for asterisk?   Andrew Kohlsmith
• [Asterisk-Users] Two FXO: How to dial a number when a RING comes in?   Andrew Kohlsmith
• [Asterisk-Users] Best $300 VoIP phone for asterisk?   Andrew Kohlsmith
• [Asterisk-Users] Best $300 VoIP phone for asterisk?   Andrew Kohlsmith
• [Asterisk-Users] What ever happened to the LTAPI, the Linux Telephony API?   Andrew Kohlsmith
• [Asterisk-Users] Fun with Echo -- Follow up   Andrew Kohlsmith
• [Asterisk-Users] 1.2.9.1 crashed today   Andrew Kohlsmith
• [Asterisk-Users] Anyone using VoIP WiFi phones?   Andrew Kohlsmith
• [Asterisk-Users] Anyone using VoIP WiFi phones?   Andrew Kohlsmith
• [Asterisk-Users] when I press "transfer" -> blind -> 700 . The user is not able to hear what extension the call was parked on   Andrew Kohlsmith
• [Asterisk-Users] Caller ID Matching in extensions.conf   Andrew Kohlsmith
• [Asterisk-Users] PCI or MiniPCI Hardware DSP for G.729, G.723.1 and/or GSM   Andrew Kohlsmith
• [Asterisk-Users] point to point T hookup?   Andrew Kohlsmith
• [Asterisk-Users] Digium Hardware Reliability   Andrew Kohlsmith
• [Asterisk-Users] Digium Hardware Reliability   Andrew Kohlsmith
• [Asterisk-Users] chan_sip.c on debian testing - weird   Michael van der Kolff
• [Asterisk-Users] app_queue and Real roundrobin   Michael Konietzny
• [Asterisk-Users] app_queue and Real roundrobin   Michael Konietzny
• [Asterisk-Users] Asterisk - Qsig   Michael Konietzny
• [Asterisk-Users] Asterisk - Qsig   Michael Konietzny
• [Asterisk-Users] Asterisk - Qsig   Michael Konietzny
• [Asterisk-Users] Trouble getting SMS working   Michael Konietzny
• [Asterisk-Users] Asterisk Zap/QSig with ChanIsAvailable   Michael Konietzny
• [Asterisk-Users] Realtime queue_members and penalties nost escalating (clue anyone?)   Michael Konietzny
• [Asterisk-Users] Call Queue NOT using RoundRobin ?!?   Michael Konietzny
• [Asterisk-Users] Queue - Log if caller disconnects   Michael Konietzny
• [Asterisk-Users] skype out   Koopmann, Jan-Peter
• [Asterisk-Users] Snom 360 with Firmware 6.1?   Koopmann, Jan-Peter
• [Asterisk-Users] Out of Office Auto Reply:   Koopmann, Jan-Peter
• [Asterisk-Users] TDMoE question   Stelios Koroneos
• [Asterisk-Users] Busy/on the phone detection with Cisco 7960   Jeremy Koski
• [Asterisk-Users] Busy/on the phone detection with Cisco 7960   Jeremy Koski
• [Asterisk-Users] Unicall Protocol Failure   Anton Krall
• [Asterisk-Users] Volume configuration on Polycom Soundpoint501phone   Anton Krall
• [Asterisk-Users] Unicall Protocol Failure   Anton Krall
• [Asterisk-Users] Roaming Users   Anton Krall
• [Asterisk-Users] Roaming Users   Anton Krall
• [Asterisk-Users] Changing RO vars like SRC   Anton Krall
• [Asterisk-Users] sangoma unicall m2rfc   Anton Krall
• [Asterisk-Users] sangoma unicall m2rfc   Anton Krall
• [Asterisk-Users] sangoma unicall m2rfc   Anton Krall
• [Asterisk-Users] sangoma unicall m2rfc   Anton Krall
• [Asterisk-Users] sangoma unicall m2rfc   Anton Krall
• [Asterisk-Users] asterisk shutdown   Anton Krall
• [Asterisk-Users] asterisk shutdown   Anton Krall
• [Asterisk-Users] asterisk shutdown   Anton Krall
• [Asterisk-Users] asterisk shutdown   Anton Krall
• [Asterisk-Users] asterisk shutdown   Anton Krall
• [Asterisk-Users] I can hear only one way when I use nokia e-60 withX-lite   Olivier Krief
• [Asterisk-Users] Quad T1 Card   Olivier Krief
• [Asterisk-Users] Which is the best user GUI ?   Olivier Krief
• [Asterisk-Users] TrixBox   Olivier Krief
• [Asterisk-Users] SIP Softphone on Thinclient?   Olivier Krief
• [Asterisk-Users] [Repost] Asterisk realtime   Michael Kromer
• [Asterisk-Users] Asterisk Realtime and "Ex-Girlfriend"   Michael E. Kromer
• [Asterisk-Users] [REPOST] Asterisk Realtime and "Ex-Girlfriend"   Michael E. Kromer
• [Asterisk-Users] Read command   Arjan Kroon
• [Asterisk-Users] Zap problem when calling out   Alexander van der Kuijl
• [Asterisk-Users] Microsoft CRM & Asterisk   Arun Kumar
• [Asterisk-Users] free sun boxes   Arun Kumar
• [Asterisk-Users] Voip / AudioCodes MP-108 Help Needed   Arun Kumar
• [Asterisk-Users] Asterisk auto-dial Help   Arun Kumar
• [Asterisk-Users] a2billing   Arun Kumar
• [Asterisk-Users] Personal Inquiry   Shyhas Kunju
• [Asterisk-Users] Sip t38 gateway tests   Andy Kuo
• [Asterisk-Users] call quality statistics?   Andy Kuo
• [Asterisk-Users] Video phones probem   Mindaugas Kuprys
• [Asterisk-Users] Video phones probem   Mindaugas Kuprys
• [Asterisk-Users] How much bandwidth needed?   Marcin Kwiatkowski
• [Asterisk-Users] Remote employees using Polycom 501 lose ability to receive incoming calls after few minutes.   Von L.
• [Asterisk-Users] Mysql Trixbox   Von L.
• [Asterisk-Users] Re: increase the volume ?   LJ
• [Asterisk-Users] AEL2   Mr. James W. Laferriere
• [Asterisk-Users] siemens pbx and asterisk   Lito Lampitoc
• [Asterisk-Users] Re: siemens pbx and asterisk   Lito Lampitoc
• [Asterisk-Users] asterisk to mobile phone   Lito Lampitoc
• [Asterisk-Users] asterisk to mobile phone   Lito Lampitoc
• [Asterisk-Users] asterisk to mobile phone   Lito Lampitoc
• [Asterisk-Users] FXO for PSTN   Lito Lampitoc
• [Asterisk-Users] FXO for PSTN   Lito Lampitoc
• [Asterisk-Users] FXO for PSTN   Lito Lampitoc
• [Asterisk-Users] FXO for PSTN   Lito Lampitoc
• [Asterisk-Users] Re: siemens pbx and asterisk   Lito Lampitoc
• [Asterisk-Users] hipath 3750   Lito Lampitoc
• [Asterisk-Users] hipath 3750 + hg1500 + asterisk   Lito Lampitoc
• [Asterisk-Users] Question about SIP or IAX2 or both for Asterisk.   Lan
• [Asterisk-Users] How to use a data T-1?   Steve Langstaff
• [Asterisk-Users] Limiting a group of phones available channels   Steve Langstaff
• [Asterisk-Users] Call Not Disconnecting   Abdul Lateef
• [Asterisk-Users] SIP Multi Call Generation   Abdul Lateef
• [Asterisk-Users] Need to Hire: PHP Programmer for PhoneCALL   Andrew Latham
• [Asterisk-Users] Snom 360 Passsword Issue   Andrew Latham
• [Asterisk-Users] Got SUBSCRIBE for extensions without hint   Andrew Latham
• [Asterisk-Users] best hardphone for Asterisk?   Andrew Latham
• [Asterisk-Users] Surge Protector for T1/PRI ?   Andrew Latham
• [Asterisk-Users] FW: asterisk and nortel meredian option 11c   Muhammad Zeeshan Latif
• [Asterisk-Users] RE: help required plzzzzzzzzzz   Muhammad Zeeshan Latif
• [Asterisk-Users] asterisk and nortel meredian option 11c   Muhammad Zeeshan Latif
• [Asterisk-Users] nortel meridian option 11c and asterisk te110p   Muhammad Zeeshan Latif
• [Asterisk-Users] nortel meridian option 11c and asterisk te110p   Muhammad Zeeshan Latif
• [Asterisk-Users] nortel meridian option 11c and asterisk   Muhammad Zeeshan Latif
• [Asterisk-Users] nortel meridian option 11c and asterisk   Muhammad Zeeshan Latif
• [Asterisk-Users] nortel meridian option 11c and asterisk te110p   Muhammad Zeeshan Latif
• [Asterisk-Users] BN8S0 Installation problem - 0 devices registrered   Christophorus Laube
• [Asterisk-Users] BN8S0 Installation problem - 0 devices registrered   Christophorus Laube
• [Asterisk-Users] BN8S0 Installation problem - 0 devices registrered   Christophorus Laube
• [Asterisk-Users] BN8S0 Installation problem - 0 devices registrered   Christophorus Laube
• [Asterisk-Users] BN8S0 Installation problem - 0 devices registrered   Christophorus Laube
• [Asterisk-Users] Dialstatus   Christophorus Laube
• [Asterisk-Users] how to delete a key from database in extensions.conf   Christophorus Laube
• [Asterisk-Users] BN8S0 problem - Extension can never match, so disconnecting   Christophorus Laube
• [Asterisk-Users] remote setting - AGI or what?   Christophorus Laube
• [Asterisk-Users] who is the mantainer ....   Christophorus Laube
• [Asterisk-Users] manager DBDel action   Christophorus Laube
• [Asterisk-Users] CDR Billing   Daniel Laursen
• [Asterisk-Users] Creating Queues on Asterisk server - Sending ingress calls off-net to either PSTN or another VoIP application - thoughts?   Lenz
• [Asterisk-Users] home routers   Lenz
• [Asterisk-Users] Asterisk h323   Lenz
• Fwd: [Asterisk-Users] Prices of g729 codec   Jon Lewis
• [Asterisk-Users] Prices of g729 codec   Jon Lewis
• [Asterisk-Users] 1.2.9.1 crashed today   Tim C. Lewis
• [Asterisk-Users] 1.2.9.1 crashed today   Tim C. Lewis
• [Asterisk-Users] PRI - Ring requested on channel errors - inbound & outbound stop working.   Tim C. Lewis
• [Asterisk-Users] Problems compiling Zaptel   Tim C. Lewis
• [Asterisk-Users] Fun with Echo   Tim C. Lewis
• [Asterisk-Users] teste E1 card   Ralph Liebessohn
• [Asterisk-Users] Fun with Echo -- Follow up   Ralph Liebessohn
• [Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!   Ralph Liebessohn
• [Asterisk-Users] "Reserving" a conference room   Sharon Lim
• [Asterisk-Users] Question about SIP or IAX2 or both for Asterisk.   Sharon Lim
• [Asterisk-Users] click to call features on asterisk   Sharon Lim
• [Asterisk-Users] click to call features on asterisk   Sharon Lim
• [Asterisk-Users] click to call features on asterisk   Sharon Lim
• [Asterisk-Users] conference   Sharon Lim
• [Asterisk-Users] open source sip softphone (Window OS version )   Sharon Lim
• [Asterisk-Users] Incoming PSTN calls not routing to Asterisk? (using Sipura 3000)   Sharon Lim
• [Asterisk-Users] Music On Hold troubleshooting   Sharon Lim
• [Asterisk-Users] how to hang the zap channel   Sharon Lim
• [Asterisk-Users] Call back features   Sharon Lim
• [Asterisk-Users] Problem when i call to asterisk from traditional phones   Omar Lopez Limonta
• [Asterisk-Users] Re: Asterisk-Users Digest, Vol 23, Issue 4   Philippe Lindheimer
• [Asterisk-Users] Re: G729, voicemail, no codec_g729   Philippe Lindheimer
• [Asterisk-Users] Converting Voicemail wav to mp3   Philippe Lindheimer
• [Asterisk-Users] Recommended Web Interface   Philippe Lindheimer
• [Asterisk-Users] FreePBX 2.1.0: Manually rewriting   Philippe Lindheimer
• [Asterisk-Users] MOS Scores and LCR   Philippe Lindheimer
• [Asterisk-Users] Canreinvite   Philippe Lindheimer
• Subject: [Asterisk-Users] Passing DID to external number?   Philippe Lindheimer
• [Asterisk-Users] Passing DID to external number?   Philippe Lindheimer
• [Asterisk-Users] Asterisk home on VMWare time sync issues   Philippe Lindheimer
• [Asterisk-Users] Trixbox maunual configuration   Philippe Lindheimer
• [Asterisk-Users] Remote employees using Polycom 501 lose   Philippe Lindheimer
• [Asterisk-Users] need help troubleshooting clipping and garbledVOIP calls   Philippe Lindheimer
• [Asterisk-Users] Digium Hardware Reliability   Philippe Lindheimer
• [Asterisk-Users] Digium Hardware Reliability   Philippe Lindheimer
• [Asterisk-Users] H.264 and Motorola Ojo   Adam Linford
• [Asterisk-Users] Realtime voicemail not working   Adam Linford
• [Asterisk-Users] HELP!!!! Weird TDM2406E unable to bridge all outgoing calls.   Anderson Ling
• [Asterisk-Users] HELP!!!! Weird TDM2406E unable to bridge all outgoing calls.   Anderson Ling
• [Asterisk-Users] HELP!!!! Weird TDM2406E unable to bridge all outgoing calls.   Anderson Ling
• [Asterisk-Users] Which application to open Zap channel?   Mailing List
• [Asterisk-Users] TE420P/TE415P?   Rob Lith
• [Asterisk-Users] TE420P/TE415P?   Rob Lith
• [Asterisk-Users] G729 Code   Rob Lith
• [Asterisk-Users] What's asterisk on FreeBSD like now a days?   Jason Lixfeld
• [Asterisk-Users] Native Music On Hold Volume LOUD! How to adjust?   Jason Lixfeld
• [Asterisk-Users] Problems with IAX   Jason Lixfeld
• [Asterisk-Users] SIP 486 "Busy Here"   Jason Lixfeld
• [Asterisk-Users] SIP 486 "Busy Here"   Jason Lixfeld
• [Asterisk-Users] SIP 486 "Busy Here"   Jason Lixfeld
• [Asterisk-Users] SIP 486 "Busy Here"   Jason Lixfeld
• [Asterisk-Users] Problems with 7960 + callwaiting   Jason Lixfeld
• [Asterisk-Users] Trouble somewhere with lib compilation   Jason Lixfeld
• [Asterisk-Users] Where's the Fiber   Jason Lixfeld
• [Asterisk-Users] NVFaxDetect   Jason Lixfeld
• [Asterisk-Users] ISDN in Japan   KokMeng Loh
• [Asterisk-Users] zaptel.conf settings for Singtel ISDN-2   KokMeng Loh
• [Asterisk-Users] zaptel.conf settings for Singtel ISDN-2   KokMeng Loh
• [Asterisk-Users] IAX Passing Variables   Alexander Lopez
• [Asterisk-Users] Executing a Function from AGI   Alexander Lopez
• [Asterisk-Users] finding mac addresses   Alexander Lopez
• [Asterisk-Users] SNOM Softphone on windows 2000   Alexander Lopez
• [Asterisk-Users] SNOM Softphone on windows 2000   Alexander Lopez
• [Asterisk-Users] SNOM Softphone on windows 2000   Alexander Lopez
• [Asterisk-Users] Limiting a group of phones available channels   Alexander Lopez
• [Asterisk-Users] Surge Protector for T1/PRI ?   Alexander Lopez
• [Asterisk-Users] Auto answer an IAXY how   Alexander Lopez
• [Asterisk-Users] AMD Machine Detect   Al Lougher
• [Asterisk-Users] AMD Machine Detect   Al Lougher
• [Asterisk-Users] Toll free number comaptible with Voicepulse   Al Lougher
• [Asterisk-Users] Asterisk home on VMWare time sync issues   Al Lougher
• [Asterisk-Users] TDM01B Card Install Problems   Cardiff IT Support Ltd
• [Asterisk-Users] call rejected tone within dialplan   Tristan Graham - Skymarket Ltd
• [Asterisk-Users] How-To monitor a specific channel?   Fernando Lujan
• [Asterisk-Users] ChanSpy on a specific channel.   Fernando Lujan
• [Asterisk-Users] Problems with dial status...   Marcin Lukasik
• [Asterisk-Users] Problems with dial status...   Marcin Lukasik
• [Asterisk-Users] Integrate asterisk with Database   Marcin Lukasik
• [Asterisk-Users] Voicemail   Marcin Lukasik
• [Asterisk-Users] Problems with dial status...   Marcin Lukasik
• [Asterisk-Users] Customer's voice not compatible with service?   Luki
• [Asterisk-Users] Registered SIP:   Luki
• [Asterisk-Users] No reinvite - reason?   Luki
• [Asterisk-Users] WRTG54GS Capacity   Luki
• [Asterisk-Users] reinvite, DISA, and switching codec's.   Luki
• [Asterisk-Users] AEL scripting, CUT use and string concatenation   Marcello Lupo
• [Asterisk-Users] logrotate and logger reload   Richard Lyman
• [Asterisk-Users] Anyone with GSM488 experience?   Jim Lynch
• [Asterisk-Users] Anyone with GSM488 experience?   Jim Lynch
• [Asterisk-Users] Can I get caller id passed to a phone connected to a Supura 2100?   Jim Lynch
• [Asterisk-Users] trunk rollover   Jim Lynch
• [Asterisk-Users] trunk rollover   Jim Lynch
• [Asterisk-Users] DTMF feedthru again...   Mike Lynchfield
• [Asterisk-Users] Zork and Asterisk   Mike Lynchfield
• [Asterisk-Users] Music On Hold not working with new 1.2.7.1 install   Mike Lynchfield
• [Asterisk-Users] Unlock / install of Cisco 7940 IP Phone ?   Mike Lynchfield
• [Asterisk-Users] Disabling debug output   Mike Lynchfield
• [Asterisk-Users] SIP 486 "Busy Here"   Mike Lynchfield
• [Asterisk-Users] Plainvoip problem.   Mike Lynchfield
• [Asterisk-Users] No CID on ZAP   Mike Lynchfield
• [Asterisk-Users] shutting down a mysql server renders cdr_mysql dead and asterisk nolonger makes or receives calls   Mike Lynchfield
• [Asterisk-Users] IAX2 Vs SIP cpu load   Mike Lynchfield
• [Asterisk-Users] IAX2 Vs SIP cpu load   Mike Lynchfield
• [Asterisk-Users] IAX2 Vs SIP cpu load   Mike Lynchfield
• [Asterisk-Users] Re: Asterisk x Siemens HiPath 4000   Mike Lynchfield
• [Asterisk-Users] best hardphone for Asterisk?   Mike Lynchfield
• SV: [Asterisk-Users] Error in config sample for GoToIf?   Mike Lynchfield
• [Asterisk-Users] Re: Asterisk x Siemens HiPath 4000   Mike Lynchfield
• [Asterisk-Users] Realtime SIP Registrations   Mike Lynchfield
• [Asterisk-Users] voting,suggestiuon,your input needed to all   Mike Lynchfield
• [Asterisk-Users] astdb entry in sip.conf   Julian Lyndon-Smith
• [Asterisk-Users] astdb entry in sip.conf   Julian Lyndon-Smith
• [Asterisk-Users] unknown host cvs.digium.com   Julian Lyndon-Smith
• [Asterisk-Users] astdb entry in sip.conf   Julian Lyndon-Smith
• [Asterisk-Users] astdb entry in sip.conf   Julian Lyndon-Smith
• [Asterisk-Users] SIP Jitter buffer. What version of Asterisk PLEASE?   Julian Lyndon-Smith
• [Asterisk-Users] astdb entry in sip.conf   Julian Lyndon-Smith
• [Asterisk-Users] AEL2 and CID   Julian Lyndon-Smith
• [Asterisk-Users] astdb entry in sip.conf   Julian Lyndon-Smith
• [Asterisk-Users] AEL #include   Julian Lyndon-Smith
• [Asterisk-Users] AEL #include (Labels and Goto app)   Julian Lyndon-Smith
• [Asterisk-Users] This should be easy: What happens when the Calling Party hangs up   Julian Lyndon-Smith
• [Asterisk-Users] This should be easy: What happens when the Calling Party hangs up   Julian Lyndon-Smith
• [Asterisk-Users] Queues and macros and agents   Julian Lyndon-Smith
• [Asterisk-Users] Queues and macros and agents   Julian Lyndon-Smith
• [Asterisk-Users] AddQueueMember and Local channels   Julian Lyndon-Smith
• [Asterisk-Users] dial plan return values   Julian Lyndon-Smith
• [Asterisk-Users] AddQueueMember and Local channels   Julian Lyndon-Smith
• [Asterisk-Users] AddQueueMember and Local channels   Julian Lyndon-Smith
• [Asterisk-Users] AddQueueMember and Local channels   Julian Lyndon-Smith
• [Asterisk-Users] Queues and local channels   Julian Lyndon-Smith
• [Asterisk-Users] Auto-pickup cisco phones   Julian Lyndon-Smith
• [Asterisk-Users] Queues and local channels   Julian Lyndon-Smith
• [Asterisk-Users] sangoma card test   Julian Lyndon-Smith
• [Asterisk-Users] sangoma card test   Julian Lyndon-Smith
• [Asterisk-Users] Queues and hangup caller on Agent hangup   Julian Lyndon-Smith
• [Asterisk-Users] Voicemail with NFS   Julian Lyndon-Smith
• [Asterisk-Users] ODBC cdr tearing my hair out   Julian Lyndon-Smith
• [Asterisk-Users] ODBC cdr tearing my hair out   Julian Lyndon-Smith
• [Asterisk-Users] Transfer call via AMI or dialplan   Julian Lyndon-Smith
• [Asterisk-Users] Asterisk 1.07 crash under Debian Sarge   Julian Lyndon-Smith
• [Asterisk-Users] Transfer call via AMI or dialplan   Julian Lyndon-Smith
• [Asterisk-Users] Transfer call via AMI or dialplan   Julian Lyndon-Smith
• [Asterisk-Users] nortel meridian option 11c and asterisk te110p   Julian Lyndon-Smith
• [Asterisk-Users] getting zap peer of sip channel   Julian Lyndon-Smith
• [Asterisk-Users] getting zap peer of sip channel (solved)   Julian Lyndon-Smith
• [Asterisk-Users] Monitor / StopMonitor => MixMonitor / ??   Julian Lyndon-Smith
• [Asterisk-Users] Monitor / StopMonitor => MixMonitor / ?? (yes there is, stupid)   Julian Lyndon-Smith
• [Asterisk-Users] Agent Dump   Julian Lyndon-Smith
• [Asterisk-Users] struggling with the "g" flag   Julian Lyndon-Smith
• [Asterisk-Users] RE: Voice calls sent to fax extension   Julian Lyndon-Smith
• [Asterisk-Users] app_sms not working anymore   Julian Lyndon-Smith
• [Asterisk-Users] app_sms not working anymore   Julian Lyndon-Smith
• [Asterisk-Users] What's the current state of using shared lines in asterisk?   Tom Lynn
• [Asterisk-Users] Can this config sustain 30 users?   Tom Lynn
• [Asterisk-Users] T1 Copper or T1 Fiber Line   Tom Lynn
• [Asterisk-Users] free sun boxes   Tom Lynn
• [Asterisk-Users] Re: Avaya phone 4610sw message waiting indicator and other settings   Tom Lynn
• [Asterisk-Users] Re: Avaya phone 4610sw message waiting indicator and other settings   Tom Lynn
• [Asterisk-Users] Re: Avaya phone 4610sw message waiting indicator and other settings   Tom Lynn
• [Asterisk-Users] Avaya 4610sw SIP setup problem   Tom Lynn
• [Asterisk-Users] Oh oh. Micro$oft just noticed VoIP   Tom Lynn
• [Asterisk-Users] Avaya 4610sw SIP setup problem   Tom Lynn
• [Asterisk-Users] Avaya 4610sw SIP setup problem   Tom Lynn
• [Asterisk-Users] unknown host cvs.digium.com   Doug Lytle
• [Asterisk-Users] Upgrading asterisk   Doug Lytle
• [Asterisk-Users] Re: unknown host cvs.digium.com   Doug Lytle
• [Asterisk-Users] Recommended Web Interface   Doug Lytle
• [Asterisk-Users] ADIT 600 <=> Asterisk Help   Doug Lytle
• [Asterisk-Users] Monitor application and e-mailing attachment   Doug Lytle
• [Asterisk-Users] Monitor application and e-mailing attachment   Doug Lytle
• [Asterisk-Users] Monitor application and e-mailing attachment   Doug Lytle
• [Asterisk-Users] Monitor application and e-mailing attachment   Doug Lytle
• [Asterisk-Users] Call-pickup function in Queue application   Doug Lytle
• [Asterisk-Users] Asterisk & iSeries AS/400   Doug Lytle
• [Asterisk-Users] how to delete a key from database in extensions.conf   Doug Lytle
• [Asterisk-Users] how to delete a key from   Doug Lytle
• [Asterisk-Users] gsm file   Doug Lytle
• [Asterisk-Users] long distance ask for pin   Doug Lytle
• [Asterisk-Users] long distance ask for pin   Doug Lytle
• [Asterisk-Users] OLD PA system.   Doug Lytle
• [Asterisk-Users] TTS engine query   Doug Lytle
• [Asterisk-Users] TTS engine query   Doug Lytle
• [Asterisk-Users] TTS engine query   Doug Lytle
• [Asterisk-Users] TTS engine query   Doug Lytle
• [Asterisk-Users] TTS engine query   Doug Lytle
• [Asterisk-Users] Single T1 card with Echo Cancellation   Doug Lytle
• [Asterisk-Users] Comedian Mail not deleting .txt file   Doug Lytle
• [Asterisk-Users] Comedian Mail not deleting .txt file   Doug Lytle
• [Asterisk-Users] Queues and hangup caller on Agent hangup   Doug Lytle
• [Asterisk-Users] Queues and hangup caller on Agent hangup   Doug Lytle
• [Asterisk-Users] FAX + Digium + SpanDSP   Doug Lytle
• [Asterisk-Users] Zaptel dialing too fast?   Doug Lytle
• [Asterisk-Users] Problem trying to SayDigits when an invalid   Doug Lytle
• [Asterisk-Users] ECHO Tutorial   Doug Lytle
• [Asterisk-Users] Time Based Goto Ifs Act Strange?   Doug Lytle
• [Asterisk-Users] when I press "transfer" -> blind -> 700 . The user   Doug Lytle
• [Asterisk-Users] '500 Internal Server' Error on SIP NOTIFY   Doug Lytle
• [Asterisk-Users] Remote employees using Polycom 501 lose ability to   Doug Lytle
• [Asterisk-Users] asterisk shutdown   Doug Lytle
• [Asterisk-Users] Standard Sound Files Distortion   Doug Lytle
• [Asterisk-Users] ISDN (E1) Hardware Echo Cancellation   Doug Lytle
• [Asterisk-Users] ISDN (E1) Hardware Echo Cancellation   Doug Lytle
• [Asterisk-Users] IAX jitter / clocking problem   Doug Lytle
• [Asterisk-Users] Problems with dial status...   Doug Lytle
• [Asterisk-Users] Asterisk and SATA Raid 1   Julian J. M.
• [Asterisk-Users] asterisk-stat display problems   Julian J. M.
• Re: [Asterisk-Users] ISDN: 3° incoming call   Julian J. M.
• [Asterisk-Users] is '9' needed for "outside" numbers   M.Hockings
• [Asterisk-Users] fine-tuning asterisk questions   M.Hockings
• [Asterisk-Users] Re: fine-tuning asterisk questions   M.Hockings
• [Asterisk-Users] Re: fine-tuning asterisk questions   M.Hockings
• [Asterisk-Users] SIP to SIP connection problem   M.Hockings
• [Asterisk-Users] Re: SIP to SIP connection problem   M.Hockings
• [Asterisk-Users] exactly what ports are required for sip phone to sip voip connection ?   M.Hockings
• [Asterisk-Users] Echo and crackle   M.Hockings
• [Asterisk-Users] Re: Echo and crackle   M.Hockings
• [Asterisk-Users] Which phones are good, or at least acceptable, for home and office   M.Hockings
• [Asterisk-Users] Re: Which phones are good, or at least acceptable, for home and office   M.Hockings
• [Asterisk-Users] Re: faxdetect questions - Please HELP!   M.Hockings
• [Asterisk-Users] Re: Echo and crackle   M.Hockings
• [Asterisk-Users] Re: Which phones are good, or at least acceptable, for home and office   M.Hockings
• [Asterisk-Users] Re: Debian Sarge or CentOS4.3   M.Hockings
• [Asterisk-Users] detecting 1-900 and like exchanges   M.Hockings
• [Asterisk-Users] Digium Hardware Reliability   M.Hockings
• [Asterisk-Users] Re: Digium Hardware Reliability   M.Hockings
• [Asterisk-Users] DUNDi Not Able to Handle ComplexFailoverSituations   Leif Madsen
• [Asterisk-Users] ztdummy   Angelito Manansala
• [Asterisk-Users] free sun boxes   Angelito Manansala
• [Asterisk-Users] Digium TE410P configuration to connect with CIsco 3800   Angelito Manansala
• [Asterisk-Users] Re: Digium TE410P configuration to connect with CIsco 3800   Angelito Manansala
• [Asterisk-Users] Broadvoice - Last Straw!   Robert Mann
• [Asterisk-Users] Asterisk --> BV: Incoming does not work....   Robert Mann
• [Asterisk-Users] Configuring behaviour of flash hook   Henry Margies
• [Asterisk-Users] Dynamic features on call waiting   Henry Margies
• [Asterisk-Users] Compiling VD_app_conference for x86_64   Ricardo Martins
• [Asterisk-Users] Compiling VD_app_conference for x86_64   Ricardo Martins
• [Asterisk-Users] Polycom 601 question   Chris Mason
• [Asterisk-Users] Recommended FXO device   Chris Mason
• [Asterisk-Users] PRI and BRI   Matic
• [Asterisk-Users] Any ideas why I can't dial this SIP phone (sometimes)?   Matt
• [Asterisk-Users] Any ideas why I can't dial this SIP phone (sometimes)?   Matt
• [Asterisk-Users] DTMF feedthru again...   Matt
• [Asterisk-Users] Customer's voice not compatible with service?   Matt
• [Asterisk-Users] Native Music On Hold Volume LOUD! How to adjust?   Matt
• [Asterisk-Users] Native Music On Hold Volume LOUD! How to adjust?   Matt
• [Asterisk-Users] Native Music On Hold Volume LOUD! How to adjust?   Matt
• [Asterisk-Users] Anyway to customize ring tones on aastra phones?   Matt
• [Asterisk-Users] Anyway to customize ring tones on aastra phones?   Matt
• [Asterisk-Users] Attended transfer and queue   Matt
• [Asterisk-Users] Single agent multiple queues....   Matt
• [Asterisk-Users] Hitting * in a queue call hangs up?   Matt
• [Asterisk-Users] Attended transfer and queue   Matt
• [Asterisk-Users] Attended transfer and queue   Matt
• [Asterisk-Users] Attended transfer and queue   Matt
• [Asterisk-Users] Attended transfer and queue   Matt
• [Asterisk-Users] MBX Servers?   Matt
• [Asterisk-Users] Directory - First Name/Last Name - How to use both? a@h?   Matt
• [Asterisk-Users] Directory - First Name/Last Name - How to use both? a@h?   Matt
• [Asterisk-Users] MBX Servers?   Matt
• [Asterisk-Users] Calls keep ringing after being picked up   Matt
• [Asterisk-Users] Comedian Mail not deleting .txt file   Matt
• [Asterisk-Users] Anyone see this?   Matt
• [Asterisk-Users] rollover simulation   Matt
• [Asterisk-Users] Hitting * in a queue call hangs up?   Matt
• [Asterisk-Users] Hitting * in a queue call hangs up?   Matt
• [Asterisk-Users] Setting caller-id when parking call   Matt
• [Asterisk-Users] Newest Asterisk doesn't compile   Matt
• [Asterisk-Users] Re: Newest Asterisk doesn't compile   Matt
• [Asterisk-Users] Hitting * in a queue call hangs up?   Matt
• [Asterisk-Users] Fun with Echo -- Follow up   Matt
• [Asterisk-Users] 1.2.9.1 crashed today   Matt
• [Asterisk-Users] Fun with Echo -- Follow up   Matt
• [Asterisk-Users] 1.2.9.1 crashed today   Matt
• [Asterisk-Users] Fun with Echo -- Follow up   Matt
• [Asterisk-Users] Voicemail cut short?   Matt
• [Asterisk-Users] 1.2.9.1 crashed today   Matt
• [Asterisk-Users] 1.2.9.1 crashed today   Matt
• [Asterisk-Users] Hitting * in a queue call hangs up?   Matt
• [Asterisk-Users] 1.2.9.1 crashed today   Matt
• [Asterisk-Users] Hitting * in a queue call hangs up?   Matt
• [Asterisk-Users] 1.2.9.1 crashed today   Matt
• [Asterisk-Users] 1.2.9.1 crashed today   Matt
• [Asterisk-Users] 1.2.9.1 crashed today   Matt
• [Asterisk-Users] 1.2.9.1 crashed today   Matt
• [Asterisk-Users] Asterisk 1.2.7/9.1 mp3 volume is good, wav file of same volume are too loud!   Matt
• [Asterisk-Users] database copy in asterisk   Matt
• [Asterisk-Users] Asterisk queue log solution?   Matt
• [Asterisk-Users] 1.2.9.1 crashed today   Matt
• [Asterisk-Users] Asterisk queue log solution?   Matt
• [Asterisk-Users] Asterisk queue log solution?   Matt
• [Asterisk-Users] Time Based Goto Ifs Act Strange?   Matt
• [Asterisk-Users] Time Based Goto Ifs Act Strange?   Matt
• [Asterisk-Users] 1.2.9.1 crashed today   Matt
• [Asterisk-Users] 1.2.9.1 crashed today   Matt
• [Asterisk-Users] when I press "transfer" -> blind -> 700 . The user is not able to hear what extension the call was parked on   Matt
• [Asterisk-Users] Setting caller-id when parking call   Matt
• [Asterisk-Users] Dell PowerEdge 1650   Matt
• [Asterisk-Users] This is getting really annoying - re: POSTFIX   Matt
• [Asterisk-Users] F3000 registering to asterisk   Matt
• [Asterisk-Users] how to identify agi crash cause   Josh McAllister
• [Asterisk-Users] how to identify agi crash cause   Josh McAllister
• [Asterisk-Users] Help with 3com router, Asterisk at Home and sipgate.   Brian McCarey
• [Asterisk-Users] Passing DID to external number?   Brian McCarey
• [Asterisk-Users] Passing DID to external number?   Brian McCarey
• [Asterisk-Users] Looking for very basic example   trixter aka Bret McDanel
• [Asterisk-Users] Polycom-Asterisk hints/presence   trixter aka Bret McDanel
• [Asterisk-Users] Restricting amount of incoming calls   trixter aka Bret McDanel
• [Asterisk-Users] Prices of g729 codec   trixter aka Bret McDanel
• [Asterisk-Users] Prices of g729 codec   trixter aka Bret McDanel
• [Asterisk-Users] Prices of g729 codec   trixter aka Bret McDanel
• [Asterisk-Users] All non US 48 area codes?   trixter aka Bret McDanel
• [Asterisk-Users] Prices of g729 codec   trixter aka Bret McDanel
• [Asterisk-Users] Prices of g729 codec   trixter aka Bret McDanel
• [Asterisk-Users] Size limitations of extensions.conf   trixter aka Bret McDanel
• [Asterisk-Users] Meetme versus app_conference   trixter aka Bret McDanel
• [Asterisk-Users] Prices of g729 codec   trixter aka Bret McDanel
• [Asterisk-Users] Prices of g729 codec   trixter aka Bret McDanel
• [Asterisk-Users] Prices of g729 codec   trixter aka Bret McDanel
• [Asterisk-Users] Prices of g729 codec   trixter aka Bret McDanel
• [Asterisk-Users] RE: Size limitations of extensions.conf   trixter aka Bret McDanel
• [Asterisk-Users] porting g729 licenses to another machine   trixter aka Bret McDanel
• [Asterisk-Users] Prices of g729 codec   trixter aka Bret McDanel
• [Asterisk-Users] Prices of g729 codec   trixter aka Bret McDanel
• [Asterisk-Users] Prices of g729 codec   trixter aka Bret McDanel
• [Asterisk-Users] show channel issue with 1.2.9   trixter aka Bret McDanel
• [Asterisk-Users] show channel issue with 1.2.9   trixter aka Bret McDanel
• [Asterisk-Users] STNU spport   trixter aka Bret McDanel
• [Asterisk-Users] STNU spport   trixter aka Bret McDanel
• [Asterisk-Users] app_flite.so   trixter aka Bret McDanel
• [Asterisk-Users] Zork and Asterisk   trixter aka Bret McDanel
• [Asterisk-Users] Re: Zork and Asterisk   trixter aka Bret McDanel
• [Asterisk-Users] Prices of g729 codec   trixter aka Bret McDanel
• [Asterisk-Users] Prices of g729 codec   trixter aka Bret McDanel
• [Asterisk-Users] revisit to legacy PBX and CID over PRI   trixter aka Bret McDanel
• [Asterisk-Users] Prices of g729 codec   trixter aka Bret McDanel
• [Asterisk-Users] Prices of g729 codec   trixter aka Bret McDanel
• [Asterisk-Users] Question setting up a "bat phone" extension.   trixter aka Bret McDanel
• [Asterisk-Users] Asterisk Bounty Doubling program   trixter aka Bret McDanel
• [Asterisk-Users] extensions.conf   trixter aka Bret McDanel
• [Asterisk-Users] Comfort noise support incomplete in Asterisk (RFC 3389)   trixter aka Bret McDanel
• [Asterisk-Users] MOS Scores and LCR   trixter aka Bret McDanel
• [Asterisk-Users] MOS Scores and LCR   trixter aka Bret McDanel
• [Asterisk-Users] MOS Scores and LCR   trixter aka Bret McDanel
• [Asterisk-Users] MOS Scores and LCR   trixter aka Bret McDanel
• [Asterisk-Users] Add Country to CDR's   trixter aka Bret McDanel
• [Asterisk-Users] Add Country to CDR's   trixter aka Bret McDanel
• [Asterisk-Users] SIP Multi Call Generation   trixter aka Bret McDanel
• [Asterisk-Users] Oh oh. Micro$oft just noticed VoIP   trixter aka Bret McDanel
• [Asterisk-Users] Problem with callerid in sip to isdn gateway   trixter aka Bret McDanel
• [Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!   trixter aka Bret McDanel
• [Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!   trixter aka Bret McDanel
• [Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!   trixter aka Bret McDanel
• [Asterisk-Users] Wiki Voip Phone reviews   trixter aka Bret McDanel
• [Asterisk-Users] Limiting a group of phones available channels   trixter aka Bret McDanel
• [Asterisk-Users] Limiting a group of phones available channels   trixter aka Bret McDanel
• [Asterisk-Users] faxdetect questions - Please HELP!   Bob McDowell
• [Asterisk-Users] faxdetect questions - Please HELP!   Bob McDowell
• [Asterisk-Users] Oh oh. Micro$oft just noticed VoIP   Rodney G. McDuff
• [Asterisk-Users] PRI & Fax Passthrough   Callum McGillivray
• [Asterisk-Users] PRI & Fax Passthrough   Callum McGillivray
• [Asterisk-Users] PRI & Fax Passthrough   Callum McGillivray
• [Asterisk-Users] h263 Video Support Questions   Raymond McKay
• [Asterisk-Users] Digium Hardware Reliability   Raymond McKay
• [Asterisk-Users] WIFI sip phone   Raymond McKay
• [Asterisk-Users] Strange one-way audio   Timothy R. McKee
• [Asterisk-Users] Polycom-Asterisk hints/presence   Rob McKrill
• [Asterisk-Users] Polycom SIP 1.6.6   Rob McKrill
• [Asterisk-Users] Asterisk and SATA Raid 1   Mike McMullen
• [Asterisk-Users] Voicemail with NFS   Mike McMullen
• [Asterisk-Users] TDM PCI Master Abort   Jeremy McNamara
• [Asterisk-Users] TDM PCI Master Abort   Jeremy McNamara
• [Asterisk-Users] sip to h323 ... direct RTP?   Jeremy McNamara
• [Asterisk-Users] can Asterisk act as a H.323 Gatekeeper?   Jeremy McNamara
• [Asterisk-Users] Work required - modify Asterisk + SEMS   Jeremy McNamara
• [Asterisk-Users] Asterisk Memory leak   Antoine Megalla
• [Asterisk-Users] Asterisk on Mini-Box M300   Antoine Megalla
• [Asterisk-Users] Unlock / install of Cisco 7940 IP Phone ?   Mateo Meier
• AW: [Asterisk-Users] Unlock / install of Cisco 7940 IP Phone ?   Mateo Meier
• [Asterisk-Users] Unicall acting really funny   Joao Mesquita
• [Asterisk-Users] Unicall acting really funny   Joao Mesquita
• [Asterisk-Users] Unicall acting really funny   Joao Mesquita
• [Asterisk-Users] Unicall acting really funny   Joao Mesquita
• [Asterisk-Users] Trouble getting SMS working   Mick
• [Asterisk-Users] Trouble getting SMS working   Mick
• [Asterisk-Users] Trouble getting SMS working   Mick
• [Asterisk-Users] Trouble getting SMS working   Mick
• [Asterisk-Users] - SOLVED - Trouble getting SMS working   Mick
• [Asterisk-Users] - SOLVED - Trouble getting SMS working   Mick
• [Asterisk-Users] - SOLVED - Trouble getting SMS working   Mick
• [Asterisk-Users] GXP-2000   Mike
• [Asterisk-Users] Web UI - Best practices?   Mike
• [Asterisk-Users] Realtime voicemail not working   Mike
• [Asterisk-Users] Realtime voicemail not working   Mike
• [Asterisk-Users] Realtime voicemail not working   Mike
• [Asterisk-Users] Issue with using dialing PBX digits after call is connected   Mike
• [Asterisk-Users] OPENSER / SER and Asterisk   Simon Miles
• [Asterisk-Users] How to configure ptime for certain codec   Nenad Milidrag
• [Asterisk-Users] Controlling Cisco 7960 Ringtone from Asterisk   Jeremiah Millay
• [Asterisk-Users] Directory - First Name/Last Name - How to, use both? a@h?   Jeremiah Millay
• [Asterisk-Users] transfer & other features   Avi Miller
• [Asterisk-Users] Configuring Polycom 501 IP phones via the console   Avi Miller
• [Asterisk-Users] A@H / Trixbox Question   Avi Miller
• [Asterisk-Users] Digital Receptionist   Avi Miller
• [Asterisk-Users] what are the elements of a good asterisk setup?   Avi Miller
• [Asterisk-Users] Voicemail with NFS   Avi Miller
• [Asterisk-Users] Polycom Intercom - almost there   Avi Miller
• [Asterisk-Users] Debian Sarge or CentOS4.3   Avi Miller
• [Asterisk-Users] ISDN (E1) Hardware Echo Cancellation   Avi Miller
• [Asterisk-Users] ISDN (E1) Hardware Echo Cancellation   Avi Miller
• [Asterisk-Users] no IVR audio but phone to phone fine   Jonathan Miller
• [Asterisk-Users] point to point T hookup?   Jonathan Miller
• [Asterisk-Users] point to point T hookup?   Jonathan Miller
• [Asterisk-Users] point to point T hookup?   Jonathan Miller
• [Asterisk-Users] logrotate and logger reload   Noah Miller
• [Asterisk-Users] Can this config sustain 30 users?   Noah Miller
• [Asterisk-Users] Can this config sustain 30 users?   Noah Miller
• [Asterisk-Users] Two FXO: How to dial a number when a RING comes in?   Noah Miller
• [Asterisk-Users] A few questions on a conversion to *   Noah Miller
• [Asterisk-Users] Two FXO: How to dial a number when a RING comes in?   Noah Miller
• [Asterisk-Users] Polycom 601 problems with multiple registrations   Noah Miller
• [Asterisk-Users] Re: Two FXO: How to dial a number when a RING comes in?   Noah Miller
• [Asterisk-Users] point to point T hookup?   Noah Miller
• [Asterisk-Users] cmdMonitor distortion/crackling   Scott Miller
• [Asterisk-Users] Re: Asterisk-Users Digest, Vol 23, Issue 114   Gregory L Miller-Kramer
• [Asterisk-Users] asterisk on AMD 64 BIT   John Millican
• [Asterisk-Users] DTMF Talk off   John Millican
• [Asterisk-Users] How to use a data T-1?   John Millican
• [Asterisk-Users] How to use a data T-1?   John Millican
• [Asterisk-Users] DTMF Talk off   John Millican
• [Asterisk-Users] DTMF Talk off   John Millican
• [Asterisk-Users] home routers   John Millican
• [Asterisk-Users] Re: DTMF Talk off   John Millican
• [Asterisk-Users] DTMF Talk off   John Millican
• [Asterisk-Users] DTMF Talk off   John Millican
• [Asterisk-Users] DTMF Talk off   John Millican
• [Asterisk-Users] Asking for phone number to dial   John Millican
• [Asterisk-Users] MWI lost after migration   Mimmus
• [Asterisk-Users] MWI lost after migration   Mimmus
• [Asterisk-Users] Re: GXP-2000 (steer clear)   Mimmus
• [Asterisk-Users] Re: GXP-2000 (steer clear)   Mimmus
• [Asterisk-Users] Re: GXP-2000 (steer clear)   Mimmus
• [Asterisk-Users] PRI & Fax Passthrough   Mimmus
• [Asterisk-Users] Monitoring transcoding and other heavy activities   Mimmus
• [Asterisk-Users] T1 passthrough/middleman   Mimmus
• [Asterisk-Users] T1 passthrough/middleman   Mimmus
• [Asterisk-Users] Festival RPM?   Mimmus
• [Asterisk-Users] Festival RPM?   Mimmus
• [Asterisk-Users] GXP-2000 1.1.0.13 Issues   Mimmus
• [Asterisk-Users] GXP-2000 and Configdownload via TFTP   Mimmus
• [Asterisk-Users] SIP call disconnected after answer   Mimmus
• [Asterisk-Users] Sangoma driver and zaptel   Mimmus
• [Asterisk-Users] MWI not working   Mimmus
• [Asterisk-Users] Snom 360 doesn't register after reboot   Mimmus
• [Asterisk-Users] zapata.conf: recent changes?   Mimmus
• [Asterisk-Users] syntax error   Mimmus
• [Asterisk-Users] zapata.conf: recent changes?   Mimmus
• [Asterisk-Users] zapata.conf: recent changes?   Mimmus
• [Asterisk-Users] syntax error   Mimmus
• [Asterisk-Users] zapata.conf: recent changes?   Mimmus
• [Asterisk-Users] zapata.conf: recent changes?   Mimmus
• [Asterisk-Users] Got SUBSCRIBE for extensions without hint   Mimmus
• [Asterisk-Users] syntax error   Mimmus
• [Asterisk-Users] Snom 360 with Firmware 6.1?   Mimmus
• [Asterisk-Users] Trunk failover   Mimmus
• [Asterisk-Users] ISDN   Mimmus
• [Asterisk-Users] Trixbox maunual configuration   Mimmus
• [Asterisk-Users] Very bad quality with AVM Fritz!card PCI andchan_capi   Mimmus
• [Asterisk-Users] Slightly OT: SQL query to find max load   Mimmus
• [Asterisk-Users] iax2 group pickup   Mimmus
• Fw: [Asterisk-Users] Compiling chan_bluetooth   Danko Miocevic
• Fw: [Asterisk-Users] Compiling chan_bluetooth   Danko Miocevic
• [Asterisk-Users] collect call   Virmones Pereira Tavares de Miranda
• [Asterisk-Users] SPA-2002 call HANGUP. May be a SIP bug.   Dmytro Mishchenko
• [Asterisk-Users] SPA-2002 call HANGUP. May be a SIP bug.   Dmytro Mishchenko
• [Asterisk-Users] very slow network from GXP-2000 switch port   Louis-David Mitterrand
• [Asterisk-Users] Re: GXP-2000 (steer clear)   Louis-David Mitterrand
• [Asterisk-Users] Re: GXP-2000 (steer clear)   Louis-David Mitterrand
• [Asterisk-Users] ST-2030 reseller (was: Re: GXP-2000 (steer clear))   Louis-David Mitterrand
• [Asterisk-Users] Re: Linksys SRW224P POE Switch   Louis-David Mitterrand
• [Asterisk-Users] no ring from zap channel   Louis-David Mitterrand
• [Asterisk-Users] cheapest Cisco Smartnet contract?   Louis-David Mitterrand
• [Asterisk-Users] Local vs. toll Dial Plan   Mojo with Horan & Company, LLC
• [Asterisk-Users] Configuring Polycom 501 IP phones via the console   Mojo with Horan & Company, LLC
• [Asterisk-Users] finding mac addresses   Mojo with Horan & Company, LLC
• [Asterisk-Users] Anyone using VoIP WiFi phones?   Mojo with Horan & Company, LLC
• [Asterisk-Users] Polycom Intercom - almost there   Mojo with Horan & Company, LLC
• [Asterisk-Users] Echo and crackle   Mojo with Horan & Company, LLC
• [Asterisk-Users] record until silence, playback, repeat   Mojo with Horan & Company, LLC
• [Asterisk-Users] asterisk-stat display problems   Mojo with Horan & Company, LLC
• [Asterisk-Users] asterisk-stat display problems   Mojo with Horan & Company, LLC
• [Asterisk-Users] Voicemail volume adjustment   Mojo with Horan & Company, LLC
• [Asterisk-Users] Auto answer an IAXY how   Mojo with Horan & Company, LLC
• [Asterisk-Users] Several asterisk processes starting with safe_asterisk   Ricardo Monteiro
• [Asterisk-Users] RE: Several asterisk processes starting with safe_asterisk   Ricardo Monteiro
• [Asterisk-Users] Linksys PAP2T-NA - call goes through but phone doesn't ring   James Moore
• [Asterisk-Users] Good explanation somewhere of SIP security?   James Moore
• [Asterisk-Users] Call stays mute   Luciano Moreira
• [Asterisk-Users] Wierd bug with MD3200   Luciano Moreira
• [Asterisk-Users] gsm file   Victor Moreno
• [Asterisk-Users] voicemail issue   Victor Moreno
• [Asterisk-Users] How to retrieve voicemail   Victor Moreno
• [Asterisk-Users] Bug in Voicemail ??   Victor Moreno
• [Asterisk-Users] timeout 't'   Victor Moreno
• [Asterisk-Users] new asterisk server...welcome message cut off   Rod Morison
• [Asterisk-Users] new asterisk server...welcome message cut off   Rod Morison
• [Asterisk-Users] x100p buying advice   Rod Morison
• [Asterisk-Users] to china: good voip service providers?   John Morris
• [Asterisk-Users] "Say" Applications fail   Jon Mosier
• [Asterisk-Users] Re: Explicit Dialplan Exit   Tony Mountifield
• [Asterisk-Users] Re: Forcing Marker bit   Tony Mountifield
• [Asterisk-Users] Re: Forcing Marker bit   Tony Mountifield
• [Asterisk-Users] Re: MeetMe - Annouce user join/leave without recording the name   Tony Mountifield
• [Asterisk-Users] Re: AGI Stderr   Tony Mountifield
• [Asterisk-Users] Re: delay in MeetMe   Tony Mountifield
• [Asterisk-Users] Re: Executing a Function from AGI   Tony Mountifield
• [Asterisk-Users] meetme public   Marco Mouta
• [Asterisk-Users] Asterisk Users Group - Portugal   Marco Mouta
• [Asterisk-Users] Asterisk Users Group - Portugal   Marco Mouta
• [Asterisk-Users] Asterisk Users Group - Portugal   Marco Mouta
• [Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!   Marco Mouta
• [Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!   Marco Mouta
• [Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!   Marco Mouta
• [Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!   Marco Mouta
• [Asterisk-Users] Any one with sending and receiving Sucessfull SMS PTSN Portugal?   Marco Mouta
• [Asterisk-Users] DTMF and ivr systems   Marco Mouta
• [Asterisk-Users] New Digium Card b410p   Marco Mouta
• Re: [Asterisk-Users] ISDN: 3° incoming call   Marco Mouta
• Re: [Asterisk-Users] ISDN: 3° incoming call   Marco Mouta
• [Asterisk-Users] Soekris net4801 and IAXy dhcp issue   Juan Luis Moyano
• [Asterisk-Users] Soekris net4801-50 + IAXY   Juan Luis Moyano
• [Asterisk-Users] registration at Voipbuster times out   Remko Muis
• [Asterisk-Users] SIP -> PSTN calls not connecting properly   Ronan Mullally
• [Asterisk-Users] How to configure asterisk to emulate FXO signaling ?   Carlos Munoz
• [Asterisk-Users] How to configure asterisk to emulate FXO signaling ?   Carlos Munoz
• [Asterisk-Users] How to configure asterisk to emulate FXO signaling ?   Carlos Munoz
• [Asterisk-Users] (no subject)   Eduardo Munoz
• [Asterisk-Users] Voicemail cut short?   Quinn Murphy
• [Asterisk-Users] AEL2 and CID   Steve Murphy
• [Asterisk-Users] AEL Status   Steve Murphy
• [Asterisk-Users] Weird Can-Reinvite problem   Brett N
• [Asterisk-Users] Weird Can-Reinvite problem   Brett N
• [Asterisk-Users] password on radius authentication   Dennis Nacino
• [Asterisk-Users] 302 Redirecting support   Sherif Nagy
• [Asterisk-Users] Config Revision Control   Dinesh Nair
• [Asterisk-Users] Attended transfer and queue   Dinesh Nair
• [Asterisk-Users] Attended transfer and queue   Dinesh Nair
• [Asterisk-Users] Attended transfer and queue   Dinesh Nair
• [Asterisk-Users] IAX2 Vs SIP cpu load   Dinesh Nair
• [Asterisk-Users] PRI Issue - Calls being rejected with unacceptable channel   Dinesh Nair
• [Asterisk-Users] Hitting * in a queue call hangs up?   Dinesh Nair
• [Asterisk-Users] Gizmo and Asterisk analysis   Dinesh Nair
• [Asterisk-Users] sip   Rajeev Natarajan
• [Asterisk-Users] Email notification   Rajeev Natarajan
• [Asterisk-Users] Can Asterisk Send a TEL URI INVITE?   Grady Neely
• [Asterisk-Users] Can Asterisk Send a TEL URI INVITE?   Grady Neely
• [Asterisk-Users] Packet8 and Asterisk, do they play nice?   Grady Neely
• [Asterisk-Users] Meetme + Sangoma issue?   Jason T. Nelson
• [Asterisk-Users] Re: Linksys PAP2T-NA - call goes through but phone doesn't ring   Stewart Nelson
• [Asterisk-Users] Canreinvite   Il Neofita
• [Asterisk-Users] Canreinvite   Il Neofita
• [Asterisk-Users] Canreinvite   Il Neofita
• [Asterisk-Users] Canreinvite   Il Neofita
• [Asterisk-Users] WebPhone   Il Neofita
• [Asterisk-Users] Sangoma A200 hangup detection   chan (Alpha Trilogies Networks)
• [Asterisk-Users] Re: Can I enter an extension to dial while voicemail is playing?   Leah Newmark
• [Asterisk-Users] User Loses Ability to Make Outgoing Calls   Leah Newmark
• [Asterisk-Users] User Loses Ability to Make Outgoing Calls   Leah Newmark
• [Asterisk-Users] Re: User Loses Ability to Make Outgoing Calls   Leah Newmark
• [Asterisk-Users] Re: User Loses Ability to Make Outgoing Calls   Leah Newmark
• [Asterisk-Users] Re: User Loses Ability to Make Outgoing Calls   Leah Newmark
• [Asterisk-Users] Re: User Loses Ability to Make Outgoing Calls   Leah Newmark
• [Asterisk-Users] Re: Changing standard Voicemail behavior   Leah Newmark
• [Asterisk-Users]Asterisk <-IP-> Siemens HiPath 3750   Nguyen
• [Asterisk-Users]Asterisk <-IP-> Siemens HiPath 3750   Nguyen
• [Asterisk-Users]Asterisk <-IP-> Siemens HiPath 3750   Nguyen
• Fwd: [Asterisk-Users]Asterisk <-IP-> Siemens HiPath 3750   Nguyen
• [Asterisk-Users] spa3102 vs spa3000 differences?   Darren Nickerson
• [Asterisk-Users] Delay on calls   Marnus van Niekerk
• [Asterisk-Users] Simple Speeddial AGI   Marnus van Niekerk
• [Asterisk-Users] (no subject)   Ninneman, Tj
• [Asterisk-Users] UNSUBSCRIBE   Joakim Nordberg
• [Asterisk-Users] TDM PCI Master Abort   John Novack
• [Asterisk-Users] GXP-2000   John Novack
• [Asterisk-Users] OLD PA system.   John Novack
• [Asterisk-Users] Trying to find good VOIP provider.   John Novack
• [Asterisk-Users] Callstatus on bridge IAX2 <-> ZAPTEL is always "answer" even if the call fails   John Novack
• [Asterisk-Users] Callstatus on bridge IAX2 <-> ZAPTEL is always "answer" even if the call fails   John Novack
• [Asterisk-Users] FXO for PSTN   John Novack
• [Asterisk-Users] (no subject)   John Novack
• [Asterisk-Users] Re: Digium Hardware Reliability   John Novack
• [Asterisk-Users] music on hold Madplay and Files not working   Jordan Novak
• [Asterisk-Users] Where has the outbound call directory gone   Jordan Novak
• [Asterisk-Users] Trixbox maunual configuration   Jordan Novak
• [Asterisk-Users] IAX2 Destroying channel to avoid deadlock   Jordan Novak
• [Asterisk-Users] Really need some help on IAX2 destroy to prevent deadlock   Jordan Novak
• [Asterisk-Users] Queue - Log if caller disconnects   Jordan Novak
• [Asterisk-Users] Asterisk server   Andrew Nowrot
• [Asterisk-Users] Asterisk server   Andrew Nowrot
• [Asterisk-Users] Call length limitation   Andrew Nowrot
• [Asterisk-Users] Call length limitation   Andrew Nowrot
• [Asterisk-Users] Call length limitation   Andrew Nowrot
• [Asterisk-Users] Call length limitation   Andrew Nowrot
• [Asterisk-Users] Call length limitation   Andrew Nowrot
• [Asterisk-Users] Call length limitation   Andrew Nowrot
• [Asterisk-Users] Digium TE410P configuration to connect with CIsco 3800   Massimo Nuvoli
• [Asterisk-Users] Re: Digium TE410P configuration to connect with CIsco 3800   Massimo Nuvoli
• [Asterisk-Users] hangup lag causing the answering of already answered calls   Carey O'Shea
• [Asterisk-Users] hangup lag causing the answering of already answered calls   Carey O'Shea
• [Asterisk-Users] hangup lag causing the answering of already answered calls   Carey O'Shea
• [Asterisk-Users] Which application to open Zap channel?   Carey O'Shea
• [Asterisk-Users] Which application to open Zap channel?   Carey O'Shea
• [Asterisk-Users] Which application to open Zap channel?   Carey O'Shea
• [Asterisk-Users] hangup lag causing the answering of already answered calls   Carey O'Shea
• [Asterisk-Users] hangup lag causing the answering of already answered calls   Carey O'Shea
• [Asterisk-Users] TDM400P bad echo problem, tried lots of things   Carey O'Shea
• [Asterisk-Users] Dead FXO Interface?   Blake OPS
• [Asterisk-Users] Is the current G729 compatible with Asterisk trunk?   Obelix
• [Asterisk-Users] Asterisk 1.4 on schedule?   Obelix
• [Asterisk-Users] How to use G729 decoded voice files?   Obelix
• [Asterisk-Users] problem dialing out thru sip - using isdn on internal   Kai Ober
• [Asterisk-Users] problem dialing out thru sip - using isdn on internal   Kai Ober
• [Asterisk-Users] sound quality problem on mISDN   Kai Ober
• [Asterisk-Users] sound quality problem on mISDN   Kai Ober
• [Asterisk-Users] isdn and PARK   Kai Ober
• [Asterisk-Users] CALLERID problems asterisk segfaults   Kai Ober
• [Asterisk-Users] Meetme max users   Kai Ober
• [Asterisk-Users] beronet BNS40 led blinking: not working or not connected?   Kai Ober
• [Asterisk-Users] BLINDTRANSFER   Kai Ober
• [Asterisk-Users] SIP Softphone on Thinclient?   Vitaly Oborsky
• [Asterisk-Users]Asterisk <-IP-> Siemens HiPath 3750   Ohad.Levy at infineon.com
• [Asterisk-Users] SIP, Microsoft RTC, and Originate problem   Ohad.Levy at infineon.com
• [Asterisk-Users] voip to voip bridge   Ohad.Levy at infineon.com
• [Asterisk-Users] SIP, Microsoft RTC, and Originate problem   Ohad.Levy at infineon.com
• [Asterisk-Users] Fran Oliveira desea chatear   Fran Oliveira
• [Asterisk-Users] sip channel monitoring   Fran Oliveira
• [Asterisk-Users] Dropped SIP connections never being closed?   Olivier
• [Asterisk-Users] Playback welcome message while phones ring, please help   Olivier
• [Asterisk-Users] ISDN BRI (I.430) over ethernet   Olivier
• [Asterisk-Users] ISDN BRI (I.430) over ethernet   Olivier
• [Asterisk-Users] ISDN BRI (I.430) over ethernet   Olivier
• [Asterisk-Users] How to check NAT behaviour before installing Asterisk   Olivier
• [Asterisk-Users] How would you tet a FastAGI script   Olivier
• [Asterisk-Users] Which is the best user GUI ?   Olivier
• [Asterisk-Users] Testing a FastAGI script   Olivier
• [Asterisk-Users] Ztdummy and Debian on Intel Macmini   Olivier
• [Asterisk-Users] Ztdummy and Debian on Intel Macmini   Olivier
• [Asterisk-Users] bristuff hangup issue   Olivier
• [Asterisk-Users] Strange Zaptel issue   J. Oquendo
• [Asterisk-Users] Compiling zaptel on FC5   Chad Osmond
• [Asterisk-Users] T1 Copper or T1 Fiber Line   Chad Osmond
• [Asterisk-Users] Block access to number@domain.com   Florian Overkamp
• [Asterisk-Users] Hard drive write cache   Florian Overkamp
• [Asterisk-Users] MOS Scores and LCR   Florian Overkamp
• [Asterisk-Users] MOS Scores and LCR   Florian Overkamp
• [Asterisk-Users] isdn-data over iax   Florian Overkamp
• [Asterisk-Users] Echo Cancellation   Florian Overkamp
• [Asterisk-Users] isdn-data over iax   Florian Overkamp
• [Asterisk-Users] DTMF Twist   Anil Kumar P
• [Asterisk-Users] Voip wifi text pager   Andres Paglayan
• [Asterisk-Users] Need to track dropped calls   George Pajari
• [Asterisk-Users] How to set overlap dial timeout in bristuff zaptel?   Benoit Panizzon
• [Asterisk-Users] Config Revision Control   Tim Panton
• [Asterisk-Users] Problem with IAX2 dialin with portunity   Tim Panton
• [Asterisk-Users] Prices of g729 codec   Tim Panton
• [Asterisk-Users] Outgoing call bridging   Tim Panton
• [Asterisk-Users] Outgoing call bridging   Tim Panton
• [Asterisk-Users] JIAX status   Tim Panton
• [Asterisk-Users] GXP-2000 Audio Quality   Tim Panton
• [Asterisk-Users] WRTG54GS Capacity   Tim Panton
• [Asterisk-Users] WRTG54GS Capacity   Tim Panton
• [Asterisk-Users] WRTG54GS Capacity   Tim Panton
• [Asterisk-Users] WRTG54GS Capacity   Tim Panton
• [Asterisk-Users] WRTG54GS Capacity   Tim Panton
• [Asterisk-Users] GXP-2000 Audio Quality   Tim Panton
• [Asterisk-Users] MOS Scores and LCR   Tim Panton
• [Asterisk-Users] troubleshooting echo on speakerphone   Tim Panton
• [Asterisk-Users] WebPhone   Tim Panton
• [Asterisk-Users] WebPhone   Tim Panton
• [Asterisk-Users] Suggested Phone   Tim Panton
• [Asterisk-Users] Realtime SIP Registrations   Tim Panton
• [Asterisk-Users] Standard Sound Files Distortion   Tim Panton
• : Re: [Asterisk-Users] Upgrade ONLY asterisk from an AAH install   David K Parker
• [Asterisk-Users] DID from Latvia?   David K Parker
• [Asterisk-Users] DID from Latvia?   David K Parker
• [Asterisk-Users] DID from Latvia?   David K Parker
• [Asterisk-Users] DID from Latvia?   David K Parker
• [Asterisk-Users] Sipura SPA-941 not available after Asterisk & Freepbx upgrade   David K Parker
• [Asterisk-Users] Re: Sipura SPA-941 not available after Asterisk & Freepbx upgrade   David K Parker
• Re: [Asterisk-Users] Can´t send emails   David K Parker
• [Asterisk-Users] Recurring Wakeup Call Schedule & play Weather Forecast   David K Parker
• [Asterisk-Users] weather   David K Parker
• [Asterisk-Users] Broken firewall or brain damaged admin?   David K Parker
• [Asterisk-Users] grandstream GXV-3000   Alvaro Parres
• [Asterisk-Users] Sip transfer, Sip on hold   Nicola Pascelupo
• [Asterisk-Users] Help with compilation of app_conference in x86_64   Patrick
• [Asterisk-Users] transfer & other features   Patrick
• [Asterisk-Users] Meetme versus app_conference   Patrick
• [Asterisk-Users] Asterisk chroot   Patrick
• [Asterisk-Users] GXP-2000   Patrick
• [Asterisk-Users] Re: revisit to legacy PBX and CID over PRI   Patrick
• [Asterisk-Users] IAX2 Vs SIP cpu load   Patrick
• [Asterisk-Users] GXP-2000 and Configdownload via TFTP   Patrick
• [Asterisk-Users] Zaptel HZ Warning   Patrick
• [Asterisk-Users] Call limit function on sip channel to external pop   Patrick
• [Asterisk-Users] Conferencing with multiple servers   Patrick
• [Asterisk-Users] Meetme max users   Patrick
• [Asterisk-Users] DUNDi Not Able to Handle ComplexFailoverSituations   Patrick
• [Asterisk-Users] DUNDi Not Able to HandleComplexFailoverSituations   Patrick
• [Asterisk-Users] Meetme max users   Patrick
• [Asterisk-Users] Realtime patch   Patrick
• [Asterisk-Users] SIP reinvite still does not occour   Patrick
• [Asterisk-Users] Prices of g729 codec   Paul
• [Asterisk-Users] Prices of g729 codec   Paul
• [Asterisk-Users] Vonage and FXO   Paul
• [Asterisk-Users] Vonage and FXO   Paul
• [Asterisk-Users] TTS to read from Database   Paul
• [Asterisk-Users] Trying to find good VOIP provider.   Nikolay Pavlov
• [Asterisk-Users] H.323 soft phone known to be run with asterisk.   Pawel
• [Asterisk-Users] H.323 soft phone known to be run with asterisk.   Pawel
• [Asterisk-Users] can Asterisk act as a H.323 Gatekeeper.   Pawel
• [Asterisk-Users] can Asterisk act as a H.323 Gatekeeper?   Pawel
• [Asterisk-Users] Call Queue NOT using RoundRobin ?!?   Aaron Paxson
• [Asterisk-Users] Call Queue NOT using RoundRobin ?!?   Aaron Paxson
• [Asterisk-Users] Call Queue NOT using RoundRobin ?!?   Aaron Paxson
• [Asterisk-Users] Call Queue NOT using RoundRobin ?!?   Aaron Paxson
• [Asterisk-Users] Cannot get back chan_zap.so module!??   Aaron Paxson
• [Asterisk-Users] Switchtype   Aaron Paxson
• [Asterisk-Users] Cannot get back chan_zap.so module!??   Aaron Paxson
• [Asterisk-Users] Routing inboud from ISDN to second * server.   Thomas Laurids Pedersen
• [Asterisk-Users] Routing inboud from ISDN to second * server.   Thomas Laurids Pedersen
• [Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!   Francesco Peeters
• [Asterisk-Users] 2 or more ISDN cards: which comes first ??   Francesco Peeters
• [Asterisk-Users] Problems with hangup on TE110P and "Unexpected Channel selection 3" messages   Daren Pereira
• [Asterisk-Users] regexp issue   Joao Pereira
• [Asterisk-Users] planet VIP 152 T   Joao Pereira
• [Asterisk-Users] Prices of g729 codec   Erick Perez
• [Asterisk-Users] Restricting amount of incoming calls   Erick Perez
• [Asterisk-Users] Asterisk 1.2.8 - WARNING[28769]: rtp.c:500 ast_rtp_read: Forcing Marker bit, because SSRC has changed   Erick Perez
• [Asterisk-Users] Meetme versus app_conference   Erick Perez
• [Asterisk-Users] Help with compilation of app_conference in x86_64   Erick Perez
• [Asterisk-Users] Meetme versus app_conference   Erick Perez
• [Asterisk-Users] Compiling VD_app_conference for x86_64   Erick Perez
• [Asterisk-Users] Compiling VD_app_conference for x86_64   Erick Perez
• [Asterisk-Users] Mixing meetme conferences   Erick Perez
• [Asterisk-Users] Mixing meetme conferences   Erick Perez
• [Asterisk-Users] wav49 size for a 3 minute voicemail   Erick Perez
• [Asterisk-Users] Many asterisk server behind a redirector?   Erick Perez
• [Asterisk-Users] Many asterisk server behind a redirector?   Erick Perez
• [Asterisk-Users] bug? asterisk -rx "show dialplan default"   Erick Perez
• [Asterisk-Users] APIC error on CPU0: 60(60) and asterisk crashes   Erick Perez
• [Asterisk-Users] Asterisk + Zimbra when?   Erick Perez
• [Asterisk-Users] Running a poll server with asterisk   Erick Perez
• [Asterisk-Users] Asterisk + Zimbra when?   Erick Perez
• [Asterisk-Users] Can this config sustain 30 users?   Erick Perez
• [Asterisk-Users] Help with Audicodes MP-104   Erick Perez
• [Asterisk-Users] Can this config sustain 30 users?   Erick Perez
• [Asterisk-Users] Can this config sustain 30 users?   Erick Perez
• [Asterisk-Users] Can this config sustain 30 users?   Erick Perez
• [Asterisk-Users] Can this config sustain 30 users?   Erick Perez
• [Asterisk-Users] OPENSER / SER and Asterisk   Erick Perez
• [Asterisk-Users] Echo Cancel with sangoma o digium   Erick Perez
• [Asterisk-Users] Odd Asterisk Stress Test Results   Erick Perez
• [Asterisk-Users] Avaya phone 4610sw message waiting indicator and other settings   Erick Perez
• [Asterisk-Users] Re: Avaya phone 4610sw message waiting indicator and other settings   Erick Perez
• [Asterisk-Users] Re: Avaya phone 4610sw message waiting indicator and other settings   Erick Perez
• [Asterisk-Users] Re: Avaya phone 4610sw message waiting indicator and other settings   Erick Perez
• [Asterisk-Users] h263 Video Support Questions   Erick Perez
• [Asterisk-Users] WebPhone   Erick Perez
• [Asterisk-Users] How to redirect an incoming call to an external phone numer   Infobox Peru
• [Asterisk-Users] Asterisk Users Group - Portugal   Infobox Peru
• [Asterisk-Users] Sound playback problems   Scott Pettit
• [Asterisk-Users] increase the volume ?   Noc Phibee
• [Asterisk-Users] increase the volume ?   Noc Phibee
• [Asterisk-Users] increase the volume ?   Noc Phibee
• [Asterisk-Users] increase the volume ?   Noc Phibee
• [Asterisk-Users] Avaya 4624 Ip phone   Mark Phillips
• [Asterisk-Users] Avaya 4624 Ip phone   Mark Phillips
• [Asterisk-Users] Re: [asterisk-biz] UK Male English Voices   Mark Phillips
• [Asterisk-Users] Anyone know anything about VoiceWing?   Mark Phillips
• [Asterisk-Users] Cell gateway for T-Mobile US??   Mark Phillips
• [Asterisk-Users] What is Echo?   Mark Phillips
• [Asterisk-Users] FAX + Digium + SpanDSP   Pibix
• [Asterisk-Users] early session audio on zap channel   Rosario Pingaro
• [Asterisk-Users] incoming call from Zap: "early audio" problem   Rosario Pingaro
• [Asterisk-Users] incoming call from Zap: "early audio" problem   Rosario Pingaro
• [Asterisk-Users] dss1 progressing message on zap channel   Rosario Pingaro
• [Asterisk-Users] VoIP Cheap & Asterisk   Scott Pinhorne
• [Asterisk-Users] how to decrease answer time !   William Piper
• [Asterisk-Users] How to redirect an incoming call to an external phone numer   William Piper
• [Asterisk-Users] DID from Latvia?   William Piper
• [Asterisk-Users] All non US 48 area codes?   William Piper
• [Asterisk-Users] fine-tuning asterisk questions   William Piper
• [Asterisk-Users] fine-tuning asterisk questions   William Piper
• [Asterisk-Users] fine-tuning asterisk questions   William Piper
• [Asterisk-Users] fine-tuning asterisk questions   William Piper
• [Asterisk-Users] Dialstatus   William Piper
• [Asterisk-Users] Re: fine-tuning asterisk questions   William Piper
• [Asterisk-Users] Plainvoip problem.   William Piper
• [Asterisk-Users] FreePBX 2.1.0: Manually rewriting   William Piper
• [Asterisk-Users] Virtual PBX Billing and Management Software   William Piper
• [Asterisk-Users] g729 or another   William Piper
• [Asterisk-Users] shutting down a mysql server renders cdr_mysql dead and asterisk nolonger makes or receives calls   William Piper
• [Asterisk-Users] Asterisk as Wholesale   William Piper
• [Asterisk-Users] Asterisk as Wholesale   William Piper
• [Asterisk-Users] Re: g729 or another   William Piper
• [Asterisk-Users] SIP call disconnected after answer   William Piper
• [Asterisk-Users] Asterisk Realtime and SIP Registration   William Piper
• [Asterisk-Users] dial if   William Piper
• [Asterisk-Users] "sip show inuse" is useless!   William Piper
• [Asterisk-Users] How to use a data T-1?   William Piper
• [Asterisk-Users] How to use a data T-1?   William Piper
• [Asterisk-Users] Looking for SIP provider with minimal call setuptime   William Piper
• [Asterisk-Users] Add Country to CDR's   William Piper
• [Asterisk-Users] Add Country to CDR's   William Piper
• [Asterisk-Users] forward a call to a SIP account on a remote server   William Piper
• [Asterisk-Users] Asterisk --> BV: Incoming does not work....   William Piper
• [Asterisk-Users] Asterisk --> BV: Incoming does not work....   William Piper
• [Asterisk-Users] Call length limitation   William Piper
• [Asterisk-Users] Call length limitation   William Piper
• [Asterisk-Users] Call length limitation   William Piper
• [Asterisk-Users] Voicemail volume adjustment   William Piper
• [Asterisk-Users] asterisk shutdown   William Piper
• [Asterisk-Users] asterisk shutdown   William Piper
• [Asterisk-Users] zap calls drop suddenly + tremendous noise when answering a call   Enrico Pizzorno
• [Asterisk-Users] FXO X100P   Pierre du Plessis
• [Asterisk-Users] DTMF feedthru again...   Don Pobanz
• [Asterisk-Users] no dialtone on channel banks   Don Pobanz
• [Asterisk-Users] dial plan return values   Mark Price
• [Asterisk-Users] multiple authentication realms   Mark Price
• [Asterisk-Users] Voice calls sent to fax extension   Paul A. Pringle
• [Asterisk-Users] RE: Voice calls sent to fax extension   Paul A. Pringle
• [Asterisk-Users] Work required - modify Asterisk + SEMS   Mike Puchol
• [Asterisk-Users] Work required - modify Asterisk + SEMS   Mike Puchol
• [Asterisk-Users] Single T1 card with Echo Cancellation toworkwithDell?   Joe Pukepail
• [Asterisk-Users] Dropping incompatible voice frame   Joe Pukepail
• [Asterisk-Users] Asterisk h323   Grigoriy Puzankin
• [Asterisk-Users] use AT320 international call   Min Qiu
• [Asterisk-Users] use AT320 international call   Min Qiu
• [Asterisk-Users] use AT320 international call   Min Qiu
• [Asterisk-Users] Anyone know anything about VoiceWing?   Radcliffe
• [Asterisk-Users] SE Michigan asterisk users group   Jon Radon
• [Asterisk-Users] free sun boxes   RandyW
• [Asterisk-Users] ECHO Tutorial   RandyW
• [Asterisk-Users] SNOM Softphone on windows 2000   RandyW
• [Asterisk-Users] OPENSER / SER and Asterisk   Santosh Rao
• [Asterisk-Users] OPENSER / SER and Asterisk   Santosh Rao
• [Asterisk-Users] SIP Presence   Faris Raouf
• [Asterisk-Users] SIP Presence   Faris Raouf
• [Asterisk-Users] Connect 2 Asterisk Servers via PRI   Bruce Reeves
• [Asterisk-Users] Voice Mail or MWI notify as a (windows) tray icon   Bruce Reeves
• [Asterisk-Users] Config Revision Control   Bruce Reeves
• [Asterisk-Users] Config Revision Control   Bruce Reeves
• [Asterisk-Users] Config Revision Control   Bruce Reeves
• [Asterisk-Users] Config Revision Control   Bruce Reeves
• [Asterisk-Users] TDM Fax Problems   Bruce Reeves
• [Asterisk-Users] TDM Fax Problems   Bruce Reeves
• [Asterisk-Users] '500 Internal Server' Error on SIP NOTIFY   Bruce Reeves
• [Asterisk-Users] SRST type functionality   Bruce Reeves
• [Asterisk-Users] Oh oh. Micro$oft just noticed VoIP   Bruce Reeves
• [Asterisk-Users] This is getting really annoying - re: POSTFIX   Bruce Reeves
• [Asterisk-Users] Music On Hold not working with new 1.2.7.1 install   Richard Reina
• [Asterisk-Users] Music On Hold not working with new 1.2.7.1 install   Richard Reina
• [Asterisk-Users] Music On Hold not working with new 1.2.7.1 install   Richard Reina
• [Asterisk-Users] ECHO Tutorial   Seth Remington
• [Asterisk-Users] what are the elements of a good asterisk setup?   Tyler Retzlaff
• [Asterisk-Users] disabling modules - how?   Tyler Retzlaff
• [Asterisk-Users] disabling modules - how?   Tyler Retzlaff
• [Asterisk-Users] s / i extension difficulty   Tyler Retzlaff
• [Asterisk-Users] SIP 486 "Busy Here"   Gary Reuter
• [Asterisk-Users] ECHO Tutorial   Gary Reuter
• [Asterisk-Users] remote setting - AGI or what?   Stefan Reuter
• [Asterisk-Users] Running a poll server with asterisk   Stefan Reuter
• [Asterisk-Users] "X-Asterisk-HangupCause: Normal Clearing"   Stephane Ricard
• [Asterisk-Users] recommended telephones   Ricardo
• [Asterisk-Users] Config Revision Control   Hadley Rich
• [Asterisk-Users] Config Revision Control   Hadley Rich
• [Asterisk-Users] Fun with Echo -- Follow up   Hadley Rich
• [Asterisk-Users] disabling modules - how?   Hadley Rich
• [Asterisk-Users] Audio cuts out   Gary Richardson
• [Asterisk-Users] Audio cuts out   Gary Richardson
• [Asterisk-Users] Audio cuts out   Gary Richardson
• [Asterisk-Users] Audio cuts out   Gary Richardson
• [Asterisk-Users] sip to h323 gateway ...   Gary Richardson
• [Asterisk-Users] sip to h323 gateway ...   Gary Richardson
• [Asterisk-Users] Voicemail with NFS   JR Richardson
• [Asterisk-Users] Voicemail with NFS   JR Richardson
• [Asterisk-Users] Voicemail with NFS (working, I think)   JR Richardson
• [Asterisk-Users] voip-magazine article "Using DUNDi with a Cluster of Asterisk Servers"   JR Richardson
• [Asterisk-Users] G729, voicemail, no codec_g729   Steven Ringwald
• [Asterisk-Users] Config Revision Control   Steven Ringwald
• [Asterisk-Users] Compiling zaptel on FC5   Steven Ringwald
• [Asterisk-Users] IAX2 Vs SIP cpu load   Steven Ringwald
• [Asterisk-Users] Snom high SIP ping time   Steven Ringwald
• [Asterisk-Users] Comedian Mail not deleting .txt file   Steven Ringwald
• [Asterisk-Users] how to hang the zap channel   Steven Ringwald
• [Asterisk-Users] Include Text file in Dial Plan   Steven Ringwald
• [Asterisk-Users] Mail loop?   Steven Ringwald
• [Asterisk-Users] Is there a way to reinstall the AMP   Yrving Rivas
• [Asterisk-Users] Why I can´t upload my wav file?   Yrving Rivas
• RE: [Asterisk-Users] In other words (Why I can´t upload my wav file?)   Yrving Rivas
• [Asterisk-Users] Sangoma driver and zaptel   Robert Roach
• [Asterisk-Users] Web UI - Best practices?   Robert Roach
• [Asterisk-Users] skype out   Alex Robar
• Re: [Asterisk-Users] Can´t send emails   Alex Robar
• Re: [Asterisk-Users] Can´t send emails   Alex Robar
• [Asterisk-Users] Wanted: CISCO 186 ATAs   Alex Robar
• Re: [Asterisk-Users] Can´t send emails   Alex Robar
• [Asterisk-Users] weather   Alex Robar
• [Asterisk-Users] directory   Alex Robar
• [Asterisk-Users] dial pattern   Alex Robar
• [Asterisk-Users] Turning off a temporary message in voicemail   Alex Robar
• [Asterisk-Users] TrixBox   Alex Robar
• [Asterisk-Users] Don't use CDRTool From AG-projescts   Alex Robar
• [Asterisk-Users] trunk rollover   Alex Robar
• [Asterisk-Users] Remote employees using Polycom 501 lose abilityto receive incoming calls after few minutes.   Alex Robar
• [Asterisk-Users] Voicemail volume adjustment   Adam Robins
• [Asterisk-Users] Converting Voicemail wav to mp3   Anthony Rodgers
• [Asterisk-Users] Out of Office Auto Reply:   Anthony Rodgers
• [Asterisk-Users] Quality monitoring   Anthony Rodgers
• [Asterisk-Users] Out of Office Auto Reply:   Anthony Rodgers
• [Asterisk-Users] Quality monitoring   Anthony Rodgers
• [Asterisk-Users] '500 Internal Server' Error on SIP NOTIFY   Anthony Rodgers
• [Asterisk-Users] help in create user group   Andre Luiz Martins Rodrigues
• [Asterisk-Users] How to redirect an incoming call to an external phone numer   Javier Rodriguez
• [Asterisk-Users] Compiling mpg123 under 1.2.9.1 / Ubuntu 6.06   Marc Rohlfing
• AW: [Asterisk-Users] Compiling mpg123 under 1.2.9.1 / Ubuntu 6.06   Marc Rohlfing
• [Asterisk-Users] Eicon Diva Server with v3.0 drivers   Marc Rohlfing
• AW: [Asterisk-Users] Eicon Diva Server with v3.0 drivers   Marc Rohlfing
• AW: [Asterisk-Users] syntax error   Marc Rohlfing
• [Asterisk-Users] Avaya 4624 Ip phone   Carlos Rojas
• [Asterisk-Users] Avaya 4624 Ip phone   Carlos Rojas
• [Asterisk-Users] Low volume/ audio problems on TDM400 card   Carlos Rojas
• [Asterisk-Users] Which simple billing application   Carlos Rojas
• [Asterisk-Users] Low volume/ audio problems on TDM400 card   Carlos Rojas
• [Asterisk-Users] Avaya 4624 Ip phone   Gabriel Rosca
• [Asterisk-Users] Asterisk and TBCT   Eric Rousse
• [Asterisk-Users] Linksys PAP2T-NA - call goes through but phone doesn't ring   Andre Ruiz
• [Asterisk-Users] Grandstream BT101/102 lost register with asterisk ?   Información Capa Tres S.L.
• [Asterisk-Users] SNOM Softphone on windows 2000   SANS
• [Asterisk-Users] statistics   Alberto Sagredo
• [Asterisk-Users] Wanted: CISCO 186 ATAs   Alberto Sagredo
• [Asterisk-Users] Compiling SVN Trunk   Alberto Sagredo
• [Asterisk-Users] Compiling SVN Trunk   Alberto Sagredo
• [Asterisk-Users] Compiling SVN Trunk   Alberto Sagredo
• [Asterisk-Users] SPA-941 Disable call waiting or Disable Call waiting via asterisk   Alberto Sagredo
• [Asterisk-Users] Asterisk h323   Alberto Sagredo
• [Asterisk-Users] echo sidetone grandstream and tdm400p   Marco Sajeva
• [Asterisk-Users] echo sidetone grandstream and tdm400p   Marco Sajeva
• [Asterisk-Users] echo sidetone grandstream and tdm400p   Marco Sajeva
• [Asterisk-Users] Attended call transfer with GXP-2000   Daniel Salama
• [Asterisk-Users] GXP-2000   Daniel Salama
• [Asterisk-Users] GXP-2000   Daniel Salama
• [Asterisk-Users] GXP-2000   Daniel Salama
• [Asterisk-Users] Re: GXP-2000 (steer clear)   Daniel Salama
• [Asterisk-Users] GXP-2000   Daniel Salama
• [Asterisk-Users] GXP-2000   Daniel Salama
• [Asterisk-Users] GXP-2000   Daniel Salama
• [Asterisk-Users] GXP-2000   Daniel Salama
• [Asterisk-Users] GXP-2000   Daniel Salama
• [Asterisk-Users] GXP-2000   Daniel Salama
• [Asterisk-Users] Virtual PBX Billing and Management Software   Daniel Salama
• [Asterisk-Users] GXP-2000 MultiPurpose Keys   Daniel Salama
• [Asterisk-Users] GXP-2000 MultiPurpose Keys   Daniel Salama
• [Asterisk-Users] GXP-2000 MultiPurpose Keys   Daniel Salama
• [Asterisk-Users] Asterisk as Wholesale   Daniel Salama
• [Asterisk-Users] GXP-2000   Daniel Salama
• [Asterisk-Users] GXP-2000 Audio Quality   Daniel Salama
• [Asterisk-Users] GXP-2000 Audio Quality   Daniel Salama
• [Asterisk-Users] GXP-2000 Audio Quality   Daniel Salama
• [Asterisk-Users] Comfort noise support incomplete in Asterisk (RFC 3389)   Daniel Salama
• [Asterisk-Users] Comfort noise support incomplete in Asterisk (RFC 3389)   Daniel Salama
• [Asterisk-Users] WRTG54GS Capacity   Daniel Salama
• [Asterisk-Users] WRTG54GS Capacity   Daniel Salama
• [Asterisk-Users] WRTG54GS Capacity   Daniel Salama
• [Asterisk-Users] MOS Scores and LCR   Daniel Salama
• [Asterisk-Users] MOS Scores and LCR   Daniel Salama
• [Asterisk-Users] WRTG54GS Capacity   Daniel Salama
• [Asterisk-Users] ECHO Tutorial   Daniel Salama
• [Asterisk-Users] Asterisk and SER   Daniel Salama
• [Asterisk-Users] GXP 2000 - BLF and Hold/Hangup Answering   Daniel Salama
• [Asterisk-Users] GXP-2000 and Shared Line Appearances   Daniel Salama
• [Asterisk-Users] Kernel 2.4 / 2.6 and timer   Daniel Salama
• [Asterisk-Users] GXP-2000 and Shared Line Appearances   Daniel Salama
• [Asterisk-Users] GXP-2000 and Shared Line Appearances   Daniel Salama
• [Asterisk-Users] ExternalIVR vs AGI   Daniel Salama
• [Asterisk-Users] Suggested Phone   Daniel Salama
• [Asterisk-Users] how to decrease answer time !   Mohammad Salaque
• [Asterisk-Users] how to identify agi crash cause   Danish Samad
• [Asterisk-Users] how to identify agi crash cause   Danish Samad
• [Asterisk-Users] agi, STREAM FILE and SIGHUP   Danish Samad
• [Asterisk-Users] recording all calls patch through asterisk   Michael Sampson
• [Asterisk-Users] Help with JIAXClient   Enrique Sanchez
• [Asterisk-Users] Compiling QuaBri cards   Olivier Saulnier
• [Asterisk-Users] Compiling QuaBri cards   Olivier Saulnier
• [Asterisk-Users] QuadBri card   Olivier Saulnier
• [Asterisk-Users] QuadBri card   Olivier Saulnier
• [Asterisk-Users] Xorcom Rapid   Olivier Saulnier
• [Asterisk-Users] Xorcom Rapid   Olivier Saulnier
• [Asterisk-Users] Xorcom Rapid   Olivier Saulnier
• [Asterisk-Users] Xorcom Rapid   Olivier Saulnier
• [Asterisk-Users] Dropping incompatible voice frame   Kevin Savoy
• [Asterisk-Users] bristuff hangup issue   Kevin Savoy
• [Asterisk-Users] Dropping incompatible voice frame   Kevin Savoy
• [Asterisk-Users] chan_capi-cm-0.6 and incoming calls problem   Armin Schindler
• [Asterisk-Users] chan_capi-cm-0.6 and incoming calls problem   Armin Schindler
• [Asterisk-Users] chan_capi-cm-0.6 and incoming calls problem   Armin Schindler
• [Asterisk-Users] chan-capi and dtmf   Armin Schindler
• [Asterisk-Users] Eicon Diva Server with v3.0 drivers   Armin Schindler
• AW: [Asterisk-Users] Eicon Diva Server with v3.0 drivers   Armin Schindler
• [Asterisk-Users] Single T1 card with Echo CancellationtoworkwithDell?   Armin Schindler
• [Asterisk-Users] Asterisk voicemail problem with isdn avm fritz!card   Armin Schindler
• [Asterisk-Users] Asterisk voicemail problem with isdn avm fritz!card   Armin Schindler
• [Asterisk-Users] Routing inboud from ISDN to second * server.   Armin Schindler
• [Asterisk-Users] 2 or more ISDN cards: which comes first ??   Armin Schindler
• [Asterisk-Users] Very bad quality with AVM Fritz!card PCI and chan_capi   Armin Schindler
• [Asterisk-Users] Very bad quality with AVM Fritz!card PCI andchan_capi   Armin Schindler
• [Asterisk-Users] Very bad quality with AVM Fritz!card PCI and chan_capi   Armin Schindler
• Re: [Asterisk-Users] ISDN: 3° incoming call   Armin Schindler
• [Asterisk-Users] ISDN: 3° incoming call   Armin Schindler
• [Asterisk-Users] CONSOLE/dsp   Juergen Schinker
• [Asterisk-Users] No reinvite - reason?   Roger Schreiter
• [Asterisk-Users] No reinvite - reason?   Roger Schreiter
• [Asterisk-Users] SIP reinvite still does not occour   Roger Schreiter
• [Asterisk-Users] SIP reinvite still does not occour   Roger Schreiter
• [Asterisk-Users] Best PRI provider, Long Island, NY   Richard Schroeder
• [Asterisk-Users] Re: Nokai E60 and E61 , working fine with Asterisk , with new access points   Markus Schuster
• [Asterisk-Users] Re: Nokai E60 and E61 , working fine with Asterisk , with new access points   Markus Schuster
• [Asterisk-Users] ipPhone and ATA with UPNP   Laurent Schweizer
• [Asterisk-Users] open source sip softphone (Window OS version )   Laurent Schweizer
• [Asterisk-Users] DID in United Arab Emirates, Iran, Kuwaiti, Iraq, Bahrain, Jordan, Saudi Arabia.   Laurent Schweizer
• [Asterisk-Users] Addon-ooh323 install problem   Richard Scobie
• [Asterisk-Users] Addon-ooh323 install problem   Richard Scobie
• [Asterisk-Users] Queues - Configuration Help needed   Jim Scott
• [Asterisk-Users] Grandstream BT101 Auto-Answer   Jon Scottorn
• [Asterisk-Users] trunk rollover   Jon Scottorn
• [Asterisk-Users] dial if   Miles Scruggs
• [Asterisk-Users] d & e options in meetme()   Miles Scruggs
• [Asterisk-Users] dial if   Miles Scruggs
• [Asterisk-Users] RxFax & Asterisk possible bug?   Sebastian
• [Asterisk-Users] Asterisk voicemail problem with isdn avm fritz!card   Benjamin Sebbah
• [Asterisk-Users] Asterisk voicemail problem with isdn avm fritz!card   Benjamin Sebbah
• [Asterisk-Users] Very bad quality with AVM Fritz!card PCI and chan_capi   Benjamin Sebbah
• [Asterisk-Users] Very bad quality with AVM Fritz!card PCI and chan_capi   Benjamin Sebbah
• [Asterisk-Users] Very bad quality with AVM Fritz!card PCI and chan_capi   Benjamin Sebbah
• [Asterisk-Users] Very bad quality with AVM Fritz!card PCI andchan_capi   Benjamin Sebbah
• [Asterisk-Users] Very bad quality with AVM Fritz!cardPCIandchan_capi   Benjamin Sebbah
• [Asterisk-Users] Integrate asterisk with Database   Vidura Senadeera
• [Asterisk-Users] New to the list.   Fran Serrano
• [Asterisk-Users] Monitor application and e-mailing attachment   Gonzalo Servat
• [Asterisk-Users] WCTDM-24xxp woes   Gonzalo Servat
• [Asterisk-Users] AEL2   Gonzalo Servat
• [Asterisk-Users] AEL2   Gonzalo Servat
• [Asterisk-Users] Time Based Goto Ifs Act Strange?   Gonzalo Servat
• [Asterisk-Users] DTMF in the middle of a call   Servetas, Andrew
• [Asterisk-Users] DTMF in the middle of a call   Servetas, Andrew
• [Asterisk-Users] FW: Asterisk Quintum A800 SIP Mode   Freddy Setiawan
• [Asterisk-Users] FW: Asterisk Quintum A800 SIP Mode   Freddy Setiawan
• [Asterisk-Users] FW: Asterisk Quintum A800 SIP Mode   Freddy Setiawan
• [Asterisk-Users] OT: Cellular boosters   Kyle Sexton
• [Asterisk-Users] Configuring Polycom 501 IP phones via the console   Curt Shaffer
• [Asterisk-Users] Vonage and FXO   Curt Shaffer
• [Asterisk-Users] Vonage and FXO   Curt Shaffer
• [Asterisk-Users] No CID on ZAP   Curt Shaffer
• [Asterisk-Users] Re: No CID on ZAP   Curt Shaffer
• [Asterisk-Users] No CID on ZAP   Curt Shaffer
• [Asterisk-Users] No CID on ZAP   Curt Shaffer
• [Asterisk-Users] rollover simulation   Curt Shaffer
• [Asterisk-Users] Zaptel dialing too fast?   Curt Shaffer
• [Asterisk-Users] massive screetch and echo from Treo 700w   Curt Shaffer
• [Asterisk-Users] Quality monitoring   Curt Shaffer
• [Asterisk-Users] Quality monitoring   Curt Shaffer
• [Asterisk-Users] RTA, jitter, MOS et al over the internet   Curt Shaffer
• [Asterisk-Users] SRST type functionality   Curt Shaffer
• [Asterisk-Users] Mail loop?   Curt Shaffer
• [Asterisk-Users] DTMF and ivr systems   Shane
• [Asterisk-Users] pickup problem   Denis Shaposhnikov
• [Asterisk-Users] pickup problem   Denis Shaposhnikov
• [Asterisk-Users] show queue ... Invalid   Denis Shaposhnikov
• [Asterisk-Users] Asterisk 1.2.9 cli "-x" doesn't flush?   Denis Shaposhnikov
• [Asterisk-Users] show queue ... Invalid   Denis Shaposhnikov
• [Asterisk-Users] show queue ... Invalid   Denis Shaposhnikov
• [Asterisk-Users] chan_sip.c: Insufficient information for SDP   Denis Shaposhnikov
• [Asterisk-Users] loading realtime peers   Sharon
• [Asterisk-Users] Asterisk/Metaswitch trunk, no inbound RTP stream on inbound calls   Clint Sharp
• [Asterisk-Users] Playback welcome message while phones ring, please help   Tim Sharp
• [Asterisk-Users] Playback welcome message while phones ring, please help   Tim Sharp
• [Asterisk-Users] No ring tone on outgoing calls   Tim Sharp
• [Asterisk-Users] No ring tone on outgoing calls   Tim Sharp
• [Asterisk-Users] Re: Can I enter an extension to dial whilevoicemail is playing?   Tim Sharp
• [Asterisk-Users] SE Michigan asterisk users group   Tim Sharp
• [Asterisk-Users] Asking for phone number to dial   Tim Sharp
• [Asterisk-Users] RE: Voice calls sent to fax extension   Tim Sharp
• [Asterisk-Users] how to delete a key from database in extensions.conf   Shaun
• [Asterisk-Users] how to delete a key from database in extensions.conf   Shaun
• [Asterisk-Users] home routers   Shaun
• [Asterisk-Users] finding mac addresses   T. Shaw
• [Asterisk-Users] troubleshooting echo on speakerphone   T. Shaw
• [Asterisk-Users] troubleshooting echo on speakerphone   T. Shaw
• [Asterisk-Users] Asking for phone number to dial   T. Shaw
• [Asterisk-Users] need help troubleshooting clipping and garbled VOIP calls   T. Shaw
• [Asterisk-Users] need help troubleshooting clipping and garbled VOIP calls   T. Shaw
• [Asterisk-Users] IAX2 debug info   T. Shaw
• [Asterisk-Users] audio streaming points different with VRRP   Shenen Shenen
• [Asterisk-Users] change of calls control with VRRP protocol   Shenen Shenen
• [Asterisk-Users] ISDN BRI (I.430) over ethernet   Shenen Shenen
• [Asterisk-Users] CLI comand to register softphones without close them:   Shenen Shenen
• [Asterisk-Users] registration SIP softphone:who is the file who makes the registration?how can I set more proxy than 1?   Shenen Shenen
• [Asterisk-Users] Registered SIP:   Shenen Shenen
• [Asterisk-Users] Database file to copy for active sessions.   Shenen Shenen
• [Asterisk-Users] Database file to copy for active sessions.   Shenen Shenen
• [Asterisk-Users] Database file to copy for active sessions.   Shenen Shenen
• [Asterisk-Users] database copy in asterisk   Shenen Shenen
• [Asterisk-Users] database copy in asterisk   Shenen Shenen
• [Asterisk-Users] SIPCALLID, but which callid?   Andreas Sikkema
• [Asterisk-Users] SIPCALLID, but which callid?   Andreas Sikkema
• [Asterisk-Users] Help with Audicodes MP-104   Mahilal Silva
• [Asterisk-Users] Help with Audicodes MP-104   Mahilal Silva
• [Asterisk-Users] MFC/R2 for Voice and Data   Moises Silva
• [Asterisk-Users] RE: Size limitations of extensions.conf   Moises Silva
• [Asterisk-Users] Dialstatus   Moises Silva
• [Asterisk-Users] Dialstatus   Moises Silva
• [Asterisk-Users] Unicall local_unblocking_expired error   Moises Silva
• [Asterisk-Users] Unicall acting really funny   Moises Silva
• [Asterisk-Users] Unicall acting really funny   Moises Silva
• [Asterisk-Users] extensions.conf   Moises Silva
• [Asterisk-Users] DTMF when using g.729   Moises Silva
• [Asterisk-Users] Transfer call via AMI or dialplan   Moises Silva
• [Asterisk-Users] sangoma unicall m2rfc   Moises Silva
• [Asterisk-Users] Transfer call via AMI or dialplan   Moises Silva
• [Asterisk-Users] sangoma unicall m2rfc   Moises Silva
• [Asterisk-Users] STUN?   Moises Silva
• [Asterisk-Users] AGI script can not print out error message to console   Moises Silva
• [Asterisk-Users] Avaya 4610sw SIP setup problem   Herchi Silviu
• [Asterisk-Users] Re: siemens pbx and asterisk   Herchi Silviu
• [Asterisk-Users] Re: Asterisk x Siemens HiPath 4000   Herchi Silviu
• [Asterisk-Users] Re: siemens pbx and asterisk   Herchi Silviu
• [Asterisk-Users] Re: Asterisk x Siemens HiPath 4000   Herchi Silviu
• [Asterisk-Users] Avaya 4610sw SIP setup problem   Herchi Silviu
• [Asterisk-Users] Avaya 4610sw SIP setup problem   Herchi Silviu
• [Asterisk-Users] HDLC Bad FCS (8)   Herchi Silviu
• [Asterisk-Users] Avaya 4610sw SIP setup problem   Herchi Silviu
• [Asterisk-Users] Avaya 4610sw SIP setup problem   Herchi Silviu
• [Asterisk-Users] Vega 50 10 FXO   Issac Simchayof
• [Asterisk-Users] Vega 50 10 FXO   Issac Simchayof
• [Asterisk-Users] FXO registration and VegaStream   Issac Simchayof
• [Asterisk-Users] FXO registration and VegaStream   Issac Simchayof
• [Asterisk-Users] FXO registration and VegaStream   Issac Simchayof
• [Asterisk-Users] FXO registration and VegaStream   Issac Simchayof
• [Asterisk-Users] MOH & Vegastream   Issac Simchayof
• [Asterisk-Users] message waiting   Issac Simchayof
• [Asterisk-Users] Rxfax with Sirrix quad BRI   Garth van Sittert
• [Asterisk-Users] Configuring Polycom 501 IP phones via the console   Kevin Smith
• [Asterisk-Users] Call-pickup function in Queue application   Kevin Smith
• [Asterisk-Users] Configuring Polycom 501 IP phones via the console   Kevin Smith
• [Asterisk-Users] Allowing multiple exchanges   Kevin Smith
• [Asterisk-Users] More Level QueueSystem   Kevin Smith
• [Asterisk-Users] Dial plan question   Kevin Smith
• [Asterisk-Users] username/auth name mismatch   Kevin Smith
• [Asterisk-Users] Polycom 601 question   Kevin Smith
• [Asterisk-Users] Polycom 601 question   Kevin Smith
• [Asterisk-Users] Auto NOTIFY   Kevin Smith
• [Asterisk-Users] Auto NOTIFY   Kevin Smith
• [Asterisk-Users] Auto NOTIFY   Kevin Smith
• [Asterisk-Users] uniden uip 200 phones lockup but rare - anyo ne seen this   Nathan C. Smith
• [Asterisk-Users] Busy Signals after hangup   Rick Smith
• [Asterisk-Users] GXP-2000 MultiPurpose Keys   Rick Smith
• [Asterisk-Users] Easiest (best?) linux distribution for dedicatedAsterisk box?   Rick Smith
• [Asterisk-Users] AstriCon Europe - Only 1 Week Away   Steven Sokol
• [Asterisk-Users] AstriCon Berlin Starts Tomorrow (Montag)   Steven Sokol
• [Asterisk-Users] AstriCon Paris Starts Wednesday   Steven Sokol
• [Asterisk-Users] AstriCon London Starts Tomorrow   Steven Sokol
• [Asterisk-Users] asterisk on AMD 64 BIT   Kanishka Somaratne
• [Asterisk-Users] skype out   Cyber Source
• [Asterisk-Users] Presentation + Asterisk Realtime doubts   Andrea Spadaccini
• [Asterisk-Users] [Repost] Asterisk realtime   Andrea Spadaccini
• [Asterisk-Users] [Repost] Asterisk realtime   Andrea Spadaccini
• [Asterisk-Users] Bug in asterisk "static" realtime?   Andrea Spadaccini
• [Asterisk-Users] Bug in asterisk "static" realtime?   Andrea Spadaccini
• [Asterisk-Users] Asterisk realtime and metrics   Andrea Spadaccini
• [Asterisk-Users] Polycom Buddies in 1.6.6   Ryan Stark
• [Asterisk-Users] Asterisk 1.2.7.1 with Polycom 501 on SIP -> Conf Calling   Mike Staver
• [Asterisk-Users] Asterisk 1.2.7.1 with Polycom 501 on SIP -> Conf Calling   Mike Staver
• [Asterisk-Users] WG: Dialplan problem with Digium tdm04p card   Frank Stefan
• [Asterisk-Users] Re: unknown host cvs.digium.com   Steven
• [Asterisk-Users] Re: Re: unknown host cvs.digium.com   Steven
• [Asterisk-Users] Re: G729, voicemail, no codec_g729   Steven
• [Asterisk-Users] Opposite iaxy?   Steven
• [Asterisk-Users] revisit to legacy PBX and CID over PRI   Steven
• [Asterisk-Users] revisit to legacy PBX and CID over PRI   Steven
• [Asterisk-Users] Re: revisit to legacy PBX and CID over PRI   Steven
• [Asterisk-Users] Re: Re: revisit to legacy PBX and CID over PRI   Steven
• [Asterisk-Users] Re: Re: revisit to legacy PBX and CID over PRI   Steven
• [Asterisk-Users] Re: Re: revisit to legacy PBX and CID over PRI   Steven
• [Asterisk-Users] Re: Re: revisit to legacy PBX and CID over PRI   Steven
• [Asterisk-Users] calleridname.agi patch to only overwrite name if it is missing   Steven
• [Asterisk-Users] Re: voiceone?   Steven
• [Asterisk-Users] Re: USB handset options for softphones   Steven
• [Asterisk-Users] me, voip.trxtel.com and early media   Steven
• [Asterisk-Users] Re: me, voip.trxtel.com and early media   Steven
• [Asterisk-Users] Re: Showing Current Calls   Steven
• [Asterisk-Users] Re: Out of Office Auto Reply:   Steven
• [Asterisk-Users] Re: Voice calls sent to fax extension   Steven
• [Asterisk-Users] Re: SE Michigan asterisk users group   Steven
• [Asterisk-Users] Re: trunk rollover   Steven
• [Asterisk-Users] FXO for PSTN   Steven
• [Asterisk-Users] Re: TE420P/TE415P?   Steven
• [Asterisk-Users] TBCT - Two B-Channel Transfer   Scott Stingel
• [Asterisk-Users] Looking for very basic example   Benjamin Stocker
• [Asterisk-Users] Asterisk Realtime and SIP Registration   Benjamin Stocker
• [Asterisk-Users] Realtime problem   Benjamin Stocker
• [Asterisk-Users] Realtime problem   Benjamin Stocker
• [Asterisk-Users] Realtime voicemail not working   Benjamin Stocker
• [Asterisk-Users] calling between contexts   Benjamin Stocker
• [Asterisk-Users] Realtime voicemail not working   Benjamin Stocker
• [Asterisk-Users] Asterisk 1.07 crash under Debian Sarge   Mark W. Stoddard
• [Asterisk-Users] Asterisk 1.07 crash under Debian Sarge   Mark W. Stoddard
• [Asterisk-Users] A@H / Trixbox Question   Johnny Stork
• [Asterisk-Users] Snom high SIP ping time   Christian Stredicke
• *** Spam *** [Asterisk-Users] Don't use CDRTool From AG-projescts   Christian Stredicke
• [Asterisk-Users] best hardphone for Asterisk?   Christian Stredicke
• *** Spam *** [Asterisk-Users] recommended telephones   Christian Stredicke
• [Asterisk-Users] SNOM Softphone on windows 2000   Christian Stredicke
• [Asterisk-Users] TE405P Dropping Calls. !! Got I-frame while link state 0   Brian Stuart
• [Asterisk-Users] PRI - Ring requested on channel errors - inbound & outbound stop working.   Dan Sully
• [Asterisk-Users] Prices of g729 codec   Leon Sun
• [Asterisk-Users] Prices of g729 codec   Leon Sun
• [Asterisk-Users] Can this config sustain 30 users?   TSS Support
• [Asterisk-Users] Voicemail volume adjustment   Technical Support
• [Asterisk-Users] Question about ring groups and ext. busy in call   Chris Sutton
• [Asterisk-Users] GXP-2000 and transferring call directly to voicemail   Chris Sutton
• [Asterisk-Users] Fun with Echo   Brian Swan
• [Asterisk-Users] Fun with Echo   Brian Swan
• [Asterisk-Users] Fun with Echo   Brian Swan
• [Asterisk-Users] Stupid question zaptel-1.2.6 vs. svn/trunk   Brian Swan
• [Asterisk-Users] Fun with Echo   Brian Swan
• [Asterisk-Users] Stupid question zaptel-1.2.6 vs. svn/trunk   Brian Swan
• [Asterisk-Users] Fun with Echo   Brian Swan
• [Asterisk-Users] Fun with Echo   Brian Swan
• [Asterisk-Users] Fun with Echo   Brian Swan
• [Asterisk-Users] Caller-ID Info with Voice Mail -- Can it display to the phone?   Brian Swan
• [Asterisk-Users] Fun with Echo -- Follow up   Brian Swan
• [Asterisk-Users] SIP -> PSTN calls not connecting properly   Brian Swan
• [Asterisk-Users] iax2 registration problems   T.S
• [Asterisk-Users] Asterisk 1.2.7.1 bad file descriptor   Administrator TOOTAI
• [Asterisk-Users] Asterisk 1.2.7.1 bad file descriptor   Administrator TOOTAI
• [Asterisk-Users] RE: Snom 360 doesn't register after reboot   Usman Tahir
• [Asterisk-Users] STUN?   Raymond Tant
• [Asterisk-Users] CISCO POE   Tarpo, Louie
• [Asterisk-Users]Asterisk <-IP-> Siemens HiPath 3750   Viktor Tatianin
• [Asterisk-Users]Asterisk <-IP-> Siemens HiPath 3750   Viktor Tatianin
• [Asterisk-Users] siemens pbx and asterisk   Viktor Tatianin
• [Asterisk-Users] Asterisk 1.2.9.1 and 1.0.11.1 Released -- Security Fix   Asterisk Development Team
• [Asterisk-Users] Asterisk-addons 1.2.3 released   The Asterisk Development Team
• [Asterisk-Users] Asterisk 1.2.9 and 1.0.11 Released -- Security Fix   The Asterisk Development Team
• [Asterisk-Users] Polycom SIP 1.6.6   MBIT Technologies
• [Asterisk-Users] Asterisk, mISDN and a Fritz card   MBIT Technologies
• [Asterisk-Users] Asterisk, mISDN and a Fritz card -- kernel crashes   MBIT Technologies
• [Asterisk-Users] Re: ISDN BRI NetJet   MBIT Technologies
• [Asterisk-Users] PRI Broke on 1.2.9.1?   Chris Teesdale
• [Asterisk-Users] New => Asterisk Queue (and CDR) Log Analyzer   Earl Terwilliger
• [Asterisk-Users] Echo Problem with T411P   James Texter
• [Asterisk-Users] when I press "transfer" -> blind -> 700 . The user is not able to hear what extension the call was parked on   James Texter
• [Asterisk-Users] RE: 2 rings after making a phone call   Ash Thakrar
• [Asterisk-Users] Re: wctdm.c RING_DEBOUNCE   Ash Thakrar
• [Asterisk-Users] Realtime SIP Registrations   David Thomas
• [Asterisk-Users] Realtime SIP Registrations   David Thomas
• FW: [Asterisk-Users] syntax error   Rob Thomas
• FW: [Asterisk-Users] syntax error   Rob Thomas
• FW: [Asterisk-Users] zapata.conf: recent changes?   Rob Thomas
• [Asterisk-Users] syntax error   Rob Thomas
• [Asterisk-Users] syntax error   Rob Thomas
• [Asterisk-Users] Asterisk Startups   Rob Thomas
• [Asterisk-Users] FW: Quality of Asterisk   Deillon Thomas-WTD008
• [Asterisk-Users] Bus Mastering   Deillon Thomas-WTD008
• [Asterisk-Users] Out of Office Auto Reply:   Mitch Thompson
• [Asterisk-Users] Re: Executing a Function from AGI   Stefan Tichy
• [Asterisk-Users] Re: Executing a Function from AGI   Stefan Tichy
• [Asterisk-Users] Re: ISDN BRI NetJet   Stefan Tichy
• [Asterisk-Users] T1 delivered via Copper   Mark Tinka
• [Asterisk-Users] ooh323 issues   Mark Tinka
• [Asterisk-Users] Integrating H.323 gateways with Asterisk?   Mark Tinka
• [Asterisk-Users] Zork and Asterisk   John Todd
• [Asterisk-Users] Re: Zork and Asterisk   John Todd
• [Asterisk-Users] Linksys SRW224P POE Switch   Tom
• [Asterisk-Users] Linksys SRW224P POE Switch   Tom
• [Asterisk-Users] Debian Sarge or CentOS4.3   Tom
• [Asterisk-Users] Re: XML to monitor queues on Cisco display ?   Brent Torrenga
• [Asterisk-Users] RE: Size limitations of extensions.conf   Brent Torrenga
• [Asterisk-Users] SIP One-way audio: == Forcing Marker bit, because SSRC has changed - trxtel.com   Brent Torrenga
• [Asterisk-Users] Directory - First Name/Last Name - How to use both? a@h?   Brent Torrenga
• [Asterisk-Users] Re: Directory - First Name/Last Name - How to use both? a@h?   Brent Torrenga
• [Asterisk-Users] RE: Auto-pickup cisco phones   Brent Torrenga
• [Asterisk-Users] Limited Queue Overflow Puzzle   Steve Totaro
• [Asterisk-Users] Multiple SIP Accounts Between Asterisk Boxes (Unreachable)   Steve Totaro
• [Asterisk-Users] quad t1 / 1U rack server combos   Steve Totaro
• [Asterisk-Users] Audio cuts out   Steve Totaro
• [Asterisk-Users] Audio cuts out   Steve Totaro
• [Asterisk-Users] Asterisk and multiple SIP registrations to the same host (team/oej/register)   Steve Totaro
• [Asterisk-Users] Need to track dropped calls   Steve Totaro
• [Asterisk-Users] Need to track dropped calls   Steve Totaro
• [Asterisk-Users] 100 lines PBX + system config - repost   Steve Totaro
• [Asterisk-Users] Asterisk and multiple SIP registrations to the same host (team/oej/register)   Steve Totaro
• [Asterisk-Users] Zaptel HZ Warning   Steve Totaro
• [Asterisk-Users] SIPCALLID, but which callid?   Steve Totaro
• [Asterisk-Users] How to use a data T-1?   Steve Totaro
• [Asterisk-Users] finding mac addresses   Steve Totaro
• [Asterisk-Users] SIP Softphone on Thinclient?   Steve Totaro
• [Asterisk-Users] 1.2.9.1 crashed today   Steve Totaro
• [Asterisk-Users] 1.2.9.1 crashed today   Steve Totaro
• [Asterisk-Users] 1.2.9.1 crashed today   Steve Totaro
• [Asterisk-Users] Hitting * in a queue call hangs up?   Steve Totaro
• [Asterisk-Users] 1.2.9.1 crashed today   Steve Totaro
• [Asterisk-Users] Monitor a particular SIP call for training purposes   Steve Totaro
• [Asterisk-Users] 1.2.9.1 crashed today   Steve Totaro
• [Asterisk-Users] PRI Issue - Calls being rejected with unacceptable channel   Steve Totaro
• *** Spam *** [Asterisk-Users] Don't use CDRTool From AG-projescts   Steve Totaro
• [Asterisk-Users] Don't use CDRTool From AG-projescts   Steve Totaro
• [Asterisk-Users] 1.2.9.1 crashed today   Steve Totaro
• [Asterisk-Users] Don't use CDRTool From AG-projescts   Steve Totaro
• [Asterisk-Users] DUNDi Not Able to Handle ComplexFailoverSituations   Steve Totaro
• [Asterisk-Users] TE420P/TE415P?   Steve Totaro
• [Asterisk-Users] Playing sound before dialing   Steve Totaro
• [Asterisk-Users] FW: Asterisk Quintum A800 SIP Mode   Steve Totaro
• [Asterisk-Users] Zaptel answering the Line   Steve Totaro
• [Asterisk-Users] Zaptel answering the Line   Steve Totaro
• [Asterisk-Users] Zaptel answering the Line   Steve Totaro
• [Asterisk-Users] Set(CDR(userfield)) Trouble   Tristan
• [Asterisk-Users] Set(CDR(userfield)) Trouble   Tristan
• [Asterisk-Users] ISDN master clock issue ?   Tristan
• [Asterisk-Users] ISDN master clock issue ?   Tristan
• [Asterisk-Users] ISDN master clock issue ?   Tristan
• [Asterisk-Users] Updates/Inserts AND MYSQL(Clear ...)   Tristan
• [Asterisk-Users] Agents And Queues   Tristan
• [Asterisk-Users] Queues and hangup caller on Agent hangup   Tristan
• [Asterisk-Users] Queues and hangup caller on Agent hangup   Tristan
• [Asterisk-Users] Queues and hangup caller on Agent hangup   Tristan
• [Asterisk-Users] 1.2.9.1 crashed today   Tristan
• [Asterisk-Users] EuroISDN and Sangoma Card   Tristan
• [Asterisk-Users] Dial Macro timeout fails   Tristan
• [Asterisk-Users] Asterisk + PRI Card -> Nortel BCM   Wojciech Tryc
• [Asterisk-Users] Unable to connect to Asterisk? (simple[?] question)   Wojciech Tryc
• [Asterisk-Users] Anyone using VoIP WiFi phones?   Wojciech Tryc
• [Asterisk-Users] MeetMe Volume Issues   Justin Tunney
• [Asterisk-Users] additional calling party number   Alyed Tzompa
• [Asterisk-Users] Block access to number@domain.com   Pietro U
• [Asterisk-Users] Block access to number@domain.com   Pietro U
• [Asterisk-Users] hangup don't realease analog line   Pietro U
• [Asterisk-Users] long distance ask for pin   Pietro U
• [Asterisk-Users] long distance ask for pin   Pietro U
• [Asterisk-Users] Prices of g729 codec   Steve Underwood
• [Asterisk-Users] Prices of g729 codec   Steve Underwood
• [Asterisk-Users] Prices of g729 codec   Steve Underwood
• [Asterisk-Users] Quad T1 Card   Steve Underwood
• [Asterisk-Users] Quad T1 Card   Steve Underwood
• [Asterisk-Users] Quad T1 Card   Steve Underwood
• [Asterisk-Users] Fun with Echo   Steve Underwood
• [Asterisk-Users] Unicall acting really funny   Steve Underwood
• [Asterisk-Users] Fun with Echo   Steve Underwood
• [Asterisk-Users] grandstream GXV-3000   Steve Underwood
• [Asterisk-Users] Hard drive write cache   Steve Underwood
• [Asterisk-Users] GXP-2000 Audio Quality   Steve Underwood
• [Asterisk-Users] FAX + Digium + SpanDSP   Steve Underwood
• [Asterisk-Users] T1 Copper or T1 Fiber Line   Steve Underwood
• [Asterisk-Users] MOS Scores and LCR   Steve Underwood
• [Asterisk-Users] MOS Scores and LCR   Steve Underwood
• [Asterisk-Users] Where's the Fiber   Steve Underwood
• [Asterisk-Users] sangoma unicall m2rfc   Steve Underwood
• [Asterisk-Users] ECHO Tutorial   Steve Underwood
• [Asterisk-Users] Unicall acting really funny   Steve Underwood
• [Asterisk-Users] New Asteresk VOIP forum Buy Sell Discuss   VOICEIN at aol.com
• [Asterisk-Users] Small form factor system w/PCI slot   Jens Vagelpohl
• [Asterisk-Users] Small form factor system w/PCI slot   Jens Vagelpohl
• [Asterisk-Users] Don't use CDRTool From AG-projescts   Jens Vagelpohl
• [Asterisk-Users] Change g729 payload   Alejandro Vargas
• [Asterisk-Users] Change g729 payload   Alejandro Vargas
• [Asterisk-Users] Prices of g729 codec   Alejandro Vargas
• [Asterisk-Users] Prices of g729 codec   Alejandro Vargas
• [Asterisk-Users] SkypeOUT proxy   Oliver Vermeulen
• [Asterisk-Users] new DID's   Oliver Vermeulen
• [Asterisk-Users] JAMAICA DID'S - 1-876   Oliver Vermeulen
• [Asterisk-Users] JAMAICA DID'S - 1-876   Oliver Vermeulen
• [Asterisk-Users] asterisk nagios plugin   Ezio Vernacotola
• [Asterisk-Users] Anyone using VoIP WiFi phones?   Karl J. Vesterling
• [Asterisk-Users] Include Text file in Dial Plan   Maxim Vexler
• [Asterisk-Users] ASTCC: How to reset periodically all "card in use" flag back?   Maxim Vexler
• [Asterisk-Users] Changing standard Voicemail behavior   Maxim Vexler
• [Asterisk-Users] SIP voice recorder   Vic
• [Asterisk-Users] best hardphone for Asterisk?   Christian Victor
• [Asterisk-Users] best hardphone for Asterisk?   Christian Victor
• [Asterisk-Users] Openion on Sipura SPA-2100   Tom Vile
• [Asterisk-Users] DID in Houston 713?   Tom Vile
• [Asterisk-Users] DID from Latvia?   Tom Vile
• [Asterisk-Users] DID from Latvia?   Tom Vile
• [Asterisk-Users] DTMF feedthru again...   Tom Vile
• [Asterisk-Users] DTMF feedthru again...   Tom Vile
• [Asterisk-Users] DTMF feedthru again...   Tom Vile
• [Asterisk-Users] DTMF feedthru again...   Tom Vile
• [Asterisk-Users] DTMF feedthru again...   Tom Vile
• [Asterisk-Users] Re: GXP-2000 (steer clear)   Tom Vile
• [Asterisk-Users] Plainvoip problem.   Tom Vile
• [Asterisk-Users] FreePBX 2.1.0: Manually rewriting extensions_additional.conf   Tom Vile
• [Asterisk-Users] Phone recommendations?   Tom Vile
• [Asterisk-Users] No CID on ZAP   Tom Vile
• [Asterisk-Users] Asterisk --> BV: Incoming does not work....   Tom Vile
• [Asterisk-Users] Re: User Loses Ability to Make Outgoing Calls   Tom Vile
• [Asterisk-Users] Re: User Loses Ability to Make Outgoing Calls   Tom Vile
• [Asterisk-Users] Asterisk --> BV: Incoming does not work....   Tom Vile
• [Asterisk-Users] Re: User Loses Ability to Make Outgoing Calls   Tom Vile
• [Asterisk-Users] Can I get caller id passed to a phone connected to a Supura 2100?   Tom Vile
• [Asterisk-Users] Mail loop?   Tom Vile
• [Asterisk-Users] Remote employees using Polycom 501 lose ability to receive incoming calls after few minutes.   Tom Vile
• [Asterisk-Users] Remote employees using Polycom 501 lose ability to receive incoming calls after few minutes.   Tom Vile
• [Asterisk-Users] using kannel with asterisk   Tomislav Vojvodic
• [Asterisk-Users] using kannel with asterisk   Tomislav Vojvodic
• [Asterisk-Users] QuadBri card   Terry Wade
• [Asterisk-Users] getting agentID and DNID help   Terry Wade
• [Asterisk-Users] finding mac addresses   Michael Wallette
• [Asterisk-Users] Re: Asterisk-Users Digest, Vol 23, Issue 135   Michael Wallette
• [Asterisk-Users] Re: Quality monitoring   Michael Wallette
• [Asterisk-Users] registering a Motorola vt1005   Brandon Warner
• [Asterisk-Users] registering a Motorola VT1005   Brandon Warner
• [Asterisk-Users] How to register a Motorola VT1005   Brandon Warner
• [Asterisk-Users] Single T1 card with Echo Cancellation to work with Dell?   Warren
• [Asterisk-Users] Best $300 VoIP phone for asterisk?   Warren
• [Asterisk-Users] EC needed in all-digital situation?   Warren
• [Asterisk-Users] Best $300 VoIP phone for asterisk?   Warren
• [Asterisk-Users] EC needed in all-digital situation?   Warren
• [Asterisk-Users] Best $300 VoIP phone for asterisk?   Warren
• [Asterisk-Users] Best PRI provider, Long Island, NY   Warren
• [Asterisk-Users] A few questions on a conversion to *   Warren
• [Asterisk-Users] How to use a data T-1?   Warren
• [Asterisk-Users] How to use a data T-1?   Warren
• [Asterisk-Users] How to use a data T-1?   Warren
• [Asterisk-Users] How to use a data T-1?   Warren
• [Asterisk-Users] How to use a data T-1?   Warren
• [Asterisk-Users] How to use a data T-1?   Warren
• [Asterisk-Users] Anyone using VoIP WiFi phones?   Warren
• [Asterisk-Users] Anyone using VoIP WiFi phones?   Warren
• [Asterisk-Users] USB handset options for softphones   Warren
• [Asterisk-Users] freepbx centos 4 install script?   Warren
• [Asterisk-Users] freepbx centos 4 install script?   Warren
• [Asterisk-Users] Dell PowerEdge 1650   Warren
• [Asterisk-Users] This is getting really annoying - re: POSTFIX   Warren
• [Asterisk-Users] JAMAICA DID'S - 1-876   Warren
• [Asterisk-Users] email a message   Matthew Warren
• [Asterisk-Users] PHP-AGI help   Matthew Warren
• [Asterisk-Users] syslog server   Matthew Warren
• [Asterisk-Users] Re: syslog server   Matthew Warren
• [Asterisk-Users] Sip stuck   Matthew Warren
• [Asterisk-Users] php-snmp   Matthew Warren
• [Asterisk-Users] PHP UnixODBC MS SQl 2000   Wasif
• [Asterisk-Users] Asterisk as Wholesale   Wasif
• [Asterisk-Users] Asterisk & Cisco 3800   Wasif
• [Asterisk-Users] Mysql Trixbox   Wasif
• [Asterisk-Users] G729 Code   Wasif
• [Asterisk-Users] astdb entry in sip.conf   Watkins, Bradley
• [Asterisk-Users] astdb entry in sip.conf   Watkins, Bradley
• [Asterisk-Users] Config Revision Control   Watkins, Bradley
• [Asterisk-Users] DUNDi Docs   Watkins, Bradley
• [Asterisk-Users] DUNDi Not Able to Handle Complex FailoverSituations   Watkins, Bradley
• [Asterisk-Users] Best $300 VoIP phone for asterisk?   Watkins, Bradley
• [Asterisk-Users] Best $300 VoIP phone for asterisk?   Watkins, Bradley
• [Asterisk-Users] DUNDi Not Able to Handle ComplexFailoverSituations   Watkins, Bradley
• [Asterisk-Users] DUNDi Not Able to HandleComplexFailoverSituations   Watkins, Bradley
• [Asterisk-Users] SE Michigan asterisk users group   Watkins, Bradley
• [Asterisk-Users] TE405P Dropping Calls. !! Got I-frame while linkstate 0   Watkins, Bradley
• [Asterisk-Users] Single T1 card with Echo CancellationtoworkwithDell?   David Waugh
• [Asterisk-Users] Queue errors when phones are down, and possible solution   Franklin Webb
• [Asterisk-Users] ISDN BRI NetJet   Hermann Wecke
• [Asterisk-Users] Packet8 and Asterisk, do they play nice?   Hermann Wecke
• [Asterisk-Users] ISDN   Hermann Wecke
• [Asterisk-Users] set file path   Jack Wei
• [Asterisk-Users] Re: [asterisk-biz] India Routes   Jon Weisman
• [Asterisk-Users] SIP Delayed Answer   Michael Welter
• [Asterisk-Users] Re: GXP-2000 (steer clear)   Michael Welter
• [Asterisk-Users] MOH too loud   Michael Welter
• [Asterisk-Users] How to use a data T-1?   Michael Welter
• [Asterisk-Users] Re: User Loses Ability to Make Outgoing Calls   Michael Welter
• [Asterisk-Users] Polycom 601 question   Michael Welter
• [Asterisk-Users] Converting Voicemail wav to mp3   BJ Weschke
• [Asterisk-Users] Large Asterisk System   BJ Weschke
• [Asterisk-Users] app_queue and Real roundrobin   BJ Weschke
• [Asterisk-Users] Polycom-Asterisk hints/presence   BJ Weschke
• [Asterisk-Users] Polycom-Asterisk hints/presence   BJ Weschke
• [Asterisk-Users] Polycom-Asterisk hints/presence   BJ Weschke
• [Asterisk-Users] Polycom-Asterisk hints/presence   BJ Weschke
• [Asterisk-Users] Polycom-Asterisk hints/presence   BJ Weschke
• [Asterisk-Users] Outgoing call bridging   BJ Weschke
• [Asterisk-Users] Disabling debug output   BJ Weschke
• [Asterisk-Users] Disabling debug output   BJ Weschke
• [Asterisk-Users] increase the volume ?   BJ Weschke
• [Asterisk-Users] Attended transfer and queue   BJ Weschke
• [Asterisk-Users] Single agent multiple queues....   BJ Weschke
• [Asterisk-Users] Attended transfer and queue   BJ Weschke
• [Asterisk-Users] Hitting * in a queue call hangs up?   BJ Weschke
• [Asterisk-Users] Attended transfer and queue   BJ Weschke
• [Asterisk-Users] Can this config sustain 30 users?   BJ Weschke
• [Asterisk-Users] No reinvite - reason?   BJ Weschke
• [Asterisk-Users] Queues and macros and agents   BJ Weschke
• [Asterisk-Users] Asterisk Follow Me   BJ Weschke
• [Asterisk-Users] Polycom Queues   BJ Weschke
• [Asterisk-Users] 302 Redirecting support   BJ Weschke
• [Asterisk-Users] multiple port   BJ Weschke
• [Asterisk-Users] Echo Problem with T411P   BJ Weschke
• [Asterisk-Users] Comedian Mail not deleting .txt file   BJ Weschke
• [Asterisk-Users] 1.2.9.1 crashed today   BJ Weschke
• [Asterisk-Users] 1.2.9.1 crashed today   BJ Weschke
• [Asterisk-Users] 1.2.9.1 crashed today   BJ Weschke
• SV: [Asterisk-Users] periodic-announce not working   BJ Weschke
• SV: [Asterisk-Users] periodic-announce not working   BJ Weschke
• SV: [Asterisk-Users] periodic-announce not working   BJ Weschke
• [Asterisk-Users] Asterisk ACD with Polycom IP501   BJ Weschke
• [Asterisk-Users] Anyway to set maximum wait time when there's only 1 user in Meetme?   Pimjai Wesnarat
• [Asterisk-Users] MeetMe - Annouce user join/leave without recording the name   Pimjai Wesnarat
• [Asterisk-Users] caller id   Joshua West
• [Asterisk-Users] Echo and crackle   Joshua West
• [Asterisk-Users] Motherboard Selection For TE110P & TDM400P   Joshua West
• [Asterisk-Users] best hardphone for Asterisk?   Joshua West
• [Asterisk-Users] open source sip softphone (Window OS version )   Derek Whitten
• [Asterisk-Users] Looking for SIP provider with minimal call setuptime   Cullin J. Wible
• [Asterisk-Users] best hardphone for Asterisk?   Cullin J. Wible
• [Asterisk-Users] Question about ring groups and ext. busy in call   Cullin J. Wible
• [Asterisk-Users] Modifying Voicemail menus?   Cullin J. Wible
• [Asterisk-Users] Voicemail volume adjustment   Cullin J. Wible
• [Asterisk-Users] Voicemail volume adjustment   Cullin J. Wible
• [Asterisk-Users] Asterisk 1.2.7.1 with Polycom 501 on SIP -> ConfCalling   Cullin J. Wible
• [Asterisk-Users] Remote employees using Polycom 501 lose abilityto receive incoming calls after few minutes.   Cullin J. Wible
• [Asterisk-Users] Voicemail volume adjustment   Cullin J. Wible
• [Asterisk-Users] Configuring behaviour of flash hook   Eric "ManxPower" Wieling
• [Asterisk-Users] TDM-400 doesn't detect far-end hangup   Eric "ManxPower" Wieling
• [Asterisk-Users] DTMF and DISA   Eric "ManxPower" Wieling
• [Asterisk-Users] This should be easy: What happens when the Calling Party hangs up   Eric "ManxPower" Wieling
• [Asterisk-Users] DTMF feedthru again...   Eric "ManxPower" Wieling
• [Asterisk-Users] GXP-2000   Eric "ManxPower" Wieling
• [Asterisk-Users] Block access to number@domain.com   Eric "ManxPower" Wieling
• [Asterisk-Users] Busy/on the phone detection with Cisco 7960   Eric "ManxPower" Wieling
• [Asterisk-Users] Native Music On Hold Volume LOUD! How to adjust?   Eric "ManxPower" Wieling
• [Asterisk-Users] Fun with Echo   Eric "ManxPower" Wieling
• [Asterisk-Users] SBC/ATT Supertrunk configuration   Eric "ManxPower" Wieling
• [Asterisk-Users] Question setting up a "bat phone" extension.   Eric "ManxPower" Wieling
• [Asterisk-Users] Callback Application: Suggestions Please.   Eric "ManxPower" Wieling
• [Asterisk-Users] TTS engine query   Eric "ManxPower" Wieling
• [Asterisk-Users] delay in MeetMe   Eric "ManxPower" Wieling
• [Asterisk-Users] delay in MeetMe   Eric "ManxPower" Wieling
• [Asterisk-Users] No ring tone on outgoing calls   Eric "ManxPower" Wieling
• [Asterisk-Users] Which application to open Zap channel?   Eric "ManxPower" Wieling
• [Asterisk-Users] transcoding problem   Eric "ManxPower" Wieling
• [Asterisk-Users] Directory - First Name/Last Name - How to use both? a@h?   Eric "ManxPower" Wieling
• [Asterisk-Users] No ring tone on outgoing calls   Eric "ManxPower" Wieling
• [Asterisk-Users] EC needed in all-digital situation?   Eric "ManxPower" Wieling
• [Asterisk-Users] No "ringing" being played to remote caller?   Eric "ManxPower" Wieling
• [Asterisk-Users] SIP codec preference order ineffective   Eric "ManxPower" Wieling
• [Asterisk-Users] DTMF in the middle of a call   Eric "ManxPower" Wieling
• [Asterisk-Users] Echo Cancelling VoIP traffic   Eric "ManxPower" Wieling
• [Asterisk-Users] Which application to open Zap channel?   Eric "ManxPower" Wieling
• [Asterisk-Users] finding mac addresses   Eric "ManxPower" Wieling
• [Asterisk-Users] Re: Two FXO: How to dial a number when a RING comes in?   Eric "ManxPower" Wieling
• [Asterisk-Users] when I press "transfer" -> blind -> 700 . The user is not able to hear what extension the call was parked on   Eric "ManxPower" Wieling
• [Asterisk-Users] when I press "transfer" -> blind -> 700 . The user is not able to hear what extension the call was parked on   Eric "ManxPower" Wieling
• [Asterisk-Users] Soekris net4801 and IAXy dhcp issue   Eric "ManxPower" Wieling
• [Asterisk-Users] Echo and crackle   Eric "ManxPower" Wieling
• [Asterisk-Users] '500 Internal Server' Error on SIP NOTIFY   Eric "ManxPower" Wieling
• [Asterisk-Users] M() option to Dial   Eric "ManxPower" Wieling
• [Asterisk-Users] Need to Hire: PHP Programmer for PhoneCALL   Dustin Wildes
• [Asterisk-Users] Asterisk queue log solution?   Dustin Wildes
• [Asterisk-Users] GXP-2000 and Shared Line Appearances   Dustin Wildes
• [Asterisk-Users] GXP-2000 and Shared Line Appearances   Dustin Wildes
• [Asterisk-Users] Voicemail volume adjustment   Dustin Wildes
• [Asterisk-Users] Voicemail volume adjustment   Dustin Wildes
• [Asterisk-Users] Voicemail volume adjustment   Dustin Wildes
• [Asterisk-Users] Surge Protector for T1/PRI ?   Dustin Wildes
• [Asterisk-Users] Conferencing with multiple servers   Wildheart
• [Asterisk-Users] manager DBDel action   Wildheart
• [Asterisk-Users] Echocancelwhenbridged   Wildheart
• [Asterisk-Users] Caller ID Matching in extensions.conf   Wildheart
• [Asterisk-Users] Caller ID Matching in extensions.conf   Wildheart
• [Asterisk-Users] MixMonitor Problems   Wildheart
• [Asterisk-Users] asterisk behind cisco pix 506   Andreas Wilkens
• [Asterisk-Users] asterisk behind cisco pix 506   Andreas Wilkens
• [Asterisk-Users] asterisk behind cisco pix 506   Andreas Wilkens
• [Asterisk-Users] asterisk behind cisco pix 506   Andreas Wilkens
• [Asterisk-Users] asterisk behind cisco pix 506   Andreas Wilkens
• [Asterisk-Users] FW: Asterisk Quintum A800 SIP Mode   Neill Wilkinson
• [Asterisk-Users] Comedian Mail not deleting .txt file   Andrew Willerding
• [Asterisk-Users] OPENSER / SER and Asterisk   Kelvin Williams
• [Asterisk-Users] Problem with simple incoming calls   Darin Willits
• [Asterisk-Users] DUNDi Not Able to Handle Complex Failover Situations   Terry Wilson
• [Asterisk-Users] SpeedTouch 780WL   WipeOut
• [Asterisk-Users] transfer & other features   Ronald Wiplinger
• [Asterisk-Users] How to make this into a Macro?   Ronald Wiplinger
• [Asterisk-Users] Xlite and # code after call is connected   Ronald Wiplinger
• [Asterisk-Users] Xlite and # code after call is connected   Ronald Wiplinger
• [Asterisk-Users] voipbuster & dtmf tones?   Ronald Wiplinger
• [Asterisk-Users] Voip* 300 minutes limit, credit expires   Ronald Wiplinger
• [Asterisk-Users] ASTCC: How to reset periodically all "card in use" flag back?   Ronald Wiplinger
• [Asterisk-Users] Is anybody using XEN in conjunction with Asterisk and/or Openser?   Ronald Wiplinger
• [Asterisk-Users] ASTCC: How to reset periodically all "card in use" flag back?   Ronald Wiplinger
• [Asterisk-Users] ASTCC: How to reset periodically all "card in use" flag back?   Ronald Wiplinger
• [Asterisk-Users] ASTCC: customer wants 100 accounts   Ronald Wiplinger
• [Asterisk-Users] Realtime: how to use column setvar?   Ronald Wiplinger
• [Asterisk-Users] ASTCC: customer wants 100 accounts   Ronald Wiplinger
• [Asterisk-Users] username in Real-time changes all the time   Ronald Wiplinger
• [Asterisk-Users] multiple includes   Ronald Wiplinger
• [Asterisk-Users] MeetMe with recording - bitrate too low   Kevin Withnall
• [Asterisk-Users] capi drivers for suse-10.1   Hans Witvliet
• [Asterisk-Users] asterisk load balance   Simon Woodhead
• [Asterisk-Users] Voicemail with NFS   Simon Woodhead
• [Asterisk-Users] Add Country to CDR's   Simon Woodhead
• [Asterisk-Users] Email notification   Roger Workman
• [Asterisk-Users] asterisk to mobile phone   Roger Workman
• [Asterisk-Users] Multiple sip proxy per * server.   Wai Wu
• [Asterisk-Users] How many TE405 ...   Wai Wu
• [Asterisk-Users] AGI script can not print out error message to console   Zichao Wu
• [Asterisk-Users] queue always hangs up/skip the next agent after ringing a agent -- help!!!   Isaac Xiao
• [Asterisk-Users] Please Help - Polycom IP 601 Buddy Watch problems   Isaac Xiao
• [Asterisk-Users] 1.2.9.1 SIP/Local/Queue behaviours weird   Isaac Xiao
• [Asterisk-Users] 1.2.9.1 SIP/Local/Queue behaviours weird   Isaac Xiao
• [Asterisk-Users] How many TE405 ...   Hao Xu
• [Asterisk-Users] ISDN BRI (I.430) over ethernet   Hao Xu
• [Asterisk-Users] A dual Asterisk server question   Tielin Xu
• [Asterisk-Users] Question about context from-internal   Tielin Xu
• [Asterisk-Users] Addon-ooh323 install problem   Tetsuya Yamamoto
• [Asterisk-Users] Addon-ooh323 install problem   Tetsuya Yamamoto
• [Asterisk-Users] Addon-ooh323 install problem   Tetsuya Yamamoto
• [Asterisk-Users] PSTN outgoing DTMF vs. transfer Problem   Vahan Yerkanian
• [Asterisk-Users] DTMF feedthru again...   Vahan Yerkanian
• [Asterisk-Users] Soundwin S2400 standalone 24FXS/FXO SIP gateways   Vahan Yerkanian
• [Asterisk-Users] spandsp with t.38   Roman Yeryomin
• [Asterisk-Users] Are zttest results relevant on a system with no telephony hardware?   Carl Youngblood
• [Asterisk-Users] Are zttest results relevant on a system with no telephony hardware?   Carl Youngblood
• [Asterisk-Users] Need to track dropped calls   Carl Youngblood
• [Asterisk-Users] Need to track dropped calls   Carl Youngblood
• [Asterisk-Users] Problem trying to SayDigits when an invalid extension is dialed   Carl Youngblood
• [Asterisk-Users] Problem trying to SayDigits when an invalid extension is dialed   Carl Youngblood
• [Asterisk-Users] Need to track dropped calls   Carl Youngblood
• [Asterisk-Users] Problem trying to SayDigits when an invalid extension is dialed   Carl Youngblood
• [Asterisk-Users] Problem trying to SayDigits when an invalid   Carl Youngblood
• [Asterisk-Users] h323 with asterisk problem   Yusuf
• [Asterisk-Users] PCI or MiniPCI Hardware DSP for G.729, G.723.1 and/or GSM   Roland Zagler
• [Asterisk-Users] FW: Asterisk Quintum A800 SIP Mode   Roland Zagler
• [Asterisk-Users] frame.c:128 ast_smoother_feed   Ever Zalazar
• [Asterisk-Users] frame.c:128 ast_smoother_feed   Ever Zalazar
• [Asterisk-Users] frame.c:128 ast_smoother_feed   Ever Zalazar
• [Asterisk-Users] Comedian Mail not deleting .txt file   Edward de Zeeuw
• [Asterisk-Users] Comedian Mail not deleting .txt file   Edward de Zeeuw
• [Asterisk-Users] Snom 360 Passsword Issue   Edward de Zeeuw
• [Asterisk-Users] Snom 360 Passsword Issue   Edward de Zeeuw
• [Asterisk-Users] Prices of g729 codec   Zoa
• Idefisk security fix - was [Asterisk-Users] Asterisk 1.2.9 and 1.0.11 Released -- Security Fix   Zoa
• [Asterisk-Users] IAX2 Vs SIP cpu load   Zoa
• [Asterisk-Users] IAX2 Vs SIP cpu load   Zoa
• [Asterisk-Users] What is Echo?   Zoa
• [Asterisk-Users] How much bandwidth needed?   Zoa
• [Asterisk-Users] Asterisk server   Zoa
• [Asterisk-Users] How much bandwidth needed?   Zoa
• [Asterisk-Users] Re: Bristuffed Asterisk: Hangup problems   Jeroen Zwarts
• [Asterisk-Users] bristuff hangup issue   Jeroen Zwarts
• [Asterisk-Users] zapata.conf: recent changes?   Wolfgang Zweimueller
• [Asterisk-Users] extensions.conf   andrutto
• [Asterisk-Users] RE: extensions.conf   andrutto
• [Asterisk-Users] problem with Asterisk DTMF   armx at inf.its-sby.edu
• [Asterisk-Users] Re: GXP-2000 (steer clear)   asterisk
• [Asterisk-Users] Linksys SRW224P POE Switch   asterisk
• [Asterisk-Users] Linksys SRW224P POE Switch   asterisk
• [Asterisk-Users] Linksys SRW224P POE Switch   asterisk
• [Asterisk-Users] Linksys SRW224P POE Switch   asterisk
• [Asterisk-Users] Linksys SRW224P POE Switch   asterisk
• [Asterisk-Users] Sangoma driver update?   asterisk
• [Asterisk-Users] how to hang the zap channel   Bartosz Wegrzyn - asterisk
• [Asterisk-Users] how to hang the zap channel   Bartosz Wegrzyn - asterisk
• [Asterisk-Users] how to hang the zap channel   Bartosz Wegrzyn - asterisk
• [Asterisk-Users] how to hang the zap channel   Bartosz Wegrzyn - asterisk
• [Asterisk-Users] iax2 registration problems   Bartosz Wegrzyn - asterisk
• [Asterisk-Users] iax2 registration problems   Bartosz Wegrzyn - asterisk
• [Asterisk-Users] Meetme max users   Bartosz Wegrzyn - asterisk
• [Asterisk-Users] Meetme max users   Bartosz Wegrzyn - asterisk
• [Asterisk-Users] show channel issue with 1.2.9   asterisk-user at dualcall.com
• [Asterisk-Users] Can I use an onboard modem?   asterisk-user at dualcall.com
• [Asterisk-Users] who is the mantainer ....   asterisk at frameweb.it
• [Asterisk-Users] who is the mantainer ....   asterisk at frameweb.it
• [Asterisk-Users] h323 phone   asterisk at frameweb.it
• [Asterisk-Users] OH323 issue on AT320 Phones   asterisk at frameweb.it
• [Asterisk-Users] OH323 issue on AT320 Phones   asterisk at frameweb.it
• [Asterisk-Users] outgoing calls   asterisk at karthaus.nl
• [Asterisk-Users] SIP or IAX client written in C   asterisk at yeahmon.net
• [Asterisk-Users] MINNESOTA: TwinCities Asterisk Users Group - Saturday June 10th - 2006   asterisk_help at iwishi.nu
• [Asterisk-Users] SIP Softphone on Thinclient?   bails
• [Asterisk-Users] skype out   simon elliston ball
• [Asterisk-Users] Dialstatus   bob at semanticedge.de
• [Asterisk-Users] use AT320 international call   brett at websmyths.com
• [Asterisk-Users] chat with asterisk   brett at websmyths.com
• [Asterisk-Users] SE Michigan asterisk users group   btb
• [Asterisk-Users] Microsoft CRM & Asterisk   calvis
• [Asterisk-Users] One way Audio   leonimar cape
• [Asterisk-Users] Problems with ZAP dial timeout   casasterisk at valnet.com
• [Asterisk-Users] test   charles
• [Asterisk-Users] hangup lag causing the answering of already answered calls   mrlord chewie
• RES: [Asterisk-Users] Meetme max users   cleviton.araujo at caixa.gov.br
• [Asterisk-Users] Re: XML to monitor queues on Cisco display ?   Jean-Louis curty
• [Asterisk-Users] Re: XML to monitor queues on Cisco display ?   Jean-Louis curty
• [Asterisk-Users] Testing a FastAGI script   didier
• [Asterisk-Users] ADSL modem, TDM400P, zaptel and not hanging up   digium at sanguinarius.co.uk
• [Asterisk-Users] ADSL modem, TDM400P, zaptel and not hanging up   digium at sanguinarius.co.uk
• [Asterisk-Users] GXP-2000 1.1.0.13 Issues   drew-asterisk-users at elysium.ltd.uk
• [Asterisk-Users] GXP-2000 1.1.0.13 Issues   drew-asterisk-users at elysium.ltd.uk
• [Asterisk-Users] GXP-2000 1.1.0.13 Issues   drew-asterisk-users at elysium.ltd.uk
• [Asterisk-Users] GXP-2000 1.1.0.13 Issues   drew-asterisk-users at elysium.ltd.uk
• [Asterisk-Users] T1 Copper or T1 Fiber Line   dthurn at tstt.net.tt
• [Asterisk-Users] T1 Copper or T1 Fiber Line   dthurn at tstt.net.tt
• [Asterisk-Users] Where's the Fiber   dthurn at tstt.net.tt
• [Asterisk-Users] T1 delivered via Copper   dthurn at tstt.net.tt
• [Asterisk-Users] What type of T1 Cards to use for my Asterisk PBX   dthurn at tstt.net.tt
• [Asterisk-Users] Unlock / install of Cisco 7940 IP Phone ?   end1r at comcast.net
• [Asterisk-Users] quad t1 / 1U rack server combos   f6hqz-m at hamwlan.net
• [Asterisk-Users] x100p buying advice   f6hqz-m at hamwlan.net
• *** Spam *** [Asterisk-Users] Example config files for Snom mass updating?   fischer at snom.de
• [Asterisk-Users] Random Zap Channel Drops to SIP   asterisk fpf
• [Asterisk-Users] a command to dump all callers in queues preferably from asterisk console   fwebb at imminc.com
• [Asterisk-Users] when I press "transfer" -> blind -> 700 . The user is not able to hear what extension the call was parked on   sdgesa gaeharth
• [Asterisk-Users] when I press "transfer" -> blind -> 700 . The user is not able to hear what extension the call was parked on   sdgesa gaeharth
• [Asterisk-Users] caller id   sdgesa gaeharth
• [Asterisk-Users] caller id   sdgesa gaeharth
• [Asterisk-Users] no dialtone on channel banks   gene
• [Asterisk-Users] no dialtone on channel banks   gene
• [Asterisk-Users] ISDN: 3° incoming call   francesco giuliani
• [Asterisk-Users] ISDN: 3° incoming call   francesco giuliani
• [Asterisk-Users] ISDN: 3° incoming call   francesco giuliani
• [Asterisk-Users] ISDN: 3° incoming call   francesco giuliani
• [Asterisk-Users] open source sip softphone (Window OS version )   Asterisk guy
• [Asterisk-Users] open source sip softphone (Window OS version )   Asterisk guy
• [Asterisk-Users] Polycom-Asterisk hints/presence   hgaillac-sip at yahoo.fr
• [Asterisk-Users] Tr: RE : Openser+Asterisk+voice mail   hgaillac-sip at yahoo.fr
• [Asterisk-Users] asterisk-1.2.9 is not stable   hgaillac-sip at yahoo.fr
• [Asterisk-Users] asterisk-1.2.9 / res_snmp.so   hgaillac-sip at yahoo.fr
• [Asterisk-Users] polycom ftp   hgaillac-sip at yahoo.fr
• [Asterisk-Users] Polycom Files   hgaillac-sip at yahoo.fr
• [Asterisk-Users] dial pattern   hgaillac-sip at yahoo.fr
• [Asterisk-Users] SRTP/SIPS   hgaillac-sip at yahoo.fr
• [Asterisk-Users] IP/SS7 gateway on Sun Ultra 20 amd64   hgaillac-sip at yahoo.fr
• [Asterisk-Users] asterisk on AMD 64 BIT   hgaillac-sip at yahoo.fr
• [Asterisk-Users] Which simple billing application   hgaillac-sip at yahoo.fr
• [Asterisk-Users] asterisk+cdrtool   hgaillac-sip at yahoo.fr
• [Asterisk-Users] CDRTool   hgaillac-sip at yahoo.fr
• [Asterisk-Users] CDRTool / asterisk billing based on realtime   hgaillac-sip at yahoo.fr
• [Asterisk-Users] Don't use CDRTool From AG-projescts   hgaillac-sip at yahoo.fr
• [Asterisk-Users] Don't use CDRTool From AG-projescts   hgaillac-sip at yahoo.fr
• [Asterisk-Users] Don't use CDRTool From AG-projescts   hgaillac-sip at yahoo.fr
• [Asterisk-Users] CDRTool +Asterisk + Ser   hgaillac-sip at yahoo.fr
• [Asterisk-Users] RE : Re: [Serusers] CDRTool +Asterisk + Ser   hgaillac-sip at yahoo.fr
• [Asterisk-Users] voipcheap.com - miracle free land line calls   listas iPfone
• [Asterisk-Users] configuration   issam
• [Asterisk-Users] configuration   issam
• [Asterisk-Users] statistics   issam
• [Asterisk-Users] bewan phonebox   issam
• [Asterisk-Users] sip   issam
• [Asterisk-Users] tdm400   issam
• [Asterisk-Users] error with tdm11b   issam
• [Asterisk-Users] chat with asterisk   issam
• [Asterisk-Users] using kannel with asterisk   issam
• [Asterisk-Users] using kannel with asterisk   issam
• [Asterisk-Users] Asterisk server   jacobso1
• [Asterisk-Users] best hardphone for Asterisk?   jacobso1
• [Asterisk-Users] TE420P/TE415P?   jan.sarin at securia.se
• [Asterisk-Users] Hardware to connect analog and ISDN fax devices   jbauer at teec.com
• [Asterisk-Users] SE Michigan asterisk users group   jglucky at blueware.net
• [Asterisk-Users] WRTG54GS Capacity   john at cusick.ws
• [Asterisk-Users] Music On Hold troubleshooting   kharris
• [Asterisk-Users] Audio problems on Zap & SIP, local network, not IRQ related?   kjcsb
• [Asterisk-Users] Re: Audio problems on Zap & SIP, local network, not IRQ related?   kjcsb
• [Asterisk-Users] Re: Audio problems on Zap & SIP, local network, not IRQ related?   kjcsb
• [Asterisk-Users] USB handset options for softphones   klubarpop
• [Asterisk-Users] Call limit function on sip channel to external pop   bram kortleven
• [Asterisk-Users] AstriCon Europe - Only 1 Week Away   law
• [Asterisk-Users] User Loses Ability to Make Outgoing Calls   lenz
• [Asterisk-Users] Asterisk queue log solution?   lenz
• [Asterisk-Users] QueueMetrics 1.2 released today   lenz
• [Asterisk-Users] Re: Asterisk-Users Digest, Vol 23, Issue 8   levi at citiprice.com
• [Asterisk-Users] Re: Asterisk-Users Digest, Vol 23, Issue 9   levi at citiprice.com
• [Asterisk-Users] Re: Asterisk-Users Digest, Vol 23, Issue 10   levi at citiprice.com
• [Asterisk-Users] Re: Asterisk-Users Digest, Vol 23, Issue 11   levi at citiprice.com
• [Asterisk-Users] Action: Originate PROBLEM   lokotes
• SV: [Asterisk-Users] I can hear only one way when I use nokia e-60withX-lite   list mail
• [Asterisk-Users] Re: GXP-2000 (steer clear)   list mail
• [Asterisk-Users] GXP-2000   list mail
• [Asterisk-Users] Voicemail to Email on Blackberry   list mail
• [Asterisk-Users] GXP-2000   list mail
• [Asterisk-Users] Voicemail to Email on Blackberry   list mail
• [Asterisk-Users] Attended transfer and queue   aston martin
• [Asterisk-Users] Attended transfer and queue   aston martin
• [Asterisk-Users] Attended transfer and queue   aston martin
• [Asterisk-Users] Attended transfer and queue   aston martin
• [Asterisk-Users] Attended transfer and queue   aston martin
• [Asterisk-Users] Turning off a temporary message in voicemail   mavince at optonline.net
• [Asterisk-Users] Turning off a temporary message in voicemail   mavince at optonline.net
• [Asterisk-Users] revisit to legacy PBX and CID over PRI   mavince at optonline.net
• [Asterisk-Users] Outgoing call bridging   mawali at news.icns.com
• [Asterisk-Users] Outgoing call bridging   mawali at news.icns.com
• [Asterisk-Users] Prices of g729 codec   mgraves at mstvp.com
• [Asterisk-Users] Sangoma A101 configuration   mhiguera
• [Asterisk-Users] Fun with Echo -- Follow up   mike
• [Asterisk-Users] Fun with Echo -- Follow up   mike
• [Asterisk-Users] Fun with Echo -- Follow up   mike
• [Asterisk-Users] Integrating Asterisk   mitcheloc at gmail.com
• [Asterisk-Users] RE: Digium pound key software appliance opinions   mitcheloc at gmail.com
• [Asterisk-Users] click to call features on asterisk   mitcheloc at gmail.com
• [Asterisk-Users] Easiest (best?) linux distribution for dedicatedAsterisk box?   mitcheloc at gmail.com
• [Asterisk-Users] 100 lines PBX + system config - repost   mitcheloc at gmail.com
• [Asterisk-Users] Transfer call via AMI or dialplan   mitcheloc at gmail.com
• [Asterisk-Users] Which is the best user GUI ?   mitcheloc at gmail.com
• [Asterisk-Users] voiceone?   mitcheloc at gmail.com
• [Asterisk-Users] SIP Softphone on Thinclient?   mitcheloc at gmail.com
• [Asterisk-Users] SIP Softphone on Thinclient?   mitcheloc at gmail.com
• [Asterisk-Users] finding mac addresses   mojowrkn
• [Asterisk-Users] Asterisk and SATA Raid 1   mustardman29
• [Asterisk-Users] Asterisk and SATA Raid 1   mustardman29
• [Asterisk-Users] Vonage and FXO   mustardman29
• [Asterisk-Users] Re: GXP-2000 (steer clear)   mustardman29
• [Asterisk-Users] GXP-2000   mustardman29
• [Asterisk-Users] Bullet-proof FXO?   mustardman29
• [Asterisk-Users] Low volume/ audio problems on TDM400 card   news.asterisk.users
• [Asterisk-Users] TDM400P static on call   news.asterisk.users
• [Asterisk-Users] Low volume/ audio problems on TDM400 card   news.asterisk.users
• [Asterisk-Users] Low volume/ audio problems on TDM400 card   news.asterisk.users
• [Asterisk-Users] TDM400P bad echo problem, tried lots of things   news.asterisk.users
• [Asterisk-Users] Problems with misdn and BN8S0   nik600
• [Asterisk-Users] BN8S0 Installation problem - 0 devices registrered   nik600
• [Asterisk-Users] BN8S0 Installation problem - 0 devices registrered   nik600
• [Asterisk-Users] BN8S0 Installation problem - 0 devices registrered   nik600
• [Asterisk-Users] CISCO POE   nik600
• [Asterisk-Users] CISCO POE   nik600
• [Asterisk-Users] BN8S0 problem - Extension can never match, so disconnecting   nik600
• [Asterisk-Users] BN8S0 problem - Extension can never match, so disconnecting   nik600
• [Asterisk-Users] pickup a call from a group   nik600
• [Asterisk-Users] fixed ring strategy   nik600
• [Asterisk-Users] forward a call to a SIP account on a remote server   nik600
• [Asterisk-Users] Cisco 7936 Conference Phone - SIP or SCCP?   nrbwpi at gmail.com
• [Asterisk-Users] SIP codec preference order ineffective   olivier.taylor
• [Asterisk-Users] Need to Hire: PHP Programmer for PhoneCALL   olivier.taylor
• [Asterisk-Users] SIP Multi Call Generation   olivier.taylor
• [Asterisk-Users] Most stable Asterisk version   olivier.taylor
• [Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!   olivier.taylor
• [Asterisk-Users] Monitor a particular SIP call for training purposes   phil.dawson at marnock.com
• [Asterisk-Users] Polycom-Asterisk hints/presence   picciuX
• [Asterisk-Users] FOSS, Science, and Public activism   proclus at gnu-darwin.org
• [Asterisk-Users] asterisk+voicemail+openser   ram
• [Asterisk-Users] SIP/2.0 484 Address Incomplete   ram
• [Asterisk-Users] asterisk 1.2.8 compilation problem   ram
• [Asterisk-Users] software to do sip stress tests   rcarvalho at iric.up.pt
• [Asterisk-Users] "I can hear them but they can't hear me" with VoipBuster   rdquiterio.si at cenfim.pt
• [Asterisk-Users] connecting asterisk to pstn help   ravi reddy
• [Asterisk-Users] Re: Asterisk-Users Digest, Vol 23, Issue 2   ravi reddy
• [Asterisk-Users] using mediaproxy for both ASTERISK and SER   ravi reddy
• [Asterisk-Users] (no subject)   Vincent renaville
• [Asterisk-Users] Can´t send emails   yrving rivas
• Re: [Asterisk-Users] Can´t send emails   yrving rivas
• Re: [Asterisk-Users] Can´t send emails   yrving rivas
• Re: [Asterisk-Users] Can´t send emails   yrving rivas
• Re: [Asterisk-Users] Can´t send emails   yrving rivas
• Re: [Asterisk-Users] Can´t send emails   yrving rivas
• Re: [Asterisk-Users] Can´t send emails   yrving rivas
• [Asterisk-Users] stuck call on asterisk   rristroph at airlink.net
• [Asterisk-Users] a new asterisk version   amna saleem
• [Asterisk-Users] asterisk-1.2.9.1   amna saleem
• [Asterisk-Users] asterisk-1.2.9.1   amna saleem
• [Asterisk-Users] delay in MeetMe   amna saleem
• [Asterisk-Users] delay in MeetMe   amna saleem
• [Asterisk-Users] Re: delay in MeetMe   amna saleem
• [Asterisk-Users] kiax - iax2 softphone   amna saleem
• [Asterisk-Users] delay in MeetMe   amna saleem
• [Asterisk-Users] Music On Hold troubleshooting   amna saleem
• [Asterisk-Users] Music On Hold troubleshooting   amna saleem
• [Asterisk-Users] Query: IAXModem   sanchal.singh at alliance-infotech.com
• [Asterisk-Users] Query   sanchal.singh at alliance-infotech.com
• [Asterisk-Users] query   sanchal.singh at alliance-infotech.com
• [Asterisk-Users] Sending Fax on local host using IAXmodem   sanchal.singh at alliance-infotech.com
• [Asterisk-Users] TSP on linux   sanchal.singh at alliance-infotech.com
• [Asterisk-Users] TSP on Linux   sanchal.singh at alliance-infotech.com
• [Asterisk-Users] username/auth name mismatch   sasa
• [Asterisk-Users] MWI on the PA168V in IAX mode?   sdcharly at gmail.com
• SV: [Asterisk-Users] I can hear only one way when I use nokiae-60withX-lite   sdcharly at gmail.com
• [Asterisk-Users] Re: Nokai E60 and E61 , working fine with Asterisk , with new access points   sdcharly at gmail.com
• [Asterisk-Users] Re: Nokai E60 and E61 , working fine with Asterisk , with new access points   sdcharly at gmail.com
• [Asterisk-Users] Globe7   sdcharly at gmail.com
• [Asterisk-Users] DID in United Arab Emirates, Iran, Kuwaiti, Iraq, Bahrain, Jordan, Saudi Arabia.   sdcharly at gmail.com
• [Asterisk-Users] RE: Digium pound key software appliance opinions   shadowym
• [Asterisk-Users] Fun with Echo   shadowym
• [Asterisk-Users] Fun with Echo   shadowym
• [Asterisk-Users] Hard drive write cache   shadowym
• [Asterisk-Users] Hard drive write cache   shadowym
• [Asterisk-Users] Intel 600SM FXS card   shadowym
• [Asterisk-Users] Hard drive write cache   shadowym
• [Asterisk-Users] Hard drive write cache   shadowym
• [Asterisk-Users] Hard drive write cache   shadowym
• [Asterisk-Users] Single T1 card with Echo Cancellation to workwithDell?   shadowym
• [Asterisk-Users] Easiest (best?) linux distribution for dedicatedAsterisk box?   shadowym
• [Asterisk-Users] Hard drive write cache   shadowym
• [Asterisk-Users] Asterisk 1.07 crash under Debian Sarge   shadowym
• [Asterisk-Users] ECHO Tutorial   shadowym
• [Asterisk-Users] Fun with Echo -- Follow up   shadowym
• [Asterisk-Users] Debian Sarge or CentOS4.3   shadowym
• [Asterisk-Users] GXP-2000 and Shared Line Appearances   shadowym
• [Asterisk-Users] best hardphone for Asterisk?   shadowym
• [Asterisk-Users] GXP-2000 and Shared Line Appearances   shadowym
• [Asterisk-Users] best hardphone for Asterisk?   shadowym
• [Asterisk-Users] Oh oh. Micro$oft just noticed VoIP   shadowym
• [Asterisk-Users] GXP-2000 and Shared Line Appearances   shadowym
• [Asterisk-Users] best hardphone for Asterisk?   shadowym
• [Asterisk-Users] Most stable Asterisk version   shadowym
• [Asterisk-Users] Trixbox maunual configuration   shadowym
• [Asterisk-Users] Asterisk Native Sound Distortion (ulaw)   shadowym
• [Asterisk-Users] Suggested Phone   shadowym
• [Asterisk-Users] Re: Zork and Asterisk   simon-asterisk at uc.org
• [Asterisk-Users] DELL PowerEdge 2850 and TE4110P and TE110P   stoffell
• [Asterisk-Users] bristuff hangup issue   stoffell
• [Asterisk-Users] bristuff hangup issue   stoffell
• [Asterisk-Users] Suggested Phone   stoffell
• [Asterisk-Users] app_sms not working anymore   stoffell
• [Asterisk-Users] app_sms not working anymore   stoffell
• [Asterisk-Users] MBX Servers?   tracinet
• [Asterisk-Users] astdb entry in sip.conf   turby
• [Asterisk-Users] Music On Hold not working with new 1.2.7.1 install   turby
• [Asterisk-Users] Set(CDR(userfield)) Trouble   turby
• [Asterisk-Users] New Member, saying Hi. :)   undrhil.1528785 at bloglines.com
• [Asterisk-Users] fine-tuning asterisk questions   undrhil.1528785 at bloglines.com
• [Asterisk-Users] fine-tuning asterisk questions   undrhil.1528785 at bloglines.com
• [Asterisk-Users] Campusing two Asterisk boxes?   undrhil.1528785 at bloglines.com
• [Asterisk-Users] This is what I want to do...   undrhil.1528785 at bloglines.com
• [Asterisk-Users] sip   undrhil.1528785 at bloglines.com
• [Asterisk-Users] Phone recommendations?   undrhil.1528785 at bloglines.com
• [Asterisk-Users] hangup lag causing the answering of already answered calls   undrhil.1528785 at bloglines.com
• [Asterisk-Users] Question setting up a "bat phone" extension.   undrhil.1528785 at bloglines.com
• [Asterisk-Users] Question setting up a   undrhil.1528785 at bloglines.com
• [Asterisk-Users] How to retrieve voicemail   undrhil.1528785 at bloglines.com
• [Asterisk-Users] ztdummy   undrhil.1528785 at bloglines.com
• [Asterisk-Users] kiax - iax2 softphone   undrhil.1528785 at bloglines.com
• [Asterisk-Users] rollover simulation   undrhil.1528785 at bloglines.com
• [Asterisk-Users] Problem trying to SayDigits when an invalid extension is dialed   undrhil.1528785 at bloglines.com
• [Asterisk-Users] kiax - iax2 softphone   undrhil.1528785 at bloglines.com
• [Asterisk-Users] One problem (MOH) and one question (incoming SIP calls)   undrhil.1528785 at bloglines.com
• [Asterisk-Users] Act-Tel G11112DS Telephony Gateway   undrhil.1528785 at bloglines.com
• [Asterisk-Users] Asking for phone number to dial   undrhil.1528785 at bloglines.com
• [Asterisk-Users] Asking for phone number to dial   undrhil.1528785 at bloglines.com
• [Asterisk-Users] TDMoE question   undrhil.1528785 at bloglines.com
• [Asterisk-Users] FXO for PSTN   undrhil.1528785 at bloglines.com
• [Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!   undrhil.1528785 at bloglines.com
• [Asterisk-Users] asterisk clustering   unplug
• [Asterisk-Users] STNU spport   unplug
• [Asterisk-Users] asterisk load balancing setup   unplug
• [Asterisk-Users] set file path   unplug
• [Asterisk-Users] enable/disable user   unplug
• [Asterisk-Users] get value from DB directly   unplug
• [Asterisk-Users] get value from DB directly   unplug
• [Asterisk-Users] get value from DB directly   unplug
• [Asterisk-Users] get value from DB directly   unplug
• [Asterisk-Users] asterisk load balance   unplug
• [Asterisk-Users] multiple port   unplug
• [Asterisk-Users] fail to make call   unplug
• [Asterisk-Users] Re: fail to make call   unplug
• [Asterisk-Users] show register users   unplug
• [Asterisk-Users] asterisk load balance   unplug
• [Asterisk-Users] Re: fail to make call   unplug
• [Asterisk-Users] Re: fail to make call   unplug
• [Asterisk-Users] Re: fail to make call   unplug
• [Asterisk-Users] Re: fail to make call   unplug
• [Asterisk-Users] using variable   unplug
• [Asterisk-Users] using variable   unplug
• [Asterisk-Users] question about the register/invite call flow   unplug
• [Asterisk-Users] Re: fail to make call   unplug
• [Asterisk-Users] 100 lines + system config   varun
• [Asterisk-Users] 100 lines + system config   varun
• [Asterisk-Users] 100 lines PBX + system config - repost   varun
• [Asterisk-Users] 100 lines PBX + system config - repost   varun
• [Asterisk-Users] 100 lines PBX + system config - repost   varun
• [Asterisk-Users] Anyone have success using LIMIT_PLAYAUDIO_CALLER or LIMIT_PLAYAUDIO_CALLER variables   vcomp
• [Asterisk-Users] Asterisk h323   hakem voip
• [Asterisk-Users] All non US 48 area codes?   voiplist
• [Asterisk-Users] All non US 48 area codes?   voiplist
• [Asterisk-Users] Size limitations of extensions.conf   voiplist
• [Asterisk-Users] Size limitations of extensions.conf   voiplist
• [Asterisk-Users] Sipura SPA-2000 & Asterisk 1.24 w/incoming calls   voiplist
• [Asterisk-Users] Sipura SPA-2000 & Asterisk 1.24 w/incoming calls   voiplist
• [Asterisk-Users] DELL PowerEdge 2850 and TE4110P and TE110P   whois wes
• [Asterisk-Users] Restricting amount of incoming calls   whois wes
• [Asterisk-Users] [HELP] - Sound cutting and dropping out - 1.2.7.1/Sangoma/PowerEdge 2850   whois wes
• [Asterisk-Users] [HELP] - Sound cutting and dropping out - 1.2.7.1/Sangoma/PowerEdge 2850   whois wes
• [Asterisk-Users] Anyone have success using LIMIT_PLAYAUDIO_CALLER or LIMIT_PLAYAUDIO_CALLER variables   whois wes
• [Asterisk-Users] Re: Audio problems on Zap & SIP, local network, not IRQ related?   whois wes
• [Asterisk-Users] SIP 486 "Busy Here"   whois wes
• [Asterisk-Users] Re: Audio problems on Zap & SIP, local netwo rk, not IRQ related?   whois wes
• [Asterisk-Users] RE: Dropped calls continued   whois wes
• [Asterisk-Users] Sangoma A104D is dropping DTMF digits during IVR   whois wes
• [Asterisk-Users] Re: Sangoma A101 configuration   yusuf
• [Asterisk-Users] Music On Hold not working with new 1.2.7.1 install   | Rurouni Alucard |