February 2006 Archives by author

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Starting: Wed Feb 1 00:18:46 MST 2006
Ending: Tue Feb 28 23:59:10 MST 2006
Messages: 3812

• [Asterisk-Users] Asterisk with T1 card on laptop   Arnaud
• [Asterisk-Users] meetme and dtmf   Francesco Peeters (Asterisk)
• [Asterisk-Users] meetme and dtmf   Francesco Peeters (Asterisk)
• [Asterisk-Users] RE: Euro-ISDN   Francesco Peeters (Asterisk)
• [Asterisk-Users] meetme and dtmf   Francesco Peeters (Asterisk)
• [Asterisk-Users] 1 ISDN BRI to IAX2/SIP... (*) best tool or?...   Francesco Peeters (Asterisk)
• [Asterisk-Users] Problem with ZAPHFC: internal S0 hangs when hanging up   Francesco Peeters (Asterisk)
• [Asterisk-Users] 1 ISDN BRI to IAX2/SIP... (*) best tool or?...   Francesco Peeters (Asterisk)
• [Asterisk-Users] Random Disconnects - or ARE they?   Francesco Peeters (Asterisk)
• [Asterisk-Users] S100-FX v2.0   Chris Earle (CBL)
• SV: [Asterisk-Users] BAD/GOOD Echo Cancel   Chris Earle (CBL)
• [Asterisk-Users] Fritz card technology & German *   Chris Earle (CBL)
• [Asterisk-Users] Re: Connection TDM400P to UK PSTN   Chris Earle (CBL)
• [Asterisk-Users] Issues in Australia? Ringing, iaxy etc   Chris Earle (CBL)
• [Asterisk-Users] indications issues in Singapore?   Chris Earle (CBL)
• [Asterisk-Users] indications issues in Singapore?   Chris Earle (CBL)
• [Asterisk-Users] Cannot boot machine up after working on zaptel....   Chris Earle (CBL)
• [Asterisk-Users] Cannot boot machine up after working on zaptel....   Chris Earle (CBL)
• [Asterisk-Users] Auto login via Remote User   Chris Earle (CBL)
• [Asterisk-Users] Auto login via Remote User   Chris Earle (CBL)
• [Asterisk-Users] Best ATA for general residential deployment??   Andre Rodrigues (Cheyenne)
• [Asterisk-Users] Receiving faxes with spandsp - strange problem   Matt Riddell (IT)
• [Asterisk-Users] Events when the target of the call answer   Matt Riddell (IT)
• [Asterisk-Users] cmd set with multiple values   Matt Riddell (IT)
• [Asterisk-Users] RE: Rewind MusicOnHold?   Matt Riddell (IT)
• [Asterisk-Users] seg fault 1.2.4   Matt Riddell (IT)
• [Asterisk-Users] Caller ID patches - updated   Marc McLaughlin (LUSYN)
• [Asterisk-Users] (newby) Is PING a good indicator of latency?   Chris Mason (Lists)
• [Asterisk-Users] spa3000   Chris Mason (Lists)
• [Asterisk-Users] IAX2 through Shorewall rpoblem   Chris Mason (Lists)
• [Asterisk-Users] Best ATA for general residential deployment??   Chris Mason (Lists)
• [Asterisk-Users] Leftover sound on isdn modem channel   Francesco Peeters (PalmOS)
• [Asterisk-Users] Re: [Amug] Next Montreal meeting - the 21st - featuring a conference call with Mark Spencer   Michel Belleau (malaiwah.com)
• [Asterisk-Users] RES: RTP and Signalling   ITN Info - 11-3898-0112
• [Asterisk-Users] MixMonitor & ControlPlayback of g729 files   1 2
• [Asterisk-Users] SIP compact headers   1 2
• [Asterisk-Users] Landmark digital key systems and Asterisk   John Fix 3rd
• [Asterisk-Users] IAX registration expiration   Vincent Régnard
• [Asterisk-Users] IAX registration expiration   Vincent Régnard
• [Asterisk-Users] Is SIP "canreinvite" working ok?   Álvaro Palma
• [Asterisk-Users] Codec order sent wrong from Asterisk   Álvaro Palma
• [Asterisk-Users] How can I force Asterisk t not override my codec order?   Álvaro Palma
• [Asterisk-Users] Asterisk native sounds now available!   Benoît Mérouze
• [Asterisk-Users] Sagoma w/EC x TE411   Bruno de Assumpção Loureiro
• [Asterisk-Users] Sagoma w/EC x TE411   Bruno de Assumpção Loureiro
• [Asterisk-Users] SIP groups   María Chóliz
• [Asterisk-Users] Problems dialing to another Asterisk server   María Chóliz
• [Asterisk-Users] SRV mapped to host   Marc Patino Gómez
• [Asterisk-Users] calling from SIP to a h.323 device with oh323   Marc Patino Gómez
• [Asterisk-Users] Install instructions for FOP Flash Operator Panel do not make sense...   Nicolás Gudiño
• [Asterisk-Users] mpg123 alternative?   Nicolás Gudiño
• [Asterisk-Users] voicemail recording format   Vincent Régnard
• [Asterisk-Users] Setting up an EICON CARD with CAPI   cédric Buzay
• [Asterisk-Users] Setting up an EICON CARD with CAPI   cédric Buzay
• [Asterisk-Users] sipura 3000 and other probs   Jean-François Rousseau
• [Asterisk-Users] RE: Asterisk-Users Digest, Vol 18, Issue 206   Michaël Gaudette
• [Asterisk-Users] RE: Asterisk-Users Digest, Vol 19, Issue 6   Michaël Gaudette
• [Asterisk-Users] Digit timeouts vs includes in diaplan   Michaël Gaudette
• [Asterisk-Users] SIP question   Michaël Gaudette
• [Asterisk-Users] Re: SIP question   Michaël Gaudette
• [Asterisk-Users] No audio for outgoing calls   Michaël Gaudette
• [Asterisk-Users] One way audio - it doesn't make sense   Michaël Gaudette
• [Asterisk-Users] RE: One way audio - it doesn't make sense   Michaël Gaudette
• [Asterisk-Users] Opinions needed on call quality vs network latency   Michaël Gaudette
• [Asterisk-Users] Re: Opinions needed on call quality vs   Michaël Gaudette
• [Asterisk-Users] RE: Asterisk-Users Digest, Vol 19, Issue 47   Michaël Gaudette
• [Asterisk-Users] Make Meetme start only when somebody puts in the admin PIN   Michaël Gaudette
• [Asterisk-Users] Why is asterisk ignoring my context?   Michaël Gaudette
• [Asterisk-Users] RE: Asterisk-Users Digest, Vol 19, Issue 89   Michaël Gaudette
• [Asterisk-Users] RE: Asterisk-Users Digest, Vol 19, Issue 90   Michaël Gaudette
• [Asterisk-Users] Voicemail problems   Michaël Gaudette
• [Asterisk-Users] Re: Voicemail problems   Michaël Gaudette
• [Asterisk-Users] Re: Voicemail problems   Michaël Gaudette
• [Asterisk-Users] Send flash through zap channel   Stefan Märkle
• [Asterisk-Users] Re: Send flash through zap channel   Stefan Märkle
• [Asterisk-Users] agents.conf   Tomislav Parčina
• [Asterisk-Users] What ATA should I buy?   Tomislav Parčina
• [Asterisk-Users] Queue - joinempty   Tomislav Parčina
• [Asterisk-Users] Queue - check agent   Tomislav Parčina
• [Asterisk-Users] Queue transfer   Tomislav Parčina
• [Asterisk-Users] Voicemail - direct call   Tomislav Parčina
• [Asterisk-Users] Call centre - * hang's up   Tomislav Parčina
• [Asterisk-Users] SIP Register   Tomislav Parčina
• [Asterisk-Users] RE: SIP Register   Tomislav Parčina
• [Asterisk-Users] RE: Queue - check agent   Tomislav Parčina
• [Asterisk-Users] RE: SIP Register   Tomislav Parčina
• [Asterisk-Users] RE: virtual extension per user ?   Tomislav Parčina
• [Asterisk-Users] Re: RE: virtual extension per user ?   Tomislav Parčina
• [Asterisk-Users] RE: RE: virtual extension per user ?   Tomislav Parčina
• [Asterisk-Users] RE: What ATA should I buy?   Tomislav Parčina
• [Asterisk-Users] Re: What ATA should I buy?   Tomislav Parčina
• [Asterisk-Users] RE: What ATA should I buy?   Tomislav Parčina
• [Asterisk-Users] Re: Re: asterisk logger - urgent!!!   Tomislav Parčina
• [Asterisk-Users] Re: SIP groups   Tomislav Parčina
• [Asterisk-Users] Re: Voicemail - direct call   Tomislav Parčina
• [Asterisk-Users] Re: segmentation fault   Tomislav Parčina
• [Asterisk-Users] Re: Call centre - * hang's up   Tomislav Parčina
• [Asterisk-Users] Re: Call centre - * hang's up   Tomislav Parčina
• [Asterisk-Users] Re: Re: RE: virtual extension per user ?   Tomislav Parčina
• [Asterisk-Users] Re: Outbound ZAP Dialing   Tomislav Parčina
• [Asterisk-Users] Re: Re: Voicemail - direct call   Tomislav Parčina
• [Asterisk-Users] Re: Re: Call centre - * hang's up   Tomislav Parčina
• [Asterisk-Users] Re: Linear Queues Strategies for 3rd Party Application   Tomislav Parčina
• [Asterisk-Users] Re: Fromstring when sending e-mail on recieved voicemail   Tomislav Parčina
• [Asterisk-Users] Re: Call queue design issues and suggestions   Tomislav Parčina
• [Asterisk-Users] Cisco 79xx firmware   Tomislav Parčina
• [Asterisk-Users] Cisco 79xx <=> Asterisk - SIP or SCCP?   Tomislav Parčina
• [Asterisk-Users] Re: Cisco 79xx firmware   Tomislav Parčina
• [Asterisk-Users] Re: FC4 and yum install; how to configure questions   Tomislav Parčina
• [Asterisk-Users] Re: Keep getting message in logs that pbx.c cannot find extension context 'default'   Tomislav Parčina
• [Asterisk-Users] Re: Important: Application DIALPLAN STANDARD/GUIDELINES needs to be established.   Tomislav Parčina
• [Asterisk-Users] Asttapi - what's wrong?   Tomislav Parčina
• [Asterisk-Users] Re: How can I debug spandsp?   Tomislav Parčina
• [Asterisk-Users] Re: res_features pickupexten   Tomislav Parčina
• [Asterisk-Users] My or provider error?   Tomislav Parčina
• [Asterisk-Users] Re: Re: How can I debug spandsp?   Tomislav Parčina
• [Asterisk-Users] Re: Re: res_features pickupexten   Tomislav Parčina
• [Asterisk-Users] Re: Asttapi - what's wrong?   Tomislav Parčina
• [Asterisk-Users] Asterisk hangs up - h323   Tomislav Parčina
• RE: [Asterisk-Users] asterisk and week-ends   Colin Anderson
• [Asterisk-Users] DTMF Mode supported by VoiceMail Application   Fabian Müller
• [Asterisk-Users] Unknown RTP codec 100 received   Fabian Müller
• [Asterisk-Users] SetCallerID and CDR   Adrian A
• [Asterisk-Users] Asterisk native sounds now available!   Adrian A
• [Asterisk-Users] How can a caller go back to the main menu from a queue?   Zach A
• [Asterisk-Users] Help on queues   Zach A
• [Asterisk-Users] Help on queues   Zach A
• [Asterisk-Users] Help on queues   Zach A
• [Asterisk-Users] Help on queues   Zach A
• [Asterisk-Users] Suggestions/comments needed on the following Asterisk step by step installation instructions   Zach A
• [Asterisk-Users] Suggestions/comments needed on the following Asterisk step by step installation instructions   Zach A
• [Asterisk-Users] Fedora Core 3 or Fedora Core 4? yum update or not?   Zach A
• [Asterisk-Users] Disabling SELinux in FC3 - good or bad   Zach A
• [Asterisk-Users] What to know for installing ARI   Zach A
• [Asterisk-Users] Connecting SPA3000 to Bell Phone Line   Zach A
• [Asterisk-Users] PSTN to SIP MoH always choppy, SIP to SIP good   Zach A
• [Asterisk-Users] Asterisk on SPA-3000 rings extensions only after third ring   Zach A
• [Asterisk-Users] Asterisk on SPA-3000 rings extensions only after third ring   Zach A
• [Asterisk-Users] Asterisk on SPA-3000 rings extensions only afterthird ring   Zach A
• [Asterisk-Users] Is Asterisk a PBX?   Zach A
• [Asterisk-Users] Channel SS7   ADEGOKE ARUNA
• [Asterisk-Users] Calling number rewriting   ADEGOKE ARUNA
• [Asterisk-Users] NEED COMMENT ON USING FEDORA CORE 3   ADEGOKE ARUNA
• [Asterisk-Users] Asterisk + WiFi Phones   ADEGOKE ARUNA
• [Asterisk-Users] realtime queue not working realtime in asterisk versions above 1.2.0   Frank Aartman
• [Asterisk-Users] FXO Line not Hanged up   KaveH Aasaraai
• [Asterisk-Users] Zap, Caller ID problem   KaveH Aasaraai
• [Asterisk-Users] Asterisk Registering with SER question   Abhishek
• [Asterisk-Users] Auto dialing   Abhishek
• [Asterisk-Users] HT-1000 chipset experience   Master Abi
• [Asterisk-Users] TDM Cross-connection   Alejandro Acosta
• [Asterisk-Users] CallerID Problem   Jason Adams
• [Asterisk-Users] Dumb Dialout Question   Jason Adams
• [Asterisk-Users] Grandstream hold one way audio -URGENT   Jason Adams
• [Asterisk-Users] Strange echo phenomenon (double tandem)   Rich Adamson
• [Asterisk-Users] winnipeg canada   Rich Adamson
• [Asterisk-Users] BAD/GOOD Echo Cancel   Rich Adamson
• [Asterisk-Users] Coppercom SIP experience?   Rich Adamson
• [Asterisk-Users] Fedora Core 3 or Fedora Core 4? yum update or not?   Rich Adamson
• [Asterisk-Users] Faint background noise/crackle on FXS port on TDM400P   Rich Adamson
• [Asterisk-Users] RE: X100P help required   Rich Adamson
• [Asterisk-Users] Bandwidth: to seperate or not to seperate   Rich Adamson
• [Asterisk-Users] Bandwidth: to seperate or not to seperate   Rich Adamson
• [Asterisk-Users] Connecting SPA3000 to Bell Phone Line   Rich Adamson
• [Asterisk-Users] Best quad-port fxo solution with EC?   Rich Adamson
• [Asterisk-Users] GR303   Rich Adamson
• [Asterisk-Users] FXO port on TDM400P hangs!!   Rich Adamson
• [Asterisk-Users] Random Hangups/Disconnects   Rich Adamson
• [Asterisk-Users] Aasterisk large-scale deployment w/analog phones   Rich Adamson
• [Asterisk-Users] MOH from RCA jack?   Rich Adamson
• [Asterisk-Users] MOH from RCA jack?   Rich Adamson
• [Asterisk-Users] MOH from RCA jack?   Rich Adamson
• [Asterisk-Users] MOH from RCA jack?   Rich Adamson
• [Asterisk-Users] Application Faxing using SIP   Rich Adamson
• [Asterisk-Users] Cisco 7960 Register Problem   Rich Adamson
• [Asterisk-Users] spa3000   Rich Adamson
• [Asterisk-Users] spa3000   Rich Adamson
• [Asterisk-Users] Multiple TDM400P's in a single machine   Rich Adamson
• [Asterisk-Users] sniffing sip password/uri/host info   Rich Adamson
• [Asterisk-Users] Sangoma A200D analog card with fxo's   Rich Adamson
• [Asterisk-Users] BAD/GOOD Echo Cancel   Rich Adamson
• [Asterisk-Users] NEED COMMENT ON USING FEDORA CORE 3   Rich Adamson
• [Asterisk-Users] how to add stun functionality in asterisk   Rich Adamson
• [Asterisk-Users] Asterisk on SPA-3000 rings extensions only after third ring   Rich Adamson
• [Asterisk-Users] Sangoma A200D analog card with fxo's   Rich Adamson
• [Asterisk-Users] did from sip trunk   Rich Adamson
• [Asterisk-Users] Asterisk on SPA-3000 rings extensions only after third ring   Rich Adamson
• [Asterisk-Users] What business IP phone to use   Rich Adamson
• [Asterisk-Users] Cisco 79xx firmware   Rich Adamson
• [Asterisk-Users] IAX2 through Shorewall rpoblem   Rich Adamson
• [Asterisk-Users] mpg123 alternative?   Rich Adamson
• [Asterisk-Users] Calls not going through   Rich Adamson
• [Asterisk-Users] mpg123 alternative?   Rich Adamson
• [Asterisk-Users] Voicemail problems   Rich Adamson
• [Asterisk-Users] OT: VoIP over bonded link   Rich Adamson
• [Asterisk-Users] mpg123 alternative?   Rich Adamson
• [Asterisk-Users] Polycom IP601 Question   Rich Adamson
• [Asterisk-Users] Re: Asterisk 1.2.4 IAX2 New Jitterbuffer Tuning   Rich Adamson
• [Asterisk-Users] Analyzer for Milliwatt   Rich Adamson
• [Asterisk-Users] mpg123 alternative?   Rich Adamson
• [Asterisk-Users] GPS-enabled cell phone/PDA   Rich Adamson
• [Asterisk-Users] fax receive using TDM400P   Rich Adamson
• [Asterisk-Users] What business IP phone to use   Rich Adamson
• [Asterisk-Users] OT: VoIP over bonded link   Rich Adamson
• [Asterisk-Users] What business IP phone to use   Rich Adamson
• [Asterisk-Users] fax receive using TDM400P   Rich Adamson
• [Asterisk-Users] What business IP phone to use   Rich Adamson
• [Asterisk-Users] fax receive using TDM400P   Rich Adamson
• [Asterisk-Users] What business IP phone to use   Rich Adamson
• [Asterisk-Users] What business IP phone to use   Rich Adamson
• [Asterisk-Users] Sound issue   Rich Adamson
• [Asterisk-Users] jitterbuffer and DTMF conflict?   Rich Adamson
• [Asterisk-Users] Covad anyone ...   Rich Adamson
• [Asterisk-Users] Echo on PRI/BRI?   Rich Adamson
• [Asterisk-Users] Weird DTMF issue   Rich Adamson
• [Asterisk-Users] Weird DTMF issue   Rich Adamson
• [Asterisk-Users] fax receive using TDM400P   Rich Adamson
• [Asterisk-Users] No Voice when canreinvite=no   Kamran Ahmad
• [Asterisk-Users] asterisk-1.2.4 + asterisk-addons-1.2.1 for mysql realtime   Kamran Ahmad
• [Asterisk-Users] asterisk-1.2.4 + asterisk-addons-1.2.1 for mysql realtime   Kamran Ahmad
• [Asterisk-Users] meetme and dtmf   Imran Ahmed
• [Asterisk-Users] meetme and dtmf   Imran Ahmed
• [Asterisk-Users] meetme and dtmf   Imran Ahmed
• [Asterisk-Users] Rtp packets being dropped   Imran Ahmed
• [Asterisk-Users] echo cancel from telco   Imran Ahmed
• [Asterisk-Users] echo cancel from telco   Imran Ahmed
• [Asterisk-Users] lists problem, Gmail????????   Imran Ahmed
• [Asterisk-Users] XLite dtmf issue?   Aisling
• [Asterisk-Users] XLite dtmf issue?   Aisling
• [Asterisk-Users] varion card   Akpome Akpoguma
• [Asterisk-Users] chan_capi setting ${DNIS}   Nathan Alberti
• [Asterisk-Users] chan_capi setting ${DNIS}   Nathan Alberti
• [Asterisk-Users] username as extension   Nathan Alberti
• [Asterisk-Users] username as extension   Nathan Alberti
• [Asterisk-Users] username as extension   Nathan Alberti
• [Asterisk-Users] register => 2345:password@sip_proxy doesn't care about port   Alex
• [Asterisk-Users] register => 2345:password@sip_proxy doesn't care about port   Alex
• [Asterisk-Users] Re: delaying "answer" for a number of ringsor an amount   ammar Ali
• [Asterisk-Users] Re: delaying "answer" for a number of ringsor anamount   ammar Ali
• [Asterisk-Users] Re: delaying "answer" for a number of ringsor anamount   ammar Ali
• [Asterisk-Users] run with incorrect E1/T1 jumper settings   Diyanat Ali
• [Asterisk-Users] How to install Zaptel?   Larry Alkoff
• [Asterisk-Users] Dumb question... block 00   Pablo Allietti
• [Asterisk-Users] A unique 'click to call' project - Could use some advice   Aloi, Christopher
• [Asterisk-Users] A unique 'click to call' project - Could usesomeadvice   Aloi, Christopher
• [Asterisk-Users] A unique 'click to call' project - Could usesome advice   Aloi, Christopher
• [Asterisk-Users] A unique 'click to call' project - Could usesome advice   Aloi, Christopher
• [Asterisk-Users] asttapi 0.08 - the memory could not be written   Victor Alvarez
• [Asterisk-Users] Regarding cdr_manager.conf   Victor Alvarez
• [Asterisk-Users] How can I debug spandsp?   Victor Alvarez
• [Asterisk-Users] Help with sip setup because can't receive calls!!!!!!   Facundo Ameal
• [Asterisk-Users] CallerID popup   Facundo Ameal
• [Asterisk-Users] CallerID popup   Facundo Ameal
• [Asterisk-Users] CallerID popup   Facundo Ameal
• [Asterisk-Users] change languages from an IVR   Facundo Ameal
• [Asterisk-Users] Asterisk with USB   Facundo Ameal
• [Asterisk-Users] Asterisk with USB   Facundo Ameal
• [Asterisk-Users] Asterisk with USB   Facundo Ameal
• [Asterisk-Users] Asterisk with USB   Facundo Ameal
• [Asterisk-Users] BAD/GOOD Echo Cancel   Richard Amerman
• [Asterisk-Users] Asterisk vs. Traditional PBX   Richard Amerman
• [Asterisk-Users] Aastra phones and common directory?   Richard Amerman
• [Asterisk-Users] Grandstream GXP-2000   Richard Amerman
• [Asterisk-Users] Grandstream GXP-2000   Richard Amerman
• [Asterisk-Users] Grandstream GXP-2000   Richard Amerman
• [Asterisk-Users] Fedora Core 3 or Fedora Core 4? yum update ornot?   Ryan Amos
• [Asterisk-Users] Dell PowerEdge 1800 and TE410P   Ryan Amos
• [Asterisk-Users] Asterisk vs. Traditional PBX   Ryan Amos
• [Asterisk-Users] Dell PowerEdge 2850   Ryan Amos
• [Asterisk-Users] routing question: multipath routing for SIP   Zac Amsler
• [Asterisk-Users] Limiting SIP bandwidth   Trond G. Andersen
• [Asterisk-Users] Will not authenticate incoming VOIP provider calls   Colin Anderson
• [Asterisk-Users] FXS with v.90 modem support?   Colin Anderson
• [Asterisk-Users] change languages from an IVR   Colin Anderson
• [Asterisk-Users] Asterisk native sounds now available!   Colin Anderson
• [Asterisk-Users] change languages from an IVR   Colin Anderson
• [Asterisk-Users] Mitel 5220 IP phones   Colin Anderson
• [Asterisk-Users] Fedora Core 3 or Fedora Core 4? yum update o r not? also: SpanDSP -pre25 for 1.0.9 is out w00t!   Colin Anderson
• [Asterisk-Users] change languages from an IVR   Colin Anderson
• [Asterisk-Users] Handset phone to replace Flash Operator Pane l   Colin Anderson
• [Asterisk-Users] Possible for Asterisk to output CLID to invo ke 3rd party app?   Colin Anderson
• [Asterisk-Users] Asterisk vs. Traditional PBX   Colin Anderson
• [Asterisk-Users] Asterisk and Snom 360   Colin Anderson
• [Asterisk-Users] [CAVPdiscussion] OT: RFC: Canadian Association o f Voice over IP Users (CAVU)   Colin Anderson
• [Asterisk-Users] is there a web interface to this mailing lis t?   Colin Anderson
• [Asterisk-Users] Random Disconnects - or ARE they?   Colin Anderson
• [Asterisk-Users] Safely editing voicemail.conf   Colin Anderson
• [Asterisk-Users] A unique 'click to call' project - Could use some advice   Colin Anderson
• [Asterisk-Users] A unique 'click to call' project - Could use some advice   Colin Anderson
• [Asterisk-Users] A unique 'click to call' project - Could use some advice   Colin Anderson
• [Asterisk-Users] A unique 'click to call' project - Could use some advice <--one thing I forgot   Colin Anderson
• [Asterisk-Users] What business IP phone to use   Colin Anderson
• [Asterisk-Users] OT: VoIP over bonded link   Colin Anderson
• [Asterisk-Users] Asterisk Follow Me   Colin Anderson
• [Asterisk-Users] OT: VoIP over bonded link   Colin Anderson
• [Asterisk-Users] OT: VoIP over bonded link   Colin Anderson
• [Asterisk-Users] What business IP phone to use   Colin Anderson
• [Asterisk-Users] Is Asterisk a PBX?   Colin Anderson
• [Asterisk-Users] fax receive using TDM400P   Colin Anderson
• [Asterisk-Users] Cannot boot machine up after working on zapt el....   Colin Anderson
• [Asterisk-Users] Cannot boot machine up after working on zapt el....   Colin Anderson
• [Asterisk-Users] Capturing DIALSTATUS on a PARTICULAR channel if multiple-dialling?   Colin Anderson
• [Asterisk-Users] Problem with two cards Digium   Colin Anderson
• [Asterisk-Users] Capturing DIALSTATUS on a PARTICULAR channel if multiple-dialling?   Colin Anderson
• [Asterisk-Users] Asterisk and Cisco AS5350   Micke Andersson
• [Asterisk-Users] Prepaid / postpaid solution   Micke Andersson
• [Asterisk-Users] Prepaid / postpaid solution   Micke Andersson
• [OFFLIST] Re: [Asterisk-Users] run with incorrect E1/T1 jumper settings   Micke Andersson
• [Asterisk-Users] run with incorrect E1/T1 jumper settings   Robert Andersson
• [Asterisk-Users] run with incorrect E1/T1 jumper settings   Robert Andersson
• [Asterisk-Users] g729 license question   Andres
• [Asterisk-Users] PRI to PRI not passing callerid   Andres
• [Asterisk-Users] Dell PowerEdge 1800 and TE410P   Andres
• [Asterisk-Users] Working SPA 841s now return 404 Not Found for INVITES and OPTION packets from *   Andres
• [Asterisk-Users] RE: Asterisk-Users Digest, Vol 19, Issue 89   Andres
• [Asterisk-Users] Telmex PRI line configuration problem   Andres
• [Asterisk-Users] always on agents and queues   Andres
• [Asterisk-Users] A room full of Cisco 7960s behind NAT   Andres
• [Asterisk-Users] ParkAndAnnounce2 Feature Request   Steven Andres
• [Asterisk-Users] ParkAndAnnounce2 Feature Request   Steven Andres
• [Asterisk-Users] ParkAndAnnounce2 Feature Request   Steven Andres
• [Asterisk-Users] ParkAndAnnounce2 Feature Request   Steven Andres
• [Asterisk-Users] supermicro server model   Cory Andrews
• [Asterisk-Users] hardware suggestion   Cory Andrews
• [Asterisk-Users] Asterisk vs. Traditional PBX   Cory Andrews
• [Asterisk-Users] TE411P Really Bad Echo   Cory Andrews
• [Asterisk-Users] Recommended rack-mountable server anyone?   Cory Andrews
• [Asterisk-Users] What business IP phone to use   Cory Andrews
• [Asterisk-Users] Linksys WIP300 WiFi Phone   Cory Andrews
• [Asterisk-Users] What business IP phone to use   Cory Andrews
• [Asterisk-Users] Voice Over WiFi   Cory Andrews
• [Asterisk-Users] Covad anyone ...   Cory Andrews
• [Asterisk-Users] Asterisk + WiFi Phones   Cory Andrews
• [Asterisk-Users] Zoom 5801 problems with *   Cory Andrews
• [Asterisk-Users] queue behaviour   Francesco Angi
• [Asterisk-Users] pickup problem on Asterisk 1.2.4   Francesco Angi
• R: [Asterisk-Users] queue behaviour   Francesco Angi
• [Asterisk-Users] pickup problem on Asterisk 1.2.4   Francesco Angi
• [Asterisk-Users] courtesy message calling mobile phones   Francesco Angi
• R: [Asterisk-Users] Re: courtesy message calling mobile phones   Francesco Angi
• [Asterisk-Users] pickup problem on Asterisk 1.2.4   Francesco Angi
• [Asterisk-Users] Bridged line appearance   David Ankers
• [Asterisk-Users] RE: Cisco 7905 can't register   David Ankers
• [Asterisk-Users] Queue Messages now playing when caller is insidequeue   David Ankers
• [Asterisk-Users] spa3000   David Ankers
• [Asterisk-Users] g729 quality at GSM bitrates   David Ankers
• [Asterisk-Users] mpg123 alternative?   David Ankers
• [Asterisk-Users] What business IP phone to use   David Ankers
• [Asterisk-Users] What business IP phone to use   David Ankers
• [Asterisk-Users] GPS-enabled cell phone/PDA   David Ankers
• [Asterisk-Users] click to talk   Samy Antoun
• [Asterisk-Users] Agent supervisor configuration   kritikus Araklidas
• [Asterisk-Users] Streaming MOH   Lee Archer
• [Asterisk-Users] Outbound Caller ID number on E1   Lee Archer
• [Asterisk-Users] Streaming MOH   Lee Archer
• [Asterisk-Users] Musiconhold in zapata.conf   Lee Archer
• [Asterisk-Users] Double ring   Lee Archer
• [Asterisk-Users] Double ring   Lee Archer
• [Asterisk-Users] Double ring   Lee Archer
• [Asterisk-Users] Double ring   Lee Archer
• [Asterisk-Users] Double ring   Lee Archer
• [Asterisk-Users] Double ring   Lee Archer
• [Asterisk-Users] Firmware version 1.3.1 released for Aastra IPphones   Lee Archer
• [Asterisk-Users] Firmware version 1.3.1 released for AastraIPphones   Lee Archer
• [Asterisk-Users] Firmware version 1.3.1 released for Aastra IPphones   Lee Archer
• [Asterisk-Users] RE: Firmware version 1.3.1 released for Aastra IPphones   Lee Archer
• [Asterisk-Users] Grandstream GXP-2000   Lee Archer
• [Asterisk-Users] RE: Firmware version 1.3.1 released for AastraIPphones   Lee Archer
• [Asterisk-Users] Grandstream GXP-2000   Lee Archer
• [Asterisk-Users] Streaming Music On Hold   Lee Archer
• [Asterisk-Users] mpg123 alternative?   Lee Archer
• [Asterisk-Users] mpg123 alternative?   Lee Archer
• [Asterisk-Users] Multiple TDM400P's in a single machine   Marc Archer
• [Asterisk-Users] Is it possible ?   Sohail Arham
• [Asterisk-Users] FXO port on TDM400P hangs!!   Jared Armstrong
• [Asterisk-Users] FXO port on TDM400P hangs!!   Jared Armstrong
• [Asterisk-Users] Application Faxing using SIP   Sanjay Arora
• [Asterisk-Users] Some Hardware & Asterisk Applications Questions   Sanjay Arora
• [Asterisk-Users] attended call transfer   Thomas Artner
• [Asterisk-Users] attended call transfer   Thomas Artner
• [Asterisk-Users] asterisk + door opener   Thomas Artner
• [Asterisk-Users] attended call transfer   Thomas Artner
• [Asterisk-Users] asterisk + door opener   Thomas Artner
• [Asterisk-Users] Multiple TDM400P's in a single machine   Thomas Artner
• [Asterisk-Users] fax receive using TDM400P   Thomas Artner
• [Asterisk-Users] fax receive using TDM400P   Thomas Artner
• [Asterisk-Users] fax receive using TDM400P   Thomas Artner
• [Asterisk-Users] fax receive using TDM400P   Thomas Artner
• [Asterisk-Users] fax receive using TDM400P   Thomas Artner
• [Asterisk-Users] fax receive using TDM400P   Thomas Artner
• [Asterisk-Users] fax receive using TDM400P   Thomas Artner
• [Asterisk-Users] fax receive using TDM400P   Thomas Artner
• [Asterisk-Users] Asterisk - Brooktrout   Stephen Arulraj
• [Asterisk-Users] asterisk + door opener   Stephen Arulraj
• [Asterisk-Users] Asterisk Contact Center   Stephen Arulraj
• [Asterisk-Users] Asterisk and Hipath interconnections   Stephen Arulraj
• [Asterisk-Users] Asterisk and Hipath interconnections   Stephen Arulraj
• [Asterisk-Users] Voice Over WiFi   Bjorn Asmul
• [Asterisk-Users] Important: Application DIALPLAN STANDARD/GUIDELINES needs to be established.   Asterisk
• [Asterisk-Users] Double ring   Domjan Attila
• [Asterisk-Users] Double ring   Domjan Attila
• [Asterisk-Users] iax2 native transfer question.   Atuc
• [Asterisk-Users] MOH from RCA jack?   Jonathan Augenstine
• [Asterisk-Users] Streaming Music On Hold   Jonathan Augenstine
• [Asterisk-Users] RE: Is my math on traffic/bandwidth correct?   Robert Augustyn
• [Asterisk-Users] Asterisk SIP phones to Cisco Unity via CCM4.0SIPTrunk   Dan Austin
• [Asterisk-Users] Asterisk SIP phones to Cisco Unity viaCCM4.0SIPTrunk   Dan Austin
• [Asterisk-Users] [Announce] Web-MeetMe v2.0.0   Dan Austin
• [Asterisk-Users] [Announce] Web-MeetMe v2.0.0   Dan Austin
• [Asterisk-Users] [Announce] Web-MeetMe v2.0.0   Dan Austin
• [Asterisk-Users] Uniden UIP200 and Asterisk v1.2.4: problem not registering   Jean-Yves Avenard
• [Asterisk-Users] Uniden UIP200 and Asterisk v1.2.4: problem notregistering   Jean-Yves Avenard
• [Asterisk-Users] meetme and dtmf   Accursio Avona
• [Asterisk-Users] meetme and dtmf   Accursio Avona
• [Asterisk-Users] meetme and dtmf   Accursio Avona
• [Asterisk-Users] meetme and dtmf   Accursio Avona
• [Asterisk-Users] meetme and dtmf   Accursio Avona
• [Asterisk-Users] meetme and dtmf   Accursio Avona
• [Asterisk-Users] meetme and dtmf   Accursio Avona
• [Asterisk-Users] no sound in NAT configuration scenrios   Amir Aziz
• [Asterisk-Users] How can I configure to call from the console by means of a sip phone,   Anthony Azzopardi
• [Asterisk-Users] How can I configure to call from the console bymeans of a sip phone,   Anthony Azzopardi
• [Asterisk-Users] How can I send DTMF from the console?   Anthony Azzopardi
• [Asterisk-Users] How come I don't have the MeetMe application registered?   Anthony Azzopardi
• [Asterisk-Users] Can I configure the console to ring on one sound card and the headset on another sound card?   Anthony Azzopardi
• [Asterisk-Users] Where can I get the tar.gz sources of libnewt?   Anthony Azzopardi
• [Asterisk-Users] How to handle "provider UNREACHABLE" in the dialplan?   Michiel van Baak
• [Asterisk-Users] How to handle "provider UNREACHABLE" in the dialplan?   Michiel van Baak
• [Asterisk-Users] (newby) Asterisk on the open internet & security   Michiel van Baak
• [Asterisk-Users] Codec Selection   Michiel van Baak
• [Asterisk-Users] Billing inbound calls per minute   Michiel van Baak
• [Asterisk-Users] cosmetic bug on CLI ?   Michiel van Baak
• [Asterisk-Users] RE: ex-girlfriend (ex-boyfriend)   Michiel van Baak
• [Asterisk-Users] Sendmail with exchange   Michiel van Baak
• [Asterisk-Users] TE411P Really Bad Echo   Michiel van Baak
• [Asterisk-Users] IP phone with many speed dial buttons   Michiel van Baak
• [Asterisk-Users] RE: virtual extension per user ?   Michiel van Baak
• [Asterisk-Users] Re: RE: virtual extension per user ?   Michiel van Baak
• FW: [Asterisk-Users] AGI onAnswer function: does it exist?   Michiel van Baak
• [Asterisk-Users] A unique 'click to call' project - Could use some advice   Michiel van Baak
• [Asterisk-Users] Cheap BRI card   Michiel van Baak
• [Asterisk-Users] A unique 'click to call' project - Could use some advice <--one thing I forgot   Michiel van Baak
• [Asterisk-Users] Grandstream GXP-2000   Michiel van Baak
• [Asterisk-Users] Grandstream GXP-2000   Michiel van Baak
• [Asterisk-Users] sipgate.de question   Michiel van Baak
• [Asterisk-Users] sipgate.de question   Michiel van Baak
• [Asterisk-Users] Cannot boot machine up after working on zapt el....   Michiel van Baak
• [Asterisk-Users] NuFone   Wes Baehr
• [Asterisk-Users] re: voipjet   Wes Baehr
• [Asterisk-Users] changing cisco 7940/7960 standard menus ?   Chris Bagnall
• [Asterisk-Users] (newby) EURO-ISDN line question   Chris Bagnall
• [Asterisk-Users] Blocked Callerid   Chris Bagnall
• [Asterisk-Users] Skype-to-Asterisk(SIP): progress   Chris Bagnall
• [Asterisk-Users] ddi???   Chris Bagnall
• [Asterisk-Users] hardware and network requirements   Chris Bagnall
• [Asterisk-Users] Modifying dialplan for DUNDi compatibility   Chris Bagnall
• [Asterisk-Users] Sipura SPA 3000 logic   Chris Bagnall
• [Asterisk-Users] Sipura SPA 3000 logic   Chris Bagnall
• [Asterisk-Users] iLBC issue: An ilbc frame that isn't a multiple of 50 bytes long from RTP (38)   Chris Bagnall
• [Asterisk-Users] Solution for 1 time blast of 200, 000 recorded calls   Chris Bagnall
• [Asterisk-Users] Instant Messaging: with SIP or XMPP   Chris Bagnall
• [Asterisk-Users] BRI Newbie - What Hardware, PCI, in the US?   Chris Bagnall
• [Asterisk-Users] BRI Newbie - What Hardware, PCI, in the US?   Chris Bagnall
• [Asterisk-Users] Grandstream GXP-2000   Chris Bagnall
• [Asterisk-Users] Grandstream GXP-2000   Chris Bagnall
• [Asterisk-Users] queue behaviour   Chris Bagnall
• [Asterisk-Users] spa3000   Chris Bagnall
• [Asterisk-Users] g729 quality at GSM bitrates   Chris Bagnall
• [Asterisk-Users] What business IP phone to use   Chris Bagnall
• [Asterisk-Users] What business IP phone to use   Chris Bagnall
• [Asterisk-Users] how to add stun functionality in asterisk   Chris Bagnall
• [Asterisk-Users] good voip   Chris Bagnall
• [Asterisk-Users] What business IP phone to use   Chris Bagnall
• [Asterisk-Users] Hints between servers?   Chris Bagnall
• [Asterisk-Users] Hints between servers?   Chris Bagnall
• [Asterisk-Users] Incoming/Outgoing call question   Chris Bagnall
• [Asterisk-Users] What business IP phone to use   Chris Bagnall
• [Asterisk-Users] What business IP phone to use   Chris Bagnall
• [Asterisk-Users] Asterisk Topology   Chris Bagnall
• [Asterisk-Users] 5,000 concurrent calls system rollout question   Boris Bakchiev
• [Asterisk-Users] 5,000 concurrent calls system rollout question   Boris Bakchiev
• [Asterisk-Users] Queus seem to work different between IAX and ZAP channels   David F Bakker
• [Asterisk-Users] Delay after first digit - dial plan   Tharindu Rukshan Bamunuarachchi
• [Asterisk-Users] Please help - access out side number after waiting few seconds   Tharindu Rukshan Bamunuarachchi
• [Asterisk-Users] Delay after first digit - dial plan   Tharindu Rukshan Bamunuarachchi
• [Asterisk-Users] change languages from an IVR   Time Bandit
• [Asterisk-Users] GotoIf number exists in file. How can i do this?   Time Bandit
• [Asterisk-Users] GotoIf number exists in file. How can i do this?   Time Bandit
• [Asterisk-Users] Call centre - * hang's up   Time Bandit
• [Asterisk-Users] Increment Variable   Time Bandit
• [Asterisk-Users] Channel bank woes - no outbound calls   Time Bandit
• [Asterisk-Users] Monitor a call in progress.   Time Bandit
• [Asterisk-Users] Matching '*'   Time Bandit
• [Asterisk-Users] How to handle "provider UNREACHABLE" in the dialplan?   Umair Bari
• [Asterisk-Users] Directed Call Pickup   Alex Barnes
• [Asterisk-Users] Asterisk native sounds now available!   Alex Barnes
• [Asterisk-Users] Asterisk native sounds now available!   Alex Barnes
• [Asterisk-Users] Fedora Core 3 or Fedora Core 4? yum update ornot?   Alex Barnes
• [Asterisk-Users] attended call transfer   Alex Barnes
• [Asterisk-Users] attended call transfer   Alex Barnes
• [Asterisk-Users] attended call transfer   Alex Barnes
• [Asterisk-Users] MixMonitor and command   Alex Barnes
• [Asterisk-Users] MixMonitor and command   Alex Barnes
• [Asterisk-Users] Asterisk hints   Alex Barnes
• [Asterisk-Users] Fedora Core 3 or Fedora Core 4? yum update ornot?   Pete Barnwell
• [Asterisk-Users] consult about Digium Card   Pete Barnwell
• [Asterisk-Users] What business IP phone to use   Pete Barnwell
• [Asterisk-Users] Asterisk Televantage integration   Anish Basu
• [Asterisk-Users] Asterisk Televantage integration   Anish Basu
• [Asterisk-Users] maxmessages and maxgreet per mailbox   Anish Basu
• [Asterisk-Users] Welltech USA? and Wellgate Products?   Ariel Batista
• [Asterisk-Users] asterisk to FWD   Bayrouni
• [Asterisk-Users] test   Bayrouni
• [Asterisk-Users] asterisk to FWD   Bayrouni
• [Asterisk-Users] MP3player Problem   Bayrouni
• [Asterisk-Users] xlite and letters   Bayrouni
• [Asterisk-Users] how to tape letters in xlite   Bayrouni
• [Asterisk-Users] test   Bayrouni
• [Asterisk-Users] not consistent log from asterisk   Bayrouni
• [Asterisk-Users] not consistent log from asterisk   Bayrouni
• [Asterisk-Users] Cannot Dial out.   Forrest Beck
• [Asterisk-Users] Cannot Dial Out From *   Forrest Beck
• [Asterisk-Users] Asterisk and Hipath interconnections   Fons van der Beek
• [Asterisk-Users] Two Lines, Two Businesses   Les Bell
• [Asterisk-Users] Two Lines, Two Businesses   Les Bell
• [Asterisk-Users] Re: Anyone using the GSMgateway from CyberTelecom ?   Benchev
• [Asterisk-Users] need help   Benchev
• [Asterisk-Users] Re: Anyone using the GSMgateway from CyberTelecom ?   Benchev
• [Asterisk-Users] Limiting Sip Calls ?   Benchev
• [Asterisk-Users] Anyone know a good ITSP in Canada that supports *?   Dovid Bender
• [Asterisk-Users] Asterisk on laptop connected to POTS line   Dovid Bender
• [Asterisk-Users] Asterisk on laptop connected to POTS line   Dovid Bender
• [Asterisk-Users] Help on queues   Dovid Bender
• [Asterisk-Users] GSM Gateway / Terminal for sale   Dovid Bender
• [Asterisk-Users] New GSM 1-8 ports Gateway / Terminal for sale (with SMS Feature and Many more) £99 per unit   Dovid Bender
• [Asterisk-Users] FXS ATA and Pots wiring   Dovid Bender
• [Asterisk-Users] Asterisk with Billing   Dovid Bender
• [Asterisk-Users] Asterisk vs. Traditional PBX   Dovid Bender
• [Asterisk-Users] Asterisk vs. Traditional PBX   Dovid Bender
• [Asterisk-Users] asterisk + door opener   Dovid Bender
• [Asterisk-Users] Solution for 1 time blast of 200, 000 recorded calls   Dovid Bender
• [Asterisk-Users] Skilled API consultant required - preferably with Salesforce.com intergration   Dovid Bender
• [Asterisk-Users] Automated wake up call   Dovid Bender
• [Asterisk-Users] Aasterisk large-scale deployment w/analog phones   Dovid Bender
• [Asterisk-Users] Good VoIP providers that support Asterisk PBX's   Dovid Bender
• [Asterisk-Users] Automated wake up call   Dovid Bender
• [Asterisk-Users] is there a web interface to this mailing list?   Dovid Bender
• [Asterisk-Users] Automated wake up call   Dovid Bender
• [Asterisk-Users] Aasterisk large-scale deployment w/analog phones   Dovid Bender
• [Asterisk-Users] GSM GATEWAY   Dovid Bender
• [Asterisk-Users] Grandstream GXP-2000   Dovid Bender
• [Asterisk-Users] g.729 woes   Dovid Bender
• [Asterisk-Users] Dell PowerEdge 2850   Dovid Bender
• [Asterisk-Users] g.729 woes   Dovid Bender
• [Asterisk-Users] good voip   Dovid Bender
• [Asterisk-Users] Good VoIP providers that support Asterisk PBX's   Dovid Bender
• [Asterisk-Users] Multiple TDM400P's in a single machine   Dovid Bender
• [Asterisk-Users] Good VoIP providers that support Asterisk PBX's   Dovid Bender
• [Asterisk-Users] good voip   Dovid Bender
• [Asterisk-Users] Asterisk behind Centrex   Dovid Bender
• [Asterisk-Users] good voip   Dovid Bender
• [Asterisk-Users] Voice conferencing server capacity   Dovid Bender
• [Asterisk-Users] Configure DID   Dovid Bender
• [Asterisk-Users] Configure DID   Dovid Bender
• [Asterisk-Users] Problem with incoming call, Please help   Dovid Bender
• [Asterisk-Users] Cheapest provider for Philippine route   Dovid Bender
• [Asterisk-Users] cmd set with multiple values   Christian Benke
• [Asterisk-Users] RE: Euro-ISDN   Aldo Bergamini
• [Asterisk-Users] RE: Euro-ISDN   Aldo Bergamini
• [Asterisk-Users] Re: Euro-ISDN   Aldo Bergamini
• [Asterisk-Users] Re: Euro-ISDN   Aldo Bergamini
• [Asterisk-Users] Chan_BT question WAS: Asterisk with USB   Aldo Bergamini
• [Asterisk-Users] TDM - Analog Trunk - CallerID question   Aldo Bergamini
• [Asterisk-Users] Re: Asterisk-Users Digest, Vol 19, Issue 72   Aldo Bergamini
• [Asterisk-Users] Zap, Caller ID problem   Aldo Bergamini
• [Asterisk-Users] BT102 and ringtones   Aldo Bergamini
• [Asterisk-Users] Re: BT102 and ringtones   Aldo Bergamini
• [Asterisk-Users] Re: Anyone using the GSMgateway from CyberTelecom ?   Aldo Bergamini
• [Asterisk-Users] Re: Anyone using the GSMgateway from CyberTelecom ?   Aldo Bergamini
• [Asterisk-Users] Re: Anyone using the GSMgateway from CyberTelecom ?   Aldo Bergamini
• [Asterisk-Users] Anyone using the GSM gateway from CyberTelecom ?   Aldo Bergamini
• [Asterisk-Users] Anyone using the GSM gateway from CyberTelecom ?   Aldo Bergamini
• [Asterisk-Users] Re: courtesy message calling mobile phones   Aldo Bergamini
• [Asterisk-Users] re: voipjet   Andrew Berman
• [Asterisk-Users] Setting up Polycom 501 with 2 Different Extensions   Andrew Berman
• [Asterisk-Users] Setting up Polycom 501 with 2 Different Extensions   Andrew Berman
• [Asterisk-Users] Setting up Polycom 501 with 2 Different Extensions   Andrew Berman
• [Asterisk-Users] Setting up Polycom 501 with 2 Different Extensions   Andrew Berman
• [Asterisk-Users] (Un)PauseQeueMamber usage   Dov Bigio
• [Asterisk-Users] callerid issue   Dov Bigio
• [Asterisk-Users] limit sip sessions   Dov Bigio
• [Asterisk-Users] error cdr mysql addon   Dov Bigio
• [Asterisk-Users] IVR Menu   Dov Bigio
• [Asterisk-Users] asterisk logger - urgent!!!   Dov Bigio
• [Asterisk-Users] Re: asterisk logger - urgent!!!   Dov Bigio
• [Asterisk-Users] asterisk logger - urgent!!!   Dov Bigio
• [Asterisk-Users] asterisk logger - urgent!!!   Dov Bigio
• [Asterisk-Users] cdr (again) and deadlocks   Dov Bigio
• [Asterisk-Users] Developing a call centre app. Communicationwithasterisk?   Dov Bigio
• [Asterisk-Users] about g729 license   Dov Bigio
• [Asterisk-Users] about g729 license   Dov Bigio
• [Asterisk-Users] queue_log analysis   Dov Bigio
• [Asterisk-Users] asterisk error   Dov Bigio
• [Asterisk-Users] disallow, allow codes   Dov Bigio
• [Asterisk-Users] chanspy instability   Dov Bigio
• [Asterisk-Users] TE411P Really Bad Echo   Eric Bishop
• [Asterisk-Users] Any way to grep through fast moving console messages?   Eric Bishop
• [Asterisk-Users] TE411P Really Bad Echo   Eric Bishop
• [Asterisk-Users] How to create latency on purpose   Eric Bishop
• [Asterisk-Users] TE411P Really Bad Echo   Eric Bishop
• [Asterisk-Users] Callerid Name   John Bittner
• [Asterisk-Users] how to add stun functionality in asterisk   Jerry Glomph Black
• [Asterisk-Users] S100U and TigerJet   Jerry Glomph Black
• [Asterisk-Users] Cisco 2620 as PRI gateway   Steve Blair
• [Asterisk-Users] ParkAndAnnounce2 Feature Request   Steve Blair
• [Asterisk-Users] ParkAndAnnounce2 Feature Request   Steve Blair
• [Asterisk-Users] problem playing back voicemail   Joseph Blake
• [Asterisk-Users] Voicemail Problem   Joseph Blake
• [Asterisk-Users] Music on hold and conferencing on OS X   Joseph Blake
• [Asterisk-Users] Good VoIP providers that support Asterisk PBX's   David Blomquist
• [Asterisk-Users] ddi???   Phil Blundell
• [Asterisk-Users] attended call transfer   Phil Blundell
• [Asterisk-Users] Application Faxing using SIP   Phil Blundell
• [Asterisk-Users] HandyTone 488 ata?   Phil Blundell
• [Asterisk-Users] Application Faxing using SIP   Phil Blundell
• [Asterisk-Users] How come I don't have the MeetMe application registered?   Kevin Bockman
• [Asterisk-Users] How come I don't have the MeetMe applicationregistered?   Kevin Bockman
• [Asterisk-Users] [slackware 10.2 and TE205P] Unknown signalling method 'pri_cpe'   Kevin Bockman
• [Asterisk-Users] ImportVar Syntax   Kevin Bockman
• [Asterisk-Users] RE: 5, 000 concurrent calls system rollout question   William Boehlke
• [Asterisk-Users] RE: 5, 000 concurrent calls system rolloutquestion   William Boehlke
• [Asterisk-Users] RE: 5, 000 concurrent calls system rolloutquestion   William Boehlke
• [Asterisk-Users] Asterisk Web-Based Voicemail?   William Boehlke
• [Asterisk-Users] RE: 5,000 concurrent calls system rollout question   Greg Boehnlein
• [Asterisk-Users] Default value for ASTERISK_VERSION_NUM   Leo Ann Boon
• [Asterisk-Users] Default value for ASTERISK_VERSION_NUM   Leo Ann Boon
• [Asterisk-Users] Unistim Packet Decoder   Leo Ann Boon
• [Asterisk-Users] indications issues in Singapore?   Leo Ann Boon
• [Asterisk-Users] Asterisk as MGCP User Agent   Leo Ann Boon
• [Asterisk-Users] Asterisk behind Centrex   Leo Ann Boon
• [Asterisk-Users] user places two calls, hangs up, they get connected to one another   Leo Ann Boon
• [Asterisk-Users] Voicemail Changes   Nathan Bowyer
• [Asterisk-Users] Asterisk Follow Me   Nathan Bowyer
• [Asterisk-Users] Polycom IP 601 Buddy Watch problems   Nathan Bowyer
• [Asterisk-Users] CallerID popup   Peter Bowyer
• [Asterisk-Users] Sendmail with exchange   Peter Bowyer
• [Asterisk-Users] sipdiscount   Peter Bowyer
• [Asterisk-Users] Streaming Music On Hold   Peter Bowyer
• [Asterisk-Users] chan_capi-cm 0.6.4 random outgoing MSN problem   Peter Braidwood
• [Asterisk-Users] Send HookFlash after answering a ZAP (analog) channel   Hans-Juergen Brand
• [Asterisk-Users] Send HookFlash after answering a ZAP (analog) channel   Hans-Juergen Brand
• [Asterisk-Users] Re: asterisk t.38 pass   Adolfo R. Brandes
• [Asterisk-Users] Re: asterisk t.38 pass   Adolfo R. Brandes
• [Asterisk-Users] Outbound Call & SIP Results   David Brazier
• [Asterisk-Users] RE: Outbound Call & SIP Results   David Brazier
• [Asterisk-Users] Lockups since upgrade 1.2.3 - anyone else? Any ideas?   David Brazier
• [Asterisk-Users] Terminating AGI Scripts   Andy Brezinsky
• [Asterisk-Users] Asterisk with Cepstral for TTS   Brian
• [Asterisk-Users] Mitel 5220 IP phones   Bromont
• [Asterisk-Users] Mitel 5220 IP phones   Bromont
• [Asterisk-Users] DeadAGI variables confusion   Dave Brooks
• [Asterisk-Users] Re: DeadAGI variables confusion   Dave Brooks
• [Asterisk-Users] Asterisk - Vega 50 Disconnect Issues   Chad Brown
• [Asterisk-Users] Voicemail 0 for operator call routing   Bruce
• [Asterisk-Users] Issue w/ Polycom 501 phones in a queue...   Rolf Brusletto
• [Asterisk-Users] Covad anyone ...   Alan Bunch
• [Asterisk-Users] Problem with ARI and seeing voicemail...   Chuck Bunn
• [Asterisk-Users] ARI - Voicemail not showing - Problem solved!   Chuck Bunn
• [Asterisk-Users] Rights problem with Voicemail and non-root user - yeah I know, I thought I had it fixed...   Chuck Bunn
• [Asterisk-Users] Rights problem with Voicemail and non-root user - yeah I know, I thought I had it fixed...   Chuck Bunn
• [Asterisk-Users] Rights problem with Voicemail and non-root user - yeah I know, I thought I had it fixed...   Chuck Bunn
• [Asterisk-Users] Rights problem with Voicemail and non-root user - yeah I know, I thought I had it fixed...   Chuck Bunn
• [Asterisk-Users] Rights problem with Voicemail and non-root user - yeah I know, I thought I had it fixed...   Chuck Bunn
• [Asterisk-Users] Install instructions for FOP Flash Operator Panel do not make sense...   Chuck Bunn
• [Asterisk-Users] Features set in the features.conf stopped working after upgrade.   Chuck Bunn
• [Asterisk-Users] Keep getting message in logs that pbx.c cannot find extension context 'default'   Chuck Bunn
• [Asterisk-Users] Is setting the variable _TRANSFER_CONTEXT required in features.conf?   Chuck Bunn
• [Asterisk-Users] Okay can somebody explain this...   Chuck Bunn
• [Asterisk-Users] Is setting the variable _TRANSFER_CONTEXT required in features.conf?   Chuck Bunn
• [Asterisk-Users] Is setting the variable _TRANSFER_CONTEXT required in features.conf?   Chuck Bunn
• [Asterisk-Users] Asterisk on Solaris 10 (AMD Opteron, Sun Fire X2100)   Alexander Burke
• [Asterisk-Users] Asterisk on Solaris 10 (AMD Opteron, Sun Fire X2100)   Alexander Burke
• [Asterisk-Users] Fwd: Asterisk on Solaris 10 (AMD Opteron, Sun Fire X2100)   Alexander Burke
• [Asterisk-Users] Dell PowerEdge 2850   Alexander Burke
• [Asterisk-Users] Multiple TDM400P's in a single machine   Alexander Burke
• [Asterisk-Users] Download "Asterisk: The Future Of Telephony"   Alexander Burke
• [Asterisk-Users] Multiple TDM400P's in a single machine   Alexander Burke
• [Asterisk-Users] Download "Asterisk: The Future Of Telephony" [More Info]   Alexander Burke
• [Asterisk-Users] Fwd: Asterisk on Solaris 10 (AMD Opteron, Sun Fire X2100)   Alexander Burke
• [Asterisk-Users] Recommended rack-mountable server anyone?   Alexander Burke
• [Asterisk-Users] Download "Asterisk: The Future Of Telephony" [More Info]   Alexander Burke
• [Asterisk-Users] Newbie config help? Wellgate 3701a   Alexander Burke
• [Asterisk-Users] Asterisk Web-Based Voicemail?   Alexander Burke
• [Asterisk-Users] Voice Over WiFi   Alexander Burke
• [Asterisk-Users] Prepaid / postpaid solution   Alexander Burke
• [Asterisk-Users] courtesy message calling mobile phones   Alexander Burke
• [Asterisk-Users] asterisk logger - urgent!!!   Warren Burstein
• [Asterisk-Users] asterisk logger - urgent!!!   Warren Burstein
• [Asterisk-Users] user places two calls, hangs up, they get connected to one another   Warren Burstein
• [Asterisk-Users] Problem calling out   Warren Burstein
• [Asterisk-Users] user places two calls, hangs up, they get connected to one another   Warren Burstein
• [Asterisk-Users] incoming calls dropout on PRI over TE110p   Paul C
• [Asterisk-Users] incoming calls dropout on PRI over TE110p   Paul C
• [Asterisk-Users] What business IP phone to use   Ross C
• [Asterisk-Users] Nortel Meridian Opt 81C and PRI   Greg Camp
• [Asterisk-Users] Nortel Meridian Opt 81C and PRI   Greg Camp
• [Asterisk-Users] Visio-type symbol for an Asterisk/VoIP server?   Brian Capouch
• [Asterisk-Users] (newby) Is PING a good indicator of latency?   JP Carballo
• [Asterisk-Users] (newby) Is PING a good indicator of latency?   JP Carballo
• [Asterisk-Users] Billing inbound calls per minute   JP Carballo
• [Asterisk-Users] Fedora Core 3 or Fedora Core 4? yum update ornot?   JP Carballo
• [Asterisk-Users] Fedora Core 3 or Fedora Core 4? yum update ornot?   JP Carballo
• [Asterisk-Users] Billing inbound calls per minute   JP Carballo
• [Asterisk-Users] SIP Aliases   JP Carballo
• [Asterisk-Users] Help with dialplan   JP Carballo
• [Asterisk-Users] API or Call command   Carl
• [Asterisk-Users] Maximum retries exceeded on call/phantom calls?   Oscar Carriles
• [Asterisk-Users] SIP massive deregistration   Oscar Carriles
• [Asterisk-Users] SIP massive deregistration   Oscar Carriles
• [Asterisk-Users] Telmex PRI line configuration problem   Oscar Carriles
• [Asterisk-Users] Telmex PRI line configuration problem   Oscar Carriles
• [Asterisk-Users] Re: How many TDM2400P's will a server take?   Juan Carlos Castro y Castro
• [Asterisk-Users] Re: How many TDM2400P's will a server take?   Juan Carlos Castro y Castro
• [Asterisk-Users] tdm400p setup in china question   Cavanna, Richard
• [Asterisk-Users] re: voipjet -- Workaround if needed   Cavanna, Richard
• [Asterisk-Users] Half Solved - Fail over to Pri on VoIP connection failure   Cavanna, Richard
• [Asterisk-Users] Zaptel CiscoHDLC / Fedore FC4   Anthony Cennami
• [Asterisk-Users] "Proxy Authentication Required" issue   Tim Chase
• [Asterisk-Users] Polycom dialplan restriction   Carlos Chavez
• [Asterisk-Users] Problem win Unicall   Carlos Chavez
• [Asterisk-Users] Problem win Unicall   Carlos Chavez
• [Asterisk-Users] Aastra phones and common directory?   Carlos Chavez
• [Asterisk-Users] Problem Faxing   Carlos Chavez
• [Asterisk-Users] Problem win Unicall   Carlos Chavez
• [Asterisk-Users] Re: Configuring Meeting Room from Asterisk Manager API   Alexander Chemeris
• [Asterisk-Users] meetme application   Alexander Chemeris
• [Asterisk-Users] can asterisk to say chinese like say english   Jeffery Chen
• [Asterisk-Users] User web portal for Asterisk   Jeffery Chen
• [Asterisk-Users] agi/cagi call limit using group_count   Raymond Chen
• [Asterisk-Users] 4 TE411P in one server installation   Raymond Chen
• [Asterisk-Users] Asterisk hangs on 1.2.1   Sergio Chersovani
• [Asterisk-Users] changing cisco 7940/7960 standard menus ?   Sergio Chersovani
• [Asterisk-Users] chan_sccp availability?   Sergio Chersovani
• [Asterisk-Users] No audio? Update your Asterisk   Sergio Chersovani
• [Asterisk-Users] broken CDR (Master.csv) reports with HFC cards in Asterixk 1.2.x?   Tomasz Chmielewski
• [Asterisk-Users] Traffic prioritization and 'class of service' for SIP   David Choo
• [Asterisk-Users] call parking "hint"   Dr. Michael J. Chudobiak
• [Asterisk-Users] snom 360 problem - only one call works after reboot   Dr. Michael J. Chudobiak
• [Asterisk-Users] snom 360 problem - only one call works after reboot   Dr. Michael J. Chudobiak
• [Asterisk-Users] metermaid patch   Dr. Michael J. Chudobiak
• [Asterisk-Users] Asterisk, SIP phone , NAT   Dr. Michael J. Chudobiak
• [Asterisk-Users] IAX provider recommendation wanted   Dr. Michael J. Chudobiak
• [Asterisk-Users] jitterbuffer and DTMF conflict?   Dr. Michael J. Chudobiak
• [Asterisk-Users] jitterbuffer and DTMF conflict?   Dr. Michael J. Chudobiak
• [Asterisk-Users] IAX provider recommendation wanted   Dr. Michael J. Chudobiak
• [Asterisk-Users] isdn problem   Piotr Chytla
• [Asterisk-Users] Calls fading in and out   John Cianfarani
• [Asterisk-Users] Asterisk and Xen   John Cianfarani
• [Asterisk-Users] Asterisk and Xen   John Cianfarani
• [Asterisk-Users] Asterisk and Xen   John Cianfarani
• [Asterisk-Users] ATA186 V2.15.ms upgrade   Jorge Cisneros
• [Asterisk-Users] double ringing tone on asterisk 1.2 ((better) workaround)   Simone Cittadini
• [Asterisk-Users] can't hear 'service messages' when iax is in the middle   Simone Cittadini
• [Asterisk-Users] Cisco Gateway and Context Issues   Simone Cittadini
• [Asterisk-Users] lists problem, Gmail????????   Simone Cittadini
• [Asterisk-Users] Re: asterisk logger - urgent!!!   Simone Cittadini
• [Asterisk-Users] Obtaining billsecs in the dialplan after a call?   Simone Cittadini
• [Asterisk-Users] Asterisk 1.2.4 IAX2 New Jitterbuffer Tuning   Simone Cittadini
• [Asterisk-Users] mysql phone number pattern match query   Simone Cittadini
• [Asterisk-Users] can't dial some particular numbers (providers ?) with asterisk sip / zap channels   Simone Cittadini
• [Asterisk-Users] Asterisk Follow Me   Max Clark
• [Asterisk-Users] Asterisk Follow Me   Max Clark
• [Asterisk-Users] Fromuser required but overrides SetCallerID   Max Clark
• [Asterisk-Users] Asterisk vs. Traditional PBX   Mike Clark
• [Asterisk-Users] TE411P Really Bad Echo   Mike Clark
• [Asterisk-Users] Sangoma analog cards?   Mike Clark
• [Asterisk-Users] Fwd: Asterisk on Solaris 10 (AMD Opteron, Sun Fire X2100)   Mike Clark
• [Asterisk-Users] Which Quad Port FXO is Best?   Mike Clark
• [Asterisk-Users] Unable to Register to Asterisk through Proxy   Aaron Clauson
• [Asterisk-Users] supermicro server model   Eduard B. Cleofe
• [Asterisk-Users] TE411P or TE406P   Eduard B. Cleofe
• [Asterisk-Users] 64bit processor and 32 bit digium card   Eduard B. Cleofe
• [Asterisk-Users] Re: supermicro server model   Eduard B. Cleofe
• [Asterisk-Users] Asterisk and Hipath interconnections   richard Coco
• [Asterisk-Users] Asterisk and Hipath interconnections   richard Coco
• [Asterisk-Users] Re: ztdummy   Tzafrir Cohen
• [Asterisk-Users] Asterisk on laptop connected to POTS line   Tzafrir Cohen
• [Asterisk-Users] Asterisk on laptop connected to POTS line   Tzafrir Cohen
• [Asterisk-Users] can asterisk to say chinese like say english   Tzafrir Cohen
• [Asterisk-Users] Search for Links for "Communicating PC to PC in the same lan through Asterisk "   Tzafrir Cohen
• [Asterisk-Users] Search for Links for "Communicating PC to PC in the same lan through Asterisk "   Tzafrir Cohen
• [Asterisk-Users] Codec Selection   Tzafrir Cohen
• [Asterisk-Users] .version in zaptel   Tzafrir Cohen
• [Asterisk-Users] ztdummy on gentoo 2005.1   Tzafrir Cohen
• [Asterisk-Users] Fedora Core 3 or Fedora Core 4? yum update ornot?   Tzafrir Cohen
• [Asterisk-Users] asterisk logger - urgent!!!   Tzafrir Cohen
• [Asterisk-Users] ztdummy on gentoo 2005.1   Tzafrir Cohen
• [Asterisk-Users] ztdummy on gentoo 2005.1   Tzafrir Cohen
• [Asterisk-Users] asterisk logger - urgent!!!   Tzafrir Cohen
• [Asterisk-Users] ztdummy on gentoo 2005.1 [SOLVED]   Tzafrir Cohen
• [Asterisk-Users] Asterisk and Xen   Tzafrir Cohen
• [Asterisk-Users] Any way to grep through fast moving console messages?   Tzafrir Cohen
• [Asterisk-Users] Fedora Core 3 or Fedora Core 4? yum update ornot?   Tzafrir Cohen
• [Asterisk-Users] Fedora Core 3 or Fedora Core 4? yum update ornot?   Tzafrir Cohen
• [Asterisk-Users] Fedora Core 3 or Fedora Core 4? yum update ornot?   Tzafrir Cohen
• [Asterisk-Users] Unistim Packet Decoder   Tzafrir Cohen
• [kpj@junghanns.net: Re: [asterisk@frameweb.it: RE: [Asterisk-Users] Corrupt CDR records in Asterisk 1.2.x]]   Tzafrir Cohen
• [Asterisk-Users] Sendmail with exchange   Tzafrir Cohen
• [Asterisk-Users] Error running iaxcomm   Tzafrir Cohen
• [Asterisk-Users] Sendmail with exchange   Tzafrir Cohen
• [Asterisk-Users] Dialing part of the extension   Tzafrir Cohen
• [Asterisk-Users] Optimizing Linux to run Asterisk   Tzafrir Cohen
• [Asterisk-Users] Waiting for your help...   Tzafrir Cohen
• [Asterisk-Users] Rights problem with Voicemail and non-root user - yeah I know, I thought I had it fixed...   Tzafrir Cohen
• [Asterisk-Users] PIKA Technologies Inc. Announces Support for Open Source Asterisk PBX   Tzafrir Cohen
• [Asterisk-Users] ChanIsAvail   Tzafrir Cohen
• [Asterisk-Users] is there a web interface to this mailing list?   Tzafrir Cohen
• [Asterisk-Users] Asterisk and Snom 360   Tzafrir Cohen
• [Asterisk-Users] Podget or Similar   Tzafrir Cohen
• [Asterisk-Users] MOH from RCA jack?   Tzafrir Cohen
• [Asterisk-Users] Rights problem with Voicemail and non-root user - yeah I know, I thought I had it fixed...   Tzafrir Cohen
• [Asterisk-Users] Loops and Variables   Tzafrir Cohen
• [Asterisk-Users] Multiple TDM400P's in a single machine   Tzafrir Cohen
• [Asterisk-Users] Loops and Variables   Tzafrir Cohen
• [Asterisk-Users] sniffing sip password/uri/host info   Tzafrir Cohen
• [Asterisk-Users] S100U and TigerJet   Tzafrir Cohen
• [Asterisk-Users] Where can I get the tar.gz sources of libnewt?   Tzafrir Cohen
• [Asterisk-Users] asterisk -rx 'commands'   Tzafrir Cohen
• [Asterisk-Users] Cannot boot machine up after working on zaptel....   Tzafrir Cohen
• [Asterisk-Users] GR303   Jerimiah Cole
• [Asterisk-Users] GR303   Jerimiah Cole
• [Asterisk-Users] click to talk   Dean Collins
• [Asterisk-Users] Faint background noise/crackle on FXS porton TDM400P   Dean Collins
• [Asterisk-Users] Skilled API consultant required - preferably with Salesforce.com intergration   Dean Collins
• [Asterisk-Users] consult about Digium Card   Dean Collins
• [Asterisk-Users] Solution for 1 time blast of 200, 000 recorded calls   Dean Collins
• [Asterisk-Users] Skilled API consultant required - preferablywith Salesforce.com intergration   Dean Collins
• [Asterisk-Users] Podget or Similar   Dean Collins
• [Asterisk-Users] Automated wake up call   Dean Collins
• [Asterisk-Users] Wildfire messsaging server   Dean Collins
• [Asterisk-Users] Wildfire messsaging server   Dean Collins
• [Asterisk-Users] salesforce   Dean Collins
• [Asterisk-Users] OT- Rwanda DSL growth   Dean Collins
• [Asterisk-Users] DTMF Sporadicaly Being Generated   Kevin Collins
• [Asterisk-Users] DTMF Sporadicaly Being Generated   Kevin Collins
• [Asterisk-Users] SetCDRUserField not working in A@H?   Michael Collins
• RESOLUTION: [Asterisk-Users] SetCDRUserField not working in A@H?   Michael Collins
• [Asterisk-Users] POTS lines vs. using T1 to connectphoneservices?? HELP   Michael Collins
• [Asterisk-Users] OT O'Reilly Asterisk TFOT   Michael Collins
• [Asterisk-Users] Slightly OT: OpenPBX.org and Freeswitch   Michael Collins
• [Asterisk-Users] Slightly OT: OpenPBX.org and Freeswitch   Michael Collins
• [Asterisk-Users] varion card   Michael Collins
• [Asterisk-Users] Asterisk native sounds now available!   Michael Collins
• [Asterisk-Users] 911 and ISDN PRI   Michael Collins
• [Asterisk-Users] 911 and ISDN PRI   Michael Collins
• [Asterisk-Users] Need to retrieve Call-ID from dialed SIP channel(w/o CDRs)   Michael Collins
• [Asterisk-Users] more cpu intensive echo cancellers ?   Michael Collins
• [Asterisk-Users] 911 and ISDN PRI   Michael Collins
• [Asterisk-Users] Possible for Asterisk to output CLID to invoke 3rdparty app?   Michael Collins
• [Asterisk-Users] Possible for Asterisk to output CLID to invoke3rdparty app?   Michael Collins
• [Asterisk-Users] Asterisk vs. Traditional PBX   Michael Collins
• [Asterisk-Users] Dialing part of the extension   Michael Collins
• [Asterisk-Users] asterisk logger - urgent!!!   Michael Collins
• [Asterisk-Users] RE: Asterisk Logger - urgent!!!   Michael Collins
• [Asterisk-Users] configure TE205P on asterisk@home   Michael Collins
• [Asterisk-Users] attended call transfer   Michael Collins
• [Asterisk-Users] attended call transfer   Michael Collins
• [Asterisk-Users] problem with outgoing calls Unable to createchannel of type 'ZAP' (cause 34 - Circuit/channel congestion)   Michael Collins
• [Asterisk-Users] problem with outgoing calls Unable tocreatechannel of type 'ZAP' (cause 34 - Circuit/channel congestion)   Michael Collins
• [Asterisk-Users] automatically start application from the commandprompt   Michael Collins
• [Asterisk-Users] Send HookFlash after answering a ZAP(analog) channel   Michael Collins
• [Asterisk-Users] problem with outgoing calls Unabletocreatechannel of type 'ZAP' (cause 34 - Circuit/channel congestion)   Michael Collins
• [Asterisk-Users] attended call transfer   Michael Collins
• [Asterisk-Users] problem with outgoing callsUnabletocreatechannel of type 'ZAP' (cause 34 - Circuit/channelcongestion)   Michael Collins
• [Asterisk-Users] Increment Variable   Michael Collins
• [Asterisk-Users] "No D-channels available!"   Michael Collins
• [Asterisk-Users] "No D-channels available!"   Michael Collins
• [Asterisk-Users] Festival and Asterisk - different voices?   Michael Collins
• [Asterisk-Users] problem with outgoing callsUnabletocreatechannelof type 'ZAP' (cause 34 - Circuit/channelcongestion)   Michael Collins
• [Asterisk-Users] Festival and Asterisk - different voices? => SOLVED!   Michael Collins
• [Asterisk-Users] asterisk error   Michael Collins
• [Asterisk-Users] problem with outgoingcallsUnabletocreatechannelof type 'ZAP' (cause 34 -Circuit/channelcongestion)   Michael Collins
• [Asterisk-Users] Is Asterisk a PBX?   Michael Collins
• [Asterisk-Users] Problem with T1 installation   Michael Collins
• [Asterisk-Users] AGI Channel Status   Michael Collins
• [Asterisk-Users] AGI Scripts Terminate too Soon   Michael Collins
• [Asterisk-Users] AGI Channel Status   Michael Collins
• [Asterisk-Users] AGI Scripts Terminate too Soon   Michael Collins
• [Asterisk-Users] AGI Scripts Terminate too Soon   Michael Collins
• [Asterisk-Users] Re: Asterisk Question   Michael Collins
• [Asterisk-Users] Re: Asterisk Question   Michael Collins
• [Asterisk-Users] Re: Asterisk Question   Michael Collins
• [Asterisk-Users] Re: Asterisk Question   Michael Collins
• [Asterisk-Users] Polycom 501 netboot not working   William M Conlon
• [Asterisk-Users] Disabling SELinux in FC3 - good or bad   William M Conlon
• [Asterisk-Users] What business IP phone to use   The VoIP Connection
• [Asterisk-Users] What business IP phone to use   The VoIP Connection
• [Asterisk-Users] What business IP phone to use   The VoIP Connection
• [Asterisk-Users] Best ATA for general residential deployment??   The VoIP Connection
• [Asterisk-Users] Best ATA for general residential deployment??   The VoIP Connection
• [Asterisk-Users] Bandwidth: to seperate or not to seperate   Derek Conniffe
• [Asterisk-Users] Bandwidth: to seperate or not to seperate   Derek Conniffe
• [Asterisk-Users] Firefly & iaxLite dont stop ringing when answering incoming call   Derek Conniffe
• [Asterisk-Users] Asterisk 1.2.4 (behind NAT) IAX registration "Refresh 0" problem   Derek Conniffe
• [Asterisk-Users] RE: [Aterisk-Users] Zapbarge feature available?   Tim Connolly
• [Asterisk-Users] TE405p -- loopback for the phone company?   Tim Connolly
• [Asterisk-Users] TE405p -- loopback for the phone company?   Tim Connolly
• [Asterisk-Users] Re: Ring requested on channel already in use - fix   Tim Connolly
• [Asterisk-Users] MOH broke with 1.2.4 .. ?   Tim Connolly
• [Asterisk-Users] Zaptel 1.2.3 with Asterisk 1.0.9   Andre Courchesne - Consultant
• [Asterisk-Users] Re: Web interface   Sean Cook
• [Asterisk-Users] lists problem, Gmail????????   Sean Cook
• [Asterisk-Users] Digium TDM04B Outbound routing   Sean Cook
• [Asterisk-Users] ztdummy on gentoo 2005.1   Sean Cook
• [Asterisk-Users] Echo PSTN Asterisk@Home 2.0 Digium TDM11B & DSL   Sean Cook
• [Asterisk-Users] Sendmail with exchange   Sean Cook
• [Asterisk-Users] Rough Two Days   Sean Cook
• [Asterisk-Users] Rough Two Days   Sean Cook
• [Asterisk-Users] Polycom buddy watch limit of 7   Sean Cook
• [Asterisk-Users] [Announce] Web-MeetMe v2.0.0   Sean Cook
• [Asterisk-Users] [Announce] Web-MeetMe v2.0.0   Sean Cook
• [Asterisk-Users] Problem making outbound calls on TE210P using NFAS   Sean Cook
• [Asterisk-Users] [Announce] Web-MeetMe v2.0.0   Sean Cook
• [Asterisk-Users] [Announce] Web-MeetMe v2.0.0   Sean Cook
• [Asterisk-Users] [Announce] Web-MeetMe v2.0.0   Sean Cook
• [Asterisk-Users] Multiple TDM400P's in a single machine   Sean Cook
• [Asterisk-Users] Asterisk behind Centrex   Sean Cook
• [Asterisk-Users] auto provision of IP501 polycom   Sean Cook
• [Asterisk-Users] Pickup call on Hold   Sean Cook
• [Asterisk-Users] Asterisk Web-Based Voicemail?   Sean Cook
• [Asterisk-Users] Working SPA 841s now return 404 Not Found for INVITES and OPTION packets from *   Johnathan Corgan
• [Asterisk-Users] Working SPA 841s now return 404 Not Found for INVITES and OPTION packets from *   Johnathan Corgan
• [Asterisk-Users] is there a web interface to this mailing list?   Johnathan Corgan
• [Asterisk-Users] Problems with voicemail   Johnathan Corgan
• [Asterisk-Users] context being ignored by inbound sip call   Johnathan Corgan
• [Asterisk-Users] courtesy message calling mobile phones   Johnathan Corgan
• [Asterisk-Users] Cheapest provider for Philippine route   Johnathan Corgan
• [Asterisk-Users] Re: Asterisk Question   Johnathan Corgan
• [Asterisk-Users] Solution for 1 time blast of 200, 000 recorded calls   Peter Corlett
• [Asterisk-Users] OT O'Reilly Asterisk TFOT   Dave Cotton
• [Asterisk-Users] OT O'Reilly Asterisk TFOT   Dave Cotton
• [Asterisk-Users] Firmware version 1.3.1 released for Aastra IPphones   Dave Cotton
• [Asterisk-Users] Asterisk Web-Based Voicemail?   Dave Cotton
• [Asterisk-Users] Dundi key Problem   Jonathan k. Creasy
• [Asterisk-Users] winnipeg canada   Jonathan k. Creasy
• [Asterisk-Users] Asterisk on laptop connected to POTS line   Jonathan k. Creasy
• [Asterisk-Users] CallerID popup   Jonathan k. Creasy
• [Asterisk-Users] How can I configure to call from the consolebymeans of a sip phone,   Jonathan k. Creasy
• [Asterisk-Users] Asterisk on laptop connected to POTS line   Jonathan k. Creasy
• SOLVED: Re: [Asterisk-Users] Polycom IP501 with Asterisk -distinctive ring?   Jonathan k. Creasy
• SOLVED: Re: [Asterisk-Users] Polycom IP501 with Asterisk -distinctive ring?   Jonathan k. Creasy
• [Asterisk-Users] Fax to Email with Asterisk and Lucent TNT   Matthew Crocker
• [Asterisk-Users] Fax to Email with Asterisk and Lucent TNT   Matthew Crocker
• [Asterisk-Users] OT: VoIP over bonded link   Matthew Crocker
• [Asterisk-Users] Any Digium Supplier/reseller accepts Paypal ?   Michael Crown
• [Asterisk-Users] Test   CyberSource
• [Asterisk-Users] good voip   CyberSource
• [Asterisk-Users] good voip   CyberSource
• [Asterisk-Users] good voip   CyberSource
• [Asterisk-Users] MWI on Polycom 501.   Ken D'Ambrosio
• [Asterisk-Users] MWI on Polycom 501.   Ken D'Ambrosio
• [Asterisk-Users] BAD/GOOD Echo Cancel   Ken D'Ambrosio
• [Asterisk-Users] BAD/GOOD Echo Cancel   Ken D'Ambrosio
• [Asterisk-Users] "No D-channels available!"   Ken D'Ambrosio
• [Asterisk-Users] Sangoma analog cards?   Ken D'Ambrosio
• [Asterisk-Users] "No D-channels available!"   Ken D'Ambrosio
• [Asterisk-Users] Choice One PRI?   Ken D'Ambrosio
• [Asterisk-Users] Server & Wildcard TE110P [ Virusgeprüft]   DRi at b-w-computer.de
• [Asterisk-Users] res_features pickupexten   DRi at b-w-computer.de
• [Asterisk-Users] Re: res_features pickupexten   DRi at b-w-computer.de
• [Asterisk-Users] Queue() with timeout=0   Bart van Daal
• [Asterisk-Users] Queue() with timeout=0   Bart van Daal
• [Asterisk-Users] Queue() with timeout=0   Bart van Daal
• [Asterisk-Users] SPA-941/2 Monitoring   Josh Dady
• [Asterisk-Users] Matching variables in extensions.conf   Adam James Dale
• [Asterisk-Users] Linksys WIP300 WiFi Phone   Dan
• [Asterisk-Users] Cisco 79XX firmware 7.5   Aaron Daniel
• [Asterisk-Users] ChanIsAvail   Aaron Daniel
• [Asterisk-Users] Cisco 79xx and SIP 7.5 Problems   Aaron Daniel
• [Asterisk-Users] mpg123 alternative?   Aaron Daniel
• [Asterisk-Users] 7940 and Subscriptions   Aaron Daniel
• [Asterisk-Users] fax receive using TDM400P   Rob Danz
• [Asterisk-Users] Web based SIP client   Klaus Darilion
• [Asterisk-Users] Dell PowerEdge 2850   Klaus Darilion
• [Asterisk-Users] ISDN interface cards with pass-through   Klaus Darilion
• [Asterisk-Users] MFC/R2 in Brazil   Darlon
• [Asterisk-Users] Problem win Unicall   Darlon
• [Asterisk-Users] 2wav2mp3, monitor, mixmonitor, mpg123, queues   Darlon
• [Asterisk-Users] Re: Echo on PRI/BRI?   Paul Davidson
• [Asterisk-Users] Re: Echo and other reasons to migrate to BRI   Paul Davidson
• [Asterisk-Users] Conference bridge dimensioning   Paul Davidson
• [Asterisk-Users] international caller id on UK (BT) PRI   Steve Davies
• [Asterisk-Users] TAPI Recommendations   Steve Davies
• [Asterisk-Users] TAPI Recommendations   Steve Davies
• [Asterisk-Users] Problem with Incoming Caller ID on Nufone Since Upgrade   Robert DeVries
• [Asterisk-Users] Problem with Incoming Caller ID on Nufone Since Upgrade   Robert DeVries
• [Asterisk-Users] Re: Problem with Incoming Caller ID on Nufone Since Upgrade   Robert DeVries
• [Asterisk-Users] Teliax - Codec Preference effective?   Rusty Dekema
• [Asterisk-Users] hardware suggestion   Rusty Dekema
• [Asterisk-Users] g.729 woes   Rusty Dekema
• [Asterisk-Users] Best ATA for general residential deployment??   Rusty Dekema
• [Asterisk-Users] Best ATA for general residential deployment??   Rusty Dekema
• [Asterisk-Users] GPS-enabled cell phone/PDA   Rusty Dekema
• [Asterisk-Users] Asterisk with T1 card on laptop   Rusty Dekema
• [Asterisk-Users] Performance differences 64-bit vs 32-bit   Austin Denyer
• [Asterisk-Users] TE210P mother board   Phone Dev
• R: [Asterisk-Users] php agi configuration issue   Phone Dev
• [Asterisk-Users] how to add stun functionality in asterisk   Deepak Dhiman
• [Asterisk-Users] NEED COMMENT ON USING FEDORA CORE 3   Saul Diaz
• [Asterisk-Users] What business IP phone to use   Saul Diaz
• [Asterisk-Users] Test my test-branch!   Saul Diaz
• [Asterisk-Users] Live Communication Server and Asterisk   Dinesh
• [Asterisk-Users] sniffing sip password/uri/host info   Dinesh
• [Asterisk-Users] CallerID popup   Iuri Gomes Diniz
• [Asterisk-Users] Re: T.38 patch instruactions   Ben Dinnerville
• [Asterisk-Users] one way / irratic voice over iax and g729   Ben Dinnerville
• [Asterisk-Users] Re: one way / irratic voice over iax and g729   Ben Dinnerville
• [Asterisk-Users] Re: one way / irratic voice over iax and g729   Ben Dinnerville
• [Asterisk-Users] Re: Multiple TDM400P's in a single machine   Ben Dinnerville
• [Asterisk-Users] realtime, iax, trunk   Ben Dinnerville
• [Asterisk-Users] Adjusting frequency asterisk sends NOTIFY's to ATA's at for MWI.   Ray Van Dolson
• [Asterisk-Users] spandsp 0.0.2pre25   Paradise Dove
• [Asterisk-Users] Not passing CALLER id on in follow me script   Paul Dracevich
• [Asterisk-Users] Grandstream Budgetone BT-101 audio problems   Cristian Draghici
• [Asterisk-Users] Re: Re: asterisk logger - urgent!!!   Cristian Draghici
• [Asterisk-Users] Developing a call centre app. Communication with asterisk?   Evan Duffield
• [Asterisk-Users] Problems with certain Global Variables not passing correctly.   Corporate IT Solutions - Michael Dunne
• [Asterisk-Users] "Say YES to continue" prompts   Philip Edelbrock
• [Asterisk-Users] Zap Auto disconnect after xx seconds of silence   Philip Edelbrock
• [Asterisk-Users] Traffic prioritization and 'class of service' for SIP   Philip Edelbrock
• [Asterisk-Users] Traffic prioritization and 'class of service' for SIP   Philip Edelbrock
• [Asterisk-Users] Traffic prioritization and 'class of service' for SIP   Philip Edelbrock
• [Asterisk-Users] Traffic prioritization and 'class of service' for SIP   Philip Edelbrock
• [Asterisk-Users] Festival and Asterisk - different voices?   Philip Edelbrock
• [Asterisk-Users] Grandstream GXP-2000   Philip Edelbrock
• [Asterisk-Users] Application Faxing using SIP   Philip Edelbrock
• [Asterisk-Users] Multiple TDM400P's in a single machine   Philip Edelbrock
• [Asterisk-Users] sniffing sip password/uri/host info   Philip Edelbrock
• [Asterisk-Users] Linksys WIP300 WiFi Phone   Philip Edelbrock
• [Asterisk-Users] Linksys WIP300 WiFi Phone   Philip Edelbrock
• [Asterisk-Users] Linksys WIP300 WiFi Phone   Philip Edelbrock
• [Asterisk-Users] Connecting to live calls   Mark Edwards
• [Asterisk-Users] PRI Group behavior - CHANUNAVAIL   Mark Edwards
• [Asterisk-Users] PRI indications.   Mark Edwards
• [Asterisk-Users] channel.c: Avoided deadlock for '0x91a8b20', 10 retries!   Mark Edwards
• [Asterisk-Users] SIP Register   Mark Edwards
• [Asterisk-Users] Asterisk on Solaris 10 (AMD Opteron, Sun Fire X2100)   Mark Edwards
• [Asterisk-Users] Asterisk on Solaris 10 (AMD Opteron, SunFire X2100)   Mark Edwards
• [Asterisk-Users] ISDN30E + T1 crossover cable woes   Mark Edwards
• [Asterisk-Users] Cannot boot machine up after working on zaptel....   Christoph Eicke
• [Asterisk-Users] Help with Grandstream Handytone 386 together with Asterisk and a connected modem   Kib Eki
• [Asterisk-Users] ATA's and faxing   Kib Eki
• [Asterisk-Users] Expression GotoIf - bug or personal misunderstanding?   Kib Eki
• [Asterisk-Users] FW: Web Interface   Dan Elder
• [Asterisk-Users] two tellabs 2572 echo board in a 253c mounting assembly?   Dan Elder
• [Asterisk-Users] Re: two tellabs 2572 echo board in a 253c mounting assembly?   Dan Elder
• [Asterisk-Users] Re: two tellabs 2572 echo board in a 253c mounting   Dan Elder
• [Asterisk-Users] ZAP extension, DTMF?   Dan Elder
• [Asterisk-Users] RE: ZAP extension, DTMF?   Dan Elder
• [Asterisk-Users] can't dial zap extensions?   Dan Elder
• [Asterisk-Users] asterisk silence suppression?   Dan Elder
• [Asterisk-Users] RE: ZAP extension, DTMF?   Dan Elder
• [Asterisk-Users] SPA-941 Selective DND   Darren Ellis
• [Asterisk-Users] Asterisk with HT 488 FXO   Pasqualotto Enrico
• [Asterisk-Users] Asterisk with HT 488 FXO   Pasqualotto Enrico
• [Asterisk-Users] Asterisk with HT 488 FXO   Pasqualotto Enrico
• [Asterisk-Users] calling to sip provider   Carles Pina i Estany
• [Asterisk-Users] 5,000 concurrent calls system rollout question   Damon Estep
• [Asterisk-Users] 5,000 concurrent calls system rollout question   Damon Estep
• [Asterisk-Users] 5,000 concurrent calls system rollout question   Damon Estep
• [Asterisk-Users] determining if a call to a SIP extensions is from a queue   Damon Estep
• [Asterisk-Users] query about Three way calling   Damon Estep
• [Asterisk-Users] (newby) Is PING a good indicator of latency?   Damon Estep
• [Asterisk-Users] (newby) Is PING a good indicator of latency?   Damon Estep
• [Asterisk-Users] determining if a call to a SIP extensions isfrom a queue   Damon Estep
• [Asterisk-Users] Asterisk on laptop connected to POTS line   Damon Estep
• [Asterisk-Users] Asterisk on laptop connected to POTS line   Damon Estep
• [Asterisk-Users] orphaned sip channels channels?   Damon Estep
• [Asterisk-Users] orphaned sip channels channels?   Damon Estep
• [Asterisk-Users] mysql phone number pattern match query   Damon Estep
• [Asterisk-Users] auto provision of IP501 polycom   Damon Estep
• [Asterisk-Users] auto provision of IP501 polycom   Damon Estep
• [Asterisk-Users] RE: auto provision of IP501 polycom   Damon Estep
• [Asterisk-Users] nat=yes and qualify=yes viable NAT solutions?   Damon Estep
• [Asterisk-Users] nat=yes and qualify=yes viable NAT solutions?   Damon Estep
• [Asterisk-Users] another nat question   Damon Estep
• [Asterisk-Users] changing source email address of pager notifications   Damon Estep
• [Asterisk-Users] A room full of Cisco 7960s behind NAT   Damon Estep
• [Asterisk-Users] Asterisk with T1 card on laptop   Damon Estep
• [Asterisk-Users] A room full of Cisco 7960s behind NAT   Damon Estep
• [Asterisk-Users] A room full of Cisco 7960s behind NAT   Damon Estep
• [Asterisk-Users] GSM GATEWAY   Dumpolid Exeplish
• [Asterisk-Users] GSM GATEWAY   Dumpolid Exeplish
• [Asterisk-Users] GSM GATEWAY   Dumpolid Exeplish
• [Asterisk-Users] Voice Over WiFi   Dumpolid Exeplish
• [Asterisk-Users] Analog with channel bank - Inbound works, outbound doesn't   C F
• [Asterisk-Users] Analog with channel bank - Inbound works, outbound doesn't   C F
• [Asterisk-Users] Analog with channel bank - Inbound works, outbound doesn't   C F
• [Asterisk-Users] CallerID popup   C F
• [Asterisk-Users] Pattern Match - 0 or 1 digit   C F
• [Asterisk-Users] Not receving anything from the list   C F
• [Asterisk-Users] Re: two tellabs 2572 echo board in a 253c mounting   C F
• [Asterisk-Users] BAD/GOOD Echo Cancel   C F
• [Asterisk-Users] two tellabs 2572 echo board in a 253c mounting assembly?   C F
• [Asterisk-Users] One way audio - it doesn't make sense   C F
• [Asterisk-Users] lists problem, Gmail????????   C F
• [Asterisk-Users] Possible for Asterisk to output CLID to invoke 3rd party app?   C F
• [Asterisk-Users] TE411P Really Bad Echo   C F
• [Asterisk-Users] TE411P Really Bad Echo   C F
• [Asterisk-Users] TE411P Really Bad Echo   C F
• [Asterisk-Users] TE411P Really Bad Echo   C F
• [Asterisk-Users] Best quad-port fxo solution with EC?   C F
• [Asterisk-Users] Lucent Avaya Partner ACS T1 module   C F
• [Asterisk-Users] Install instructions for FOP Flash Operator Panel do not make sense...   C F
• [Asterisk-Users] Safely editing voicemail.conf   C F
• [Asterisk-Users] SPA-941 stutter tone   C F
• [Asterisk-Users] [OT] List messages and end user outages   C F
• [Asterisk-Users] Loops and Variables   C F
• [Asterisk-Users] Send flash through zap channel   C F
• [Asterisk-Users] Download "Asterisk: The Future Of Telephony" [More Info]   C F
• [Asterisk-Users] Loops and Variables   C F
• [Asterisk-Users] Matching variables in extensions.conf   C F
• [Asterisk-Users] Asterisk Follow Me   C F
• [Asterisk-Users] Asterisk Follow Me   C F
• [Asterisk-Users] Asterisk Follow Me   C F
• [Asterisk-Users] Polycom IP601 Question   C F
• [Asterisk-Users] Cisco 79xx and SIP 7.5 Problems   C F
• [Asterisk-Users] Voicemail problems   C F
• [Asterisk-Users] Cisco 79xx and SIP 7.5 Problems   C F
• [Asterisk-Users] Missing 31 DTMF tones over ZAP   C F
• [Asterisk-Users] Internal Server Error   C F
• [Asterisk-Users] nat=yes and qualify=yes viable NAT solutions?   C F
• [Asterisk-Users] Internal Server Error   C F
• [Asterisk-Users] nat=yes and qualify=yes viable NAT solutions?   C F
• [Asterisk-Users] courtesy message calling mobile phones   C F
• [Asterisk-Users] SPA-941 Selective DND   C F
• [Asterisk-Users] Asterisk with T1 card on laptop   C F
• [Asterisk-Users] codecs choice   FaberK
• [Asterisk-Users] Dial and Congestion   FaberK
• [Asterisk-Users] Dial and Congestion   FaberK
• [Asterisk-Users] Comfort noise support incomplete in Asterisk (RFC 3389)   FaberK
• [Asterisk-Users] OT O'Reilly Asterisk TFOT   Fabrice
• [Asterisk-Users] PRI to PRI not passing callerid   Mark Farver
• [Asterisk-Users] PRI to PRI not passing callerid   Mark Farver
• [Asterisk-Users] T1 Channel splitting PRI/data not working   Mark Farver
• [Asterisk-Users] 911 and ISDN PRI   Jonathan Feally
• [Asterisk-Users] TDM2400P FXS Only vs. T1/E1 to FXS Channel Banks   Jonathan Feally
• [Asterisk-Users] TDM2400P FXS Only vs. T1/E1 to FXS Channel Banks   Jonathan Feally
• [Asterisk-Users] No Voice when canreinvite=no   Jonathan Feally
• [Asterisk-Users] Networking voicemail   Peter Fern
• [Asterisk-Users] determining if a call to a SIP extensions is from a queue   Peter Fern
• [Asterisk-Users] changing cisco 7940/7960 standard menus ?   Peter Fern
• [Asterisk-Users] (newby) Is PING a good indicator of latency?   Peter Fern
• [Asterisk-Users] MOH sourced from a sound card?   Peter Fern
• [Asterisk-Users] Digit timeouts vs includes in diaplan   Peter Fern
• [Asterisk-Users] Re: Asterisk hangs on 1.2.1   Peter Fern
• [Asterisk-Users] dummy Technology/resource for Dial   Peter Fern
• [Asterisk-Users] sip channel status - how?   Peter Fern
• [Asterisk-Users] Dial command to connect two channels and bypass asterisk server   Peter Fern
• [Asterisk-Users] 79xx's and call queues   Peter Fern
• [Asterisk-Users] Can I escape queue with a '*'?   Peter Fern
• [Asterisk-Users] Queue Messages not playing when caller is inside queue   Peter Fern
• [Asterisk-Users] Asterisk 1.2.4 IAX2 New Jitterbuffer Tuning   Peter Fern
• [Asterisk-Users] Matching '*'   Peter Fern
• [Asterisk-Users] TEST   Krystian Filiks
• SV: [Asterisk-Users] callback script?   Krystian Filiks
• [Asterisk-Users] BAD/GOOD Echo Cancel   Krystian Filiks
• [Asterisk-Users] BAD/GOOD Echo Cancel   Krystian Filiks
• [Asterisk-Users] echo cancel from telco   Krystian Filiks
• [Asterisk-Users] No sound on 10% of incoming calls   Krystian Filiks
• [Asterisk-Users] Problem with ZAPHFC: internal S0 hangs when hanging up   Sven Fischer
• [Asterisk-Users] Problem with ZAPHFC: internal S0 hangs when hanging up   Sven Fischer
• [Asterisk-Users] Problem with ZAPHFC: internal S0 hangs when hanging up   Sven Fischer
• [Asterisk-Users] Leading 0 on caller ID with internal S0 (HFC)   Sven Fischer
• [Asterisk-Users] Question on SIP authentication with users from OpenSER   Barry Flanagan
• [Asterisk-Users] Re: How do I install speex for asterisk?   Barry Flanagan
• [Asterisk-Users] Re: Fromstring when sending e-mail on recieved voicemail   Barry Flanagan
• [Asterisk-Users] Re: SV: Re: Fromstring when sending e-mail on recievedvoicemail   Barry Flanagan
• [Asterisk-Users] Re: SV: Re: Fromstring when sending e-mail on recievedvoicemail   Barry Flanagan
• [Asterisk-Users] Re: SV: Re: Fromstring when sending e-mail on recievedvoicemail   Barry Flanagan
• [Asterisk-Users] Re: SV: Re: SV: Re: Fromstring when sending e-mail onrecievedvoicemail   Barry Flanagan
• [Asterisk-Users] Re: SV: Re: SV: Re: SV: Re: Fromstring when sending e-mailonrecievedvoicemail   Barry Flanagan
• [Asterisk-Users] Re: context being ignored by inbound sip call   Barry Flanagan
• [Asterisk-Users] Re: a2billing without IVR   Barry Flanagan
• [Asterisk-Users] Default value for ASTERISK_VERSION_NUM   Kevin P. Fleming
• [Asterisk-Users] Re: How many TDM2400P's will a server take?   Kevin P. Fleming
• [Asterisk-Users] Default value for ASTERISK_VERSION_NUM   Kevin P. Fleming
• [Asterisk-Users] meetme and dtmf   Kevin P. Fleming
• [Asterisk-Users] DTMF Sporadicaly Being Generated   Kevin P. Fleming
• [Asterisk-Users] TE411P or TE406P   Kevin P. Fleming
• [Asterisk-Users] meetme and dtmf   Kevin P. Fleming
• [Asterisk-Users] Call completes but then drops?   Kevin P. Fleming
• [Asterisk-Users] Call completes but then drops?   Kevin P. Fleming
• [Asterisk-Users] meetme and dtmf   Kevin P. Fleming
• [Asterisk-Users] Default value for ASTERISK_VERSION_NUM   Kevin P. Fleming
• [Asterisk-Users] Slightly OT: OpenPBX.org and Freeswitch   Kevin P. Fleming
• [Asterisk-Users] Re: Web interface   Kevin P. Fleming
• [Asterisk-Users] meetme and dtmf   Kevin P. Fleming
• [Asterisk-Users] TE411P Really Bad Echo   Kevin P. Fleming
• [Asterisk-Users] TE210P mother board   Kevin P. Fleming
• [Asterisk-Users] .version in zaptel   Kevin P. Fleming
• [Asterisk-Users] queues   Kevin P. Fleming
• Re: [Asterisk-Users] New GSM 1-8 ports Gateway / Terminal for sale (with SMS Feature and Many more) £99 per unit   Kevin P. Fleming
• [Asterisk-Users] BAD/GOOD Echo Cancel   Kevin P. Fleming
• [Asterisk-Users] TE405p -- loopback for the phone company?   Kevin P. Fleming
• [Asterisk-Users] TE405p -- loopback for the phone company?   Kevin P. Fleming
• [Asterisk-Users] Asterisk native sounds now available!   Kevin P. Fleming
• [Asterisk-Users] BAD/GOOD Echo Cancel   Kevin P. Fleming
• [Asterisk-Users] DTMF Sporadicaly Being Generated   Kevin P. Fleming
• [Asterisk-Users] Need to retrieve Call-ID from dialed SIP channel (w/o CDRs)   Kevin P. Fleming
• [Asterisk-Users] Re: Need to retrieve Call-ID from dialed SIP channel (w/o CDRs)   Kevin P. Fleming
• [Asterisk-Users] Re: Need to retrieve Call-ID from dialed number   Kevin P. Fleming
• [Asterisk-Users] TDM400p   Kevin P. Fleming
• [Asterisk-Users] asterisk logger - urgent!!!   Kevin P. Fleming
• [Asterisk-Users] Dell PowerEdge 1800 and TE410P   Kevin P. Fleming
• [Asterisk-Users] Meetme echo cancellation   Kevin P. Fleming
• [Asterisk-Users] Polycom IP501 with Asterisk - distinctive ring?   Kevin P. Fleming
• [Asterisk-Users] asterisk logger - urgent!!!   Kevin P. Fleming
• [Asterisk-Users] Re: How many TDM2400P's will a server take?   Kevin P. Fleming
• [Asterisk-Users] asterisk logger - urgent!!!   Kevin P. Fleming
• [Asterisk-Users] asterisk logger - urgent!!!   Kevin P. Fleming
• [Asterisk-Users] asterisk logger - urgent!!!   Kevin P. Fleming
• [Asterisk-Users] asterisk logger - urgent!!!   Kevin P. Fleming
• [Asterisk-Users] dtmfmode=auto, but doesn't work   Kevin P. Fleming
• [Asterisk-Users] dtmfmode=auto, but doesn't work   Kevin P. Fleming
• [Asterisk-Users] about g729 license   Kevin P. Fleming
• [Asterisk-Users] Can Asterisk send RTP to a specific port number?   Kevin P. Fleming
• [Asterisk-Users] Dial command to connect two channels and bypassasterisk server   Kevin P. Fleming
• [Asterisk-Users] Nat, SIP, Realtime problem   Kevin P. Fleming
• [Asterisk-Users] Dial command to connect two channels and bypassasterisk server   Kevin P. Fleming
• [Asterisk-Users] Polycom buddy watch limit of 7   Kevin P. Fleming
• [Asterisk-Users] Asterisk large-scale deployment w/analog phones   Kevin P. Fleming
• [Asterisk-Users] Channel bleedover?   Kevin P. Fleming
• [Asterisk-Users] "Proxy Authentication Required" issue   Kevin P. Fleming
• [Asterisk-Users] Hints between servers?   Kevin P. Fleming
• [Asterisk-Users] Codec order sent wrong from Asterisk   Kevin P. Fleming
• [Asterisk-Users] mpg123 alternative?   Kevin P. Fleming
• [Asterisk-Users] OT: VoIP over bonded link   Kevin P. Fleming
• [Asterisk-Users] Re: Voicemail problems   Kevin P. Fleming
• [Asterisk-Users] chanspy instability   Kevin P. Fleming
• [Asterisk-Users] How to determine the power draw on TDM2400P?   Kevin P. Fleming
• [Asterisk-Users] Zhone channel Banks   Matt Florell
• [Asterisk-Users] asterisk logger - urgent!!!   Matt Florell
• [Asterisk-Users] Re: Help on Vicidial   Matt Florell
• [Asterisk-Users] Solution for 1 time blast of 200, 000 recorded calls   Matt Florell
• [Asterisk-Users] Aasterisk large-scale deployment w/analog phones   Matt Florell
• [Asterisk-Users] Developing a call centre app. Communication with asterisk?   Matt Florell
• [Asterisk-Users] Aasterisk large-scale deployment w/analog phones   Matt Florell
• [Asterisk-Users] Aasterisk large-scale deployment w/analog phones   Matt Florell
• [Asterisk-Users] Asterisk Contact Center   Matt Florell
• [Asterisk-Users] TDM 400P in Malaysia   Chee Foong
• [Asterisk-Users] segmentation fault   Patrick Fortin
• [Asterisk-Users] arris e-mta   Patrick Fortin
• [Asterisk-Users] commercial package for vertical services   Patrick Fortin
• [Asterisk-Users] Replicating functionality from our prior PBX   Patrick W. Foster
• [Asterisk-Users] Will not authenticate incoming VOIP provider calls   Francois
• [Asterisk-Users] TE411P Really Bad Echo   Matthew Fredrickson
• [Asterisk-Users] TE405p -- loopback for the phone company?   Matthew Fredrickson
• [Asterisk-Users] more cpu intensive echo cancellers ?   Matthew Fredrickson
• [Asterisk-Users] more cpu intensive echo cancellers ?   Matthew Fredrickson
• [Asterisk-Users] more cpu intensive echo cancellers ?   Matthew Fredrickson
• [Asterisk-Users] more cpu intensive echo cancellers ?   Matthew Fredrickson
• [Asterisk-Users] TE411P Really Bad Echo   Matthew Fredrickson
• [Asterisk-Users] Softphone --> Bluetooth Smartphone   Arik Funke
• [Asterisk-Users] Any way to grep through fast moving console messages?   Andrew Furey
• [Asterisk-Users] chan_capi setting ${DNIS}   Andrew Furey
• [Asterisk-Users] 5,000 concurrent calls system rollout question   Doug G
• [Asterisk-Users] Looking for programer...   Doug G
• [Asterisk-Users] Important: Application DIALPLAN STANDARD/GUIDELINES needs to be established.   James Gardiner
• [Asterisk-Users] Sipura SPA-2002 rings randomly   Kerry Garrison
• [Asterisk-Users] Asterisk on laptop connected to POTS line   Kerry Garrison
• [Asterisk-Users] virtual extension per user ?   Kerry Garrison
• [Asterisk-Users] Handset phone to replace Flash Operator Panel   Kerry Garrison
• [Asterisk-Users] Asterisk vs. Traditional PBX   Kerry Garrison
• [Asterisk-Users] AAH 2.5 pone paging broken   Kerry Garrison
• [Asterisk-Users] Solution for 1 time blast of 200, 000 recordedcalls   Kerry Garrison
• [Asterisk-Users] SPA-941 stutter tone   Kerry Garrison
• [Asterisk-Users] Multiple TDM400P's in a single machine   Kerry Garrison
• [Asterisk-Users] Asterisk native sounds now available!   Douglas Garstang
• [Asterisk-Users] Asterisk native sounds now available!   Douglas Garstang
• [Asterisk-Users] Asterisk native sounds now available!   Douglas Garstang
• [Asterisk-Users] AGI Scripts Staying in Memory   Douglas Garstang
• [Asterisk-Users] Terminating AGI Scripts   Douglas Garstang
• [Asterisk-Users] Terminating AGI Scripts   Douglas Garstang
• [Asterisk-Users] Multiple AGI Issues   Douglas Garstang
• [Asterisk-Users] Multiple AGI Issues   Douglas Garstang
• [Asterisk-Users] Increment Variable   Douglas Garstang
• [Asterisk-Users] Multiple AGI Issues   Douglas Garstang
• [Asterisk-Users] is there a web interface to this mailing list?   Douglas Garstang
• [Asterisk-Users] is there a web interface to this mailing list?   Douglas Garstang
• [Asterisk-Users] Non sensical AGI Error   Douglas Garstang
• [Asterisk-Users] Non sensical AGI Error   Douglas Garstang
• [Asterisk-Users] AGI Flakyness *sigh*   Douglas Garstang
• [Asterisk-Users] RE: AGI Flakyness *sigh*   Douglas Garstang
• [Asterisk-Users] RE: AGI Flakyness *sigh*   Douglas Garstang
• [Asterisk-Users] Loops and Variables   Douglas Garstang
• [Asterisk-Users] Loops and Variables   Douglas Garstang
• [Asterisk-Users] Catching _ALL_ characters   Douglas Garstang
• [Asterisk-Users] Streaming Music On Hold   Douglas Garstang
• [Asterisk-Users] Streaming Music On Hold   Douglas Garstang
• [Asterisk-Users] Hints between servers?   Douglas Garstang
• [Asterisk-Users] Polycom 501 ACDlogin   Douglas Garstang
• [Asterisk-Users] auto provision of IP501 polycom   Douglas Garstang
• [Asterisk-Users] Streaming Music On Hold - Reality Check   Douglas Garstang
• [Asterisk-Users] Polycom IP 601 Buddy Watch problems   Douglas Garstang
• [Asterisk-Users] What business IP phone to use   Douglas Garstang
• [Asterisk-Users] BLF not working after reload   Douglas Garstang
• [Asterisk-Users] BLF not working after reload   Douglas Garstang
• [Asterisk-Users] Matching '*'   Douglas Garstang
• [Asterisk-Users] AGI Channel Status   Douglas Garstang
• [Asterisk-Users] AGI Channel Status   Douglas Garstang
• [Asterisk-Users] AGI Channel Status   Douglas Garstang
• [Asterisk-Users] AGI Scripts Terminate too Soon   Douglas Garstang
• [Asterisk-Users] AGI Scripts Terminate too Soon   Douglas Garstang
• [Asterisk-Users] AGI Scripts Terminate too Soon   Douglas Garstang
• [Asterisk-Users] PrivacyManager Broken?   Jeremy G. Gault
• [Asterisk-Users] Streaming Music On Hold - Reality Check   Doug Geary
• [Asterisk-Users] Re: asterisk logger - urgent!!!   Allan Gee
• [Asterisk-Users] Bristuff-0.3.0-PRE-1l and TDM400 with fxo ports   Allan Gee
• [Asterisk-Users] BRI Newbie - What Hardware, PCI, in the US?   Allan Gee
• [Asterisk-Users] vISDN with Asterisk and HFC passive cards.   Allan Gee
• [Asterisk-Users] stream file 16k sample and 16 bit data   Jerry Geis
• [Asterisk-Users] asterisk 1.2.4 seg faulting today had been working fine since update   Jerry Geis
• [Asterisk-Users] seg fault 1.2.4   Jerry Geis
• [Asterisk-Users] seg fault 1.2.4   Jerry Geis
• [Asterisk-Users] seg fault 1.2.4   Jerry Geis
• [Asterisk-Users] sip expire 60   Jerry Geis
• [Asterisk-Users] uniden uip200 loosing registeration   Jerry Geis
• [Asterisk-Users] uniden 1 touch dial   Jerry Geis
• [Asterisk-Users] Anyway to do this?   Scott Geist
• [Asterisk-Users] Polycom IP501 Endless Loop   Michael George
• [Asterisk-Users] Strange SIP registration situation   Michael George
• [Asterisk-Users] Polycom 301 line key display   Bill Gibbs
• [Asterisk-Users] Fwd: Asterisk on Solaris 10 (AMD Opteron, Sun Fire X2100)   Bill Gibbs
• [Asterisk-Users] how to add stun functionality in asterisk   Bill Gibbs
• [Asterisk-Users] Pri Hang up outgoing calls   Xavier Gil
• [Asterisk-Users] Pri Hang up outgoing calls   Xavier Gil
• [Asterisk-Users] PRI in spain with ONO   Xavier Gil
• [Asterisk-Users] dummy Technology/resource for Dial   Morgan Gilroy
• [Asterisk-Users] Performance differences 64-bit vs 32-bit   Morgan Gilroy
• [Asterisk-Users] Asterisk with USB   Morgan Gilroy
• [Asterisk-Users] GotoIf number exists in file. How can i do this?   Morgan Gilroy
• [Asterisk-Users] GotoIf number exists in file. How can i do this?   Morgan Gilroy
• [Asterisk-Users] SIP on IP aliases   Morgan Gilroy
• [Asterisk-Users] Any way to grep through fast moving consolemessages?   Morgan Gilroy
• [Asterisk-Users] No audio? Update your Asterisk   Steve Gladden
• [Asterisk-Users] Streaming MOH   Steve Gladden
• [Asterisk-Users] problems with dialing   Will Glass-Husain
• [Asterisk-Users] chan_bluetooth - concurrent calls?   Rico -mc- Gloeckner
• [Asterisk-Users] fax <-> misdn <-> zap <-> fax // fax <-> misdn <-> ATA <-> fax   Rico -mc- Gloeckner
• [Asterisk-Users] DTMF Tones in RTP Payload as Well as in Events = Duplicate Tones   Max Glucksmann
• [Asterisk-Users] RE: [Asterisk-Users ] RE: Monitor a call in progress. (Steve Totaro)   Max Glucksmann
• [Asterisk-Users] Directed Call Pickup   Bob Goddard
• [Asterisk-Users] asterisk logger - urgent!!!   Bob Goddard
• [Asterisk-Users] Problems with Cepstral and Asterisk   Bob Goddard
• [Asterisk-Users] asterisk logger - urgent!!!   Bob Goddard
• [Asterisk-Users] asterisk logger - urgent!!!   Bob Goddard
• [Asterisk-Users] asterisk logger - urgent!!!   Bob Goddard
• [Asterisk-Users] iLBC issue: An ilbc frame that isn't a multiple of 50 bytes long from RTP (38)   Bob Goddard
• [Asterisk-Users] CISCO 1760 with 1 BRI   Bob Goddard
• [Asterisk-Users] What business IP phone to use   Bob Goddard
• [Asterisk-Users] What business IP phone to use   Bob Goddard
• [Asterisk-Users] OT: VoIP over bonded link   Bob Goddard
• [Asterisk-Users] OT: VoIP over bonded link   Bob Goddard
• [Asterisk-Users] Safely editing voicemail.conf   John Goerzen
• [Asterisk-Users] Problem making outbound calls on TE210P using NFAS   Aldo Gonzalez
• [Asterisk-Users] Problem making outbound calls on TE210P using NFAS   Aldo Gonzalez
• [Asterisk-Users] Problem making outbound calls on TE210P using NFAS   Aldo Gonzalez
• [Asterisk-Users] Hardware recommendations   Adam Goryachev
• [Asterisk-Users] PRI Presentation Restricted bit honored?   Jim Gottlieb
• [Asterisk-Users] TDM04B FXO Asterisk@Home   Nelson Granados
• [Asterisk-Users] Digium TDM04B Outbound routing   Nelson Granados
• [Asterisk-Users] ZapRas   Giordano Grandis
• I: [Asterisk-Users] ZapRas   Giordano Grandis
• [Asterisk-Users] Application pppd   Giordano Grandis
• [Asterisk-Users] Sangoma analog cards?   Michael Graves
• [Asterisk-Users] What business IP phone to use   Michael Graves
• [Asterisk-Users] What business IP phone to use   Michael Graves
• [Asterisk-Users] Asterisk as a dedicated Analog PSTN gateway   Michael Graves
• [Asterisk-Users] Skype vs. an Xlite registered to Asterisk   Michael Graves
• [Asterisk-Users] Covad anyone ...   Michael Graves
• [Asterisk-Users] Aasterisk large-scale deployment w/analog phones   Ed Greenberg
• [Asterisk-Users] Fwd: Which ATA device do you recommend?   Ed Greenberg
• [Asterisk-Users] An array of extensions in my lab   Ed Greenberg
• [Asterisk-Users] Fwd: Which ATA device do you recommend?   Ed Greenberg
• [Asterisk-Users] asterisk error   Ed Greenberg
• [Asterisk-Users] A room full of Cisco 7960s behind NAT   Ed Greenberg
• [Asterisk-Users] A room full of Cisco 7960s behind NAT   Ed Greenberg
• [Asterisk-Users] A room full of Cisco 7960s behind NAT   Ed Greenberg
• [Asterisk-Users] TDM421p: Noisy FXS problem   Robert E. Griffith
• [Asterisk-Users] Channel juggling, what is it good for?   Edwin Groothuis
• [Asterisk-Users] Vegastream clockslip problems   Edwin Groothuis
• [Asterisk-Users] dual TE410, both span 3 is broken   Edwin Groothuis
• [Asterisk-Users] Re: dual TE410, both span 3 is broken (Josh Krueger)   Edwin Groothuis
• [Asterisk-Users] Update to the latest zaptel driver - Congestion gone, but scary write errors replaced it   Edwin Groothuis
• [Asterisk-Users] Re: Incoming Calls Getting Crossed - Weird   Edwin Groothuis
• [Asterisk-Users] spandsp 0.0.2pre25   Jesse Guardiani
• [Asterisk-Users] Re: spandsp 0.0.2pre25   Jesse Guardiani
• [Asterisk-Users] Trouble Chan Spy   David Guarnido
• [Asterisk-Users] bad sound frequency   Nitin Gupta
• [Asterisk-Users] Problem with Playback sound in 64 bit machine   Nitin Gupta
• [Asterisk-Users] Problem with Playback sound in 64 bit machine   Nitin Gupta
• [Asterisk-Users] Problem with Playback sound in 64 bit machine   Nitin Gupta
• [Asterisk-Users] Problem with Playback sound in 64 bit machine   Nitin Gupta
• [Asterisk-Users] Problem with Playback sound in 64 bit machine   Nitin Gupta
• [Asterisk-Users] Dial command to connect two channels and bypass asterisk server   Nitin Gupta
• [Asterisk-Users] Dial command to connect two channels and bypass asterisk server   Nitin Gupta
• [Asterisk-Users] Cisco AS5350   Sahil Gupta
• [Asterisk-Users] Application Faxing using SIP   Craig Guy
• [Asterisk-Users] spandsp 0.0.2pre25   Craig Guy
• [Asterisk-Users] Re: spandsp 0.0.2pre25   Craig Guy
• [Asterisk-Users] AGI Scripts Terminate too Soon   Craig Guy
• [Asterisk-Users] Wildfire messsaging server   Fusion at Gyantec
• [Asterisk-Users] Wildfire messsaging server   Fusion at Gyantec
• [Asterisk-Users] Softphone and 911   Kyle Hagan
• [Asterisk-Users] Solution for 1 time blast of 200, 000 recorded calls   Kyle Hagan
• [Asterisk-Users] Good VoIP providers that support Asterisk PBX's   Kyle Hagan
• [Asterisk-Users] IP phone with many speed dial buttons   David Hajek
• [Asterisk-Users] re: questions about sip requests to asterisk 1.2   Yair Hakak
• [Asterisk-Users] re: questions about sip requests to asterisk 1.2   Yair Hakak
• [Asterisk-Users] re: questions about sip requests to asterisk 1.2   Yair Hakak
• [Asterisk-Users] re: voipjet   Yair Hakak
• [Asterisk-Users] Call centre - * hang's up   Paul Hales
• [Asterisk-Users] Connecting two phones with different codecs   Paul Hales
• [Asterisk-Users] Re: auto provision of IP501 polycom   Paul Hales
• [Asterisk-Users] Asterisk question   Paul Hales
• [Asterisk-Users] Re: Asterisk Question   Paul Hales
• [Asterisk-Users] monitor outgoing calls in queue / campaings   Paul Hales
• [Asterisk-Users] Re: Asterisk Question   Paul Hales
• [Asterisk-Users] Nat, SIP, Realtime problem   Hall, Eric M.
• [Asterisk-Users] Nat, SIP, Realtime problem   Hall, Eric M.
• [Asterisk-Users] Nat, SIP, Realtime problem   Hall, Eric M.
• [Asterisk-Users] Dell PowerEdge 1800 and TE410P   Niall Hallett
• [Asterisk-Users] Dell PowerEdge 1800 and TE410P   Niall Hallett
• [Asterisk-Users] Re: 5, 000 concurrent calls system rollout question   Mike Hammett
• [Asterisk-Users] Multiple call groups   Mike Hammett
• [Asterisk-Users] TDMoE   Mike Hammett
• [Asterisk-Users] Re: Asterisk-Users Digest, Vol 19, Issue 58   Mike Hammett
• [Asterisk-Users] TDMoE   Mike Hammett
• [Asterisk-Users] MOH from RCA jack?   Trevor G. Hammonds
• [Asterisk-Users] Multiple AGI Issues   Freddi Hansen
• [Asterisk-Users] Multiple AGI Issues   Freddi Hansen
• [Asterisk-Users] RE: AGI Flakyness *sigh*   Freddi Hansen
• [Asterisk-Users] Anyway to pass CIC in sip header   Kevin Hanson
• [Asterisk-Users] Anyway to pass CIC in sip header   Kevin Hanson
• [Asterisk-Users] Search for Links for "Communicating PC to PC in the same lan through Asterisk "   Kristof Hardy
• [Asterisk-Users] fax possibilities   James Harper
• [Asterisk-Users] BAD/GOOD Echo Cancel   James Harper
• [Asterisk-Users] TE210P + MicroITX as E1 to TDMoE appliance?   James Harper
• [Asterisk-Users] Re: Multiple TDM400P's in a single machine   James Harper
• [Asterisk-Users] Multiple TDM400P's in a single machine   James Harper
• [Asterisk-Users] TDMoIP and Asterisk   James Harper
• [Asterisk-Users] TDMoIP and Asterisk   James Harper
• [Asterisk-Users] TDMoIP and Asterisk   James Harper
• [Asterisk-Users] TDMoIP and Asterisk   James Harper
• [Asterisk-Users] HST Saphir III ML PCI and Linux/Asterisk   James Harper
• [Asterisk-Users] Asterisk with T1 card on laptop   James Harper
• [Asterisk-Users] SIP Aliases   Darrick Hartman
• [Asterisk-Users] simple iaxmoden configuration   Darrick Hartman
• [Asterisk-Users] Asterisk Follow Me   Darrick Hartman
• [Asterisk-Users] IAXModem/Hylafax problem   Darrick Hartman
• [Asterisk-Users] ztmonitor output weirdness   Ronald Hartmann
• [Asterisk-Users] AGI onAnswer function: does it exist?   Vlasis Hatzistavrou
• FW: [Asterisk-Users] AGI onAnswer function: does it exist?   Vlasis Hatzistavrou
• [Asterisk-Users] routing question: multipath routing for SIP   Script Head
• [Asterisk-Users] limit sip sessions   Script Head
• [Asterisk-Users] inform the agent about the queue he is answering   Script Head
• [Asterisk-Users] Asterisk behind Centrex   Devin Heckman
• [Asterisk-Users] Problem with Wait() and chan_capi-cm?   Florian Heer
• [Asterisk-Users] Problem with Wait() and chan_capi-cm?   Florian Heer
• [Asterisk-Users] Problem with Wait() and chan_capi-cm?   Florian Heer
• [Asterisk-Users] Problem with Wait() and chan_capi-cm?   Florian Heer
• [Asterisk-Users] Calling number rewriting   Jean-Christophe Heger
• [Asterisk-Users] Lots of lost interrupts when running HFC ISDN card in NT1 mode   Erik Hensema
• [Asterisk-Users] 5,000 concurrent calls system rollout question   Joash Herbrink
• [Asterisk-Users] 5,000 concurrent calls system rollout question   Joash Herbrink
• [Asterisk-Users] 5,000 concurrent calls system rollout question   Joash Herbrink
• [Asterisk-Users] PSTN connection via IP/ethernet   Joash Herbrink
• [Asterisk-Users] sipura 841 mass provisioning   Joash Herbrink
• [Asterisk-Users] SetCDRUserField not working in A@H?   Paul Hewlett
• [Asterisk-Users] more cpu intensive echo cancellers ?   Paul Hewlett
• [Asterisk-Users] [help] warning 4246   Paul Hewlett
• [Asterisk-Users] Chan_BT question WAS: Asterisk with USB   Ben Higley
• Need to terminate 7 lines (was: [Asterisk-Users] RE: Asterisk-Users Digest, Vol 19, Issue 10)   Jean-Michel Hiver
• [Asterisk-Users] Slightly OT: OpenPBX.org and Freeswitch   Jean-Michel Hiver
• [Asterisk-Users] IP PAX gateway to PSTN   Jean-Michel Hiver
• [Asterisk-Users] IP PAX gateway to PSTN   Jean-Michel Hiver
• [Asterisk-Users] IP PAX gateway to PSTN   Jean-Michel Hiver
• [Asterisk-Users] re: questions about sip requests to asterisk 1.2   Jean-Michel Hiver
• [Asterisk-Users] change languages from an IVR   Jean-Michel Hiver
• [Asterisk-Users] One way audio - it doesn't make sense   Jean-Michel Hiver
• [Asterisk-Users] Better i18n for Asterisk?   Jean-Michel Hiver
• [Asterisk-Users] SPA-3000 VOIP-PSTN gateway - long delay between answering and ringing   Jean-Michel Hiver
• [Asterisk-Users] Asterisk native sounds now available!   Jean-Michel Hiver
• [Asterisk-Users] Better i18n for Asterisk?   Jean-Michel Hiver
• [Asterisk-Users] PSTN connection via IP/ethernet   Jean-Michel Hiver
• [Asterisk-Users] TDMoIP and Asterisk   Jean-Michel Hiver
• [Asterisk-Users] TDMoIP and Asterisk   Jean-Michel Hiver
• [Asterisk-Users] TDMoIP and Asterisk   Jean-Michel Hiver
• [Asterisk-Users] OT: VoIP over bonded link   Jean-Michel Hiver
• [Asterisk-Users] Is Asterisk a PBX?   Jean-Michel Hiver
• [Asterisk-Users] AGI Scripts Terminate too Soon   Jean-Michel Hiver
• [Asterisk-Users] AGI Scripts Terminate too Soon   Jean-Michel Hiver
• [Asterisk-Users] SIP-H323 Help and Multiple Listening Port   Kenige Ho
• [Asterisk-Users] ISDN30E + T1 crossover cable woes   Phillip Hodges
• [Asterisk-Users] Austria isdn p2p empty DID   Marcus Hofbauer
• [Asterisk-Users] SER + Asterisk   Nick Hoffman
• [Asterisk-Users] To connect between more than 2 asterisk server [links needed ]   Nick Hoffman
• [Asterisk-Users] Hooking up with Ser   Nick Hoffman
• [Asterisk-Users] PSTN connection via IP/ethernet   Nick Hoffman
• [Asterisk-Users] sniffing sip password/uri/host info   Nick Hoffman
• [Asterisk-Users] Asterisk: Agent logs into queue, and there are calls in the queue, but calls don't go to agent.   Scott Hooten
• [Asterisk-Users] Asterisk: Agent logs into queue, and there are calls in the queue, but calls don't go to agent   Scott Hooten
• [Asterisk-Users] Asterisk: Agent logs into queue, and there are calls in the queue, but calls don't go to agent   Scott Hooten
• [Asterisk-Users] Asterisk: Agent logs into queue, and there are calls in the queue, but calls don't go to agent.   Scott Hooten
• [Asterisk-Users] sip channel status - how?   Peter Hoppe
• [Asterisk-Users] sip channel status - how?   Peter Hoppe
• [Asterisk-Users] Winnipeg Canada   Richard Houston
• [Asterisk-Users] fax <-> misdn <-> zap <-> fax // fax <-> misdn <-> ATA <-> fax   Lee Howard
• [Asterisk-Users] Application Faxing using SIP   Lee Howard
• [Asterisk-Users] Application Faxing using SIP   Lee Howard
• [Asterisk-Users] Application Faxing using SIP   Lee Howard
• [Asterisk-Users] Application Faxing using SIP   Lee Howard
• [Asterisk-Users] Asterisk and T38 Fax   Lee Howard
• [Asterisk-Users] fax receive using TDM400P   Lee Howard
• [Asterisk-Users] fax receive using TDM400P   Lee Howard
• [Asterisk-Users] TFTP server for GrandStream BT phones / need testing   Peter Hudec
• [Asterisk-Users] TFTP server for GrandStream BT phones / need testing   Peter Hudec
• [Asterisk-Users] Re: delaying "answer" for a number of rings or an amount   Mark Hulber
• [Asterisk-Users] DTMF Sporadicaly Being Generated   Mark Hulber
• [Asterisk-Users] DTMF Sporadicaly Being Generated   Mark Hulber
• [Asterisk-Users] FW: Re: Delay on Phone ringing   Mark Hulber
• [Asterisk-Users] Best quad-port fxo solution with EC?   Hunt, Bill
• [Asterisk-Users] Asterisk large-scale deployment w/analog phones   Hunt, Bill
• [Asterisk-Users] username not stabled?   Chris A. Icide
• [Asterisk-Users] 5,000 concurrent calls system rollout question   Chris A. Icide
• [Asterisk-Users] Recommended rack-mountable server anyone?   Chris A. Icide
• [Asterisk-Users] How to query a table from the keypad?   Chris A. Icide
• [Asterisk-Users] Different Voice Prompts at Different Times   Faisal Inam
• [Asterisk-Users] Playing sound File using GotoifTime function   Faisal Inam
• [Asterisk-Users] asterisk 1.2.1: everything ok but strange messages appear on linux console   Giorgio Incantalupo
• [Asterisk-Users] transferred calls: not 2 but only 1 recorded by cdr   Giorgio Incantalupo
• [Asterisk-Users] Rights problem with Voicemail and non-root user - yeah I know, I thought I had it fixed...   Giorgio Incantalupo
• [Asterisk-Users] Rights problem with Voicemail and non-root user - yeah I know, I thought I had it fixed...   Giorgio Incantalupo
• [Asterisk-Users] fax receive using TDM400P   Ioan Indreias
• [Asterisk-Users] Swapping lines using dtmf   Ira
• SV: SV: [Asterisk-Users] delaying "answer" for a number of ringsor anamount of time   Ira
• [Asterisk-Users] cmd set with multiple values   Ira
• SV: SV: [Asterisk-Users] delaying "answer" for a number of ringsor anamount of time   Ira
• [Asterisk-Users] a couple of questions   Ira
• [Asterisk-Users] Cisco AS5350   Ira
• [Asterisk-Users] a couple of questions   Ira
• [Asterisk-Users] Asterisk native sounds now available!   Ira
• [Asterisk-Users] Two Lines, Two Businesses   Ira
• [Asterisk-Users] Expression GotoIf - bug or personal misunderstanding?   Ira
• [Asterisk-Users] RE: ex-girlfriend (ex-boyfriend)   Ira
• [Asterisk-Users] RE: ex-girlfriend (ex-boyfriend)   Ira
• [Asterisk-Users] Dialing part of the extension   Ira
• [Asterisk-Users] asterisk + door opener   Ira
• [Asterisk-Users] Dundi Examples   Ira
• [Asterisk-Users] attended call transfer   Ira
• [Asterisk-Users] Aastra phones and common directory?   Ira
• [Asterisk-Users] attended call transfer   Ira
• [Asterisk-Users] is there a web interface to this mailing list?   Ira
• [Asterisk-Users] Grandstream GXP-2000   Ira
• [Asterisk-Users] Grandstream GXP-2000   Ira
• [Asterisk-Users] Send flash through zap channel   Ira
• [Asterisk-Users] Asterisk on SPA-3000 rings extensions only after third ring   Ira
• [Asterisk-Users] Voice Over WiFi   Ira
• [Asterisk-Users] TDM400P digium card   Ira
• [Asterisk-Users] (newby) Is PING a good indicator of latency?   Dmitry Ivanov
• [Asterisk-Users] Voicepulse Connect! and SIP settings   JCC
• [Asterisk-Users] attended call transfer   JCC
• [Asterisk-Users] winnipeg canada   Nabeel Jafferali
• [Asterisk-Users] Call Waiting x100P and Cisco IP Phone   Nabeel Jafferali
• [Asterisk-Users] PRI Presentation Restricted bit honored?   Nabeel Jafferali
• [Asterisk-Users] Uniden UIP200 and Asterisk v1.2.4: problem notregistering   Nabeel Jafferali
• [Asterisk-Users] RE: 488 Not Acceptable Here   Nabeel Jafferali
• [Asterisk-Users] Asterisk 1.2.x + ooh323 from addons - incoming call goes always to default context.   Jarek Jarzebowski
• [Asterisk-Users] Asterisk 1.2.x + oh323 on Debian Sarge.   Jarek Jarzebowski
• [Asterisk-Users] SIP ATA gives no ring tone on IAX2 route   Frederic Jean
• [Asterisk-Users] Dial from AGI = no ring back ??   Frederic Jean
• [Asterisk-Users] Dial from AGI = no ring back ??   Frederic Jean
• [Asterisk-Users] What's with Indications/SetLanguage/Zaptel/RingBack ?   Frederic Jean
• [Asterisk-Users] hardware and network requirements   John Jensen
• [Asterisk-Users] TDM04B FXO Asterisk@Home   John Jensen
• [Asterisk-Users] SIP and firewalls?   John Jensen
• [Asterisk-Users] FYI: new firmware for 7905/12 - RPID support   Pavel Jezek
• [Asterisk-Users] chan iax2 auto congest   Pavel Jezek
• [Asterisk-Users] Re: Asterisk 1.2.4 IAX2 New Jitterbuffer Tuning   Pavel Jezek
• [Asterisk-Users] billing - different tarif per phone   Pavel Jezek
• [Asterisk-Users] Re: Asterisk 1.2.4 IAX2 New Jitterbuffer Tuning   Pavel Jezek
• [Asterisk-Users] Comfort noise support incomplete in Asterisk (RFC 3389)   Pavel Jezek
• [Asterisk-Users] ATA186 V2.15.ms upgrade   Weiming Jiang
• [Asterisk-Users] Asterisk & Broadvoice Incoming Calls Problems   Luis Jimenez
• [Asterisk-Users] Can Asterisk and new ShoreTel 6 talk to each other?   Jimmy
• [Asterisk-Users] Can Asterisk send RTP to a specific port number?   Jimmy
• [Asterisk-Users] Problems with gnugk, asterisk, and ooh323   Joe
• [Asterisk-Users] Call queue design issues and suggestions   Joe
• [Asterisk-Users] Help on queues   Johann
• [Asterisk-Users] Detecting Agents and Chanspy   Johann
• [Asterisk-Users] Polycom Default Ring Volume   Johann
• [Asterisk-Users] callback script?   Arne Morten Johansen
• SV: [Asterisk-Users] callback script?   Arne Morten Johansen
• SV: [Asterisk-Users] callback script?   Arne Morten Johansen
• [Asterisk-Users] GotoIf number exists in file. How can i do this?   Arne Morten Johansen
• SV: [Asterisk-Users] GotoIf number exists in file. How can i do this?   Arne Morten Johansen
• [Asterisk-Users] Developing a call centre app. Communication with asterisk?   Arne Morten Johansen
• [Asterisk-Users] Fromstring when sending e-mail on recieved voicemail   Arne Morten Johansen
• SV: [Asterisk-Users] Re: Fromstring when sending e-mail on recievedvoicemail   Arne Morten Johansen
• SV: [Asterisk-Users] Re: Fromstring when sending e-mail on recievedvoicemail   Arne Morten Johansen
• SV: [Asterisk-Users] Re: Fromstring when sending e-mail on recievedvoicemail   Arne Morten Johansen
• SV: [Asterisk-Users] Re: SV: Re: Fromstring when sending e-mail onrecievedvoicemail   Arne Morten Johansen
• SV: [Asterisk-Users] Re: SV: Re: SV: Re: Fromstring when sending e-mailonrecievedvoicemail   Arne Morten Johansen
• [Asterisk-Users] Asterisk Registering with SER question   Olle E Johansson
• [Asterisk-Users] Asterisk Registering with SER question   Olle E Johansson
• [Asterisk-Users] Unable to Register to Asterisk through Proxy   Olle E Johansson
• [Asterisk-Users] username not stabled? * DO NOT USE USERNAME for locally attached phones!!!   Olle E Johansson
• [Asterisk-Users] (newby) EURO-ISDN line question   Olle E Johansson
• [Asterisk-Users] (newby) Is PING a good indicator of latency?   Olle E Johansson
• [Asterisk-Users] (newby) Is PING a good indicator of latency?   Olle E Johansson
• [Asterisk-Users] username not stabled? * DO NOT USE USERNAME for locally attached phones!!!   Olle E Johansson
• [Asterisk-Users] (newby) Is PING a good indicator of latency?   Olle E Johansson
• [Asterisk-Users] Outbound Call & SIP Results   Olle E Johansson
• [Asterisk-Users] Asterisk video conference   Olle E Johansson
• [Asterisk-Users] PRI Presentation Restricted bit honored?   Olle E Johansson
• [Asterisk-Users] username not stabled? * DO NOT USE USERNAME for locally attached phones!!!   Olle E Johansson
• [Asterisk-Users] Better i18n for Asterisk?   Olle E Johansson
• [Asterisk-Users] IP Authorization   Olle E Johansson
• [Asterisk-Users] SIP groups   Olle E Johansson
• [Asterisk-Users] Test my test-branch!   Olle E Johansson
• [Asterisk-Users] how to add stun functionality in asterisk   Olle E Johansson
• [Asterisk-Users] how to add stun functionality in asterisk   Olle E Johansson
• [Asterisk-Users] * INSTRUCTIONS FOR THE ASTERISK COMMUNITY - PLEASE READ NOW *   Olle E Johansson
• [Asterisk-Users] Fromuser required but overrides SetCallerID   Olle E Johansson
• [Asterisk-Users] Hints between servers?   Olle E Johansson
• [Asterisk-Users] Polycom IP 601 Buddy Watch problems   Olle E Johansson
• [Asterisk-Users] context being ignored by inbound sip call   Olle E Johansson
• [Asterisk-Users] username as extension   Olle E Johansson
• [Asterisk-Users] Re: [asterisk-dev] Possible Bug in SIP Stack.   Olle E Johansson
• [Asterisk-Users] Re: [asterisk-dev] Possible Bug in SIP Stack.   Olle E Johansson
• [Asterisk-Users] Asterisk hangs on 1.2.1   Mark Johnson
• [Asterisk-Users] changing cisco 7940/7960 standard menus ?   Mark Johnson
• [Asterisk-Users] changing cisco 7940/7960 standard menus ?   Mark Johnson
• [Asterisk-Users] Re: Asterisk hangs on 1.2.1   Mark Johnson
• [Asterisk-Users] Voicemail Changes   Mark Johnson
• [Asterisk-Users] asterisk 1.2.4 seg faulting today had been working fine since update   Mark Johnson
• [Asterisk-Users] asterisk 1.2.4 seg faulting today had been working fine since update   Mark Johnson
• [Asterisk-Users] Cisco 79xx and SIP 7.5 Problems   Mark Johnson
• [Asterisk-Users] Difference between VoiceMail and VoiceMail2?   Mojo Jojo
• [Asterisk-Users] I need help on VICIDIAL and auto dial   Vic Jolin
• [Asterisk-Users] Re: Help on Vicidial   Vic Jolin
• SV: [Asterisk-Users] BAD/GOOD Echo Cancel   Jerry Jones
• [Asterisk-Users] More Polycom IP501 questions   Jerry Jones
• [Asterisk-Users] Application Faxing using SIP   Jerry Jones
• [Asterisk-Users] Search for Links for "Communicating PC to PC in the same lan through Asterisk "   John Joseph
• [Asterisk-Users] Search for Links for "Communicating PC to PC in the same lan through Asterisk "   John Joseph
• [Asterisk-Users] To connect between more than 2 asterisk server [ links needed ]   John Joseph
• [Asterisk-Users] trunk 2 IAX server :- getting error ' Unable to support trunking on user 'ho' without zaptel timing'   John Joseph
• [Asterisk-Users] Guidance need for trunking using SIP   John Joseph
• [Asterisk-Users] To connect between more than 2 asterisk server [links needed ]   John Joseph
• [Asterisk-Users] To connect between more than 2 asterisk server [links needed ]   John Joseph
• [Asterisk-Users] Cannot see the caller id , When calls made from one server to another   John Joseph
• [Asterisk-Users] Cannot see the caller id , When calls made from one server to another   John Joseph
• [Asterisk-Users] Anyone in or around Redmond, WA?   Martin Joseph
• [Asterisk-Users] (newby) Is PING a good indicator of latency?   Martin Joseph
• [Asterisk-Users] Re: delaying "answer" for a number of ringsor anamount   Martin Joseph
• [Asterisk-Users] Welltech USA? and Wellgate Products?   Martin Joseph
• [Asterisk-Users] Welltech USA? and Wellgate Products?   Martin Joseph
• [Asterisk-Users] STUPID question? Tellabs echo can cards and PSTN?   Martin Joseph
• [Asterisk-Users] Problem with Playback sound in 64 bit machine   Martin Joseph
• [Asterisk-Users] Best quad-port fxo solution with EC?   Martin Joseph
• [Asterisk-Users] lists problem, Gmail????????   Martin Joseph
• [Asterisk-Users] Re: asterisk t.38 pass   Martin Joseph
• [Asterisk-Users] zoom FXS/FXO gateways   Martin Joseph
• [Asterisk-Users] [OT] List messages and end user outages   Martin Joseph
• [Asterisk-Users] HandyTone 488 ata?   Martin Joseph
• [Asterisk-Users] g.729 woes   Martin Joseph
• [Asterisk-Users] good voip   Martin Joseph
• [Asterisk-Users] Best ATA for general residential deployment??   Martin Joseph
• [Asterisk-Users] DTMF Mode supported by VoiceMail Application   Martin Joseph
• [Asterisk-Users] Asterisk 1.2.4 IAX2 New Jitterbuffer Tuning   Martin Joseph
• [Asterisk-Users] Best ATA for general residential deployment??   Martin Joseph
• [Asterisk-Users] Newbie config help? Wellgate 3701a   Martin Joseph
• [Asterisk-Users] fax receive using TDM400P   Martin Joseph
• [Asterisk-Users] Asterisk Web-Based Voicemail?   Martin Joseph
• [Asterisk-Users] authenticate problem   Martin Joseph
• [Asterisk-Users] Skype vs. an Xlite registered to Asterisk   Martin Joseph
• [Asterisk-Users] Asterisk Web-Based Voicemail?   Martin Joseph
• [Asterisk-Users] Newbie config help? Wellgate 3701a (answers)   Martin Joseph
• [Asterisk-Users] jitterbuffer and DTMF conflict?   Martin Joseph
• [Asterisk-Users] Music on hold and conferencing on OS X   Martin Joseph
• [Asterisk-Users] IAX provider recommendation wanted   Martin Joseph
• [Asterisk-Users] Zoom 5801 problems with *   Martin Joseph
• [Asterisk-Users] FW: Re: Delay on Phone ringing   Martin Joseph
• [Asterisk-Users] Polycom Echo   Martin Joseph
• [Asterisk-Users] Asterisk with T1 card on laptop   Martin Joseph
• [Asterisk-Users] Zoom 5801 problems with *   Martin Joseph
• [Asterisk-Users] Problem with T1 installation   Nitin Joshi
• [Asterisk-Users] Rewind MusicOnHold?   Dan Journo
• [Asterisk-Users] Rewind MusicOnHold?   Dan Journo
• [Asterisk-Users] Rewind MusicOnHold?   Dan Journo
• [Asterisk-Users] Rewind MusicOnHold?   Dan Journo
• [Asterisk-Users] RE: Rewind MusicOnHold?   Dan Journo
• [Asterisk-Users] RE: Rewind MusicOnHold?   Dan Journo
• [Asterisk-Users] RE: Rewind MusicOnHold?   Dan Journo
• [Asterisk-Users] RE: Rewind MusicOnHold?   Dan Journo
• [Asterisk-Users] RE: Rewind MusicOnHold?   Dan Journo
• [Asterisk-Users] Anyone know a good ITSP in Canada that suppo rts *?   Lawrence Jovellanos
• [Asterisk-Users] Receiving faxes with spandsp - strange problem   Bartosz Jozwiak
• [Asterisk-Users] ISDN Eicon Diva Server V-BRI   Bartosz Jozwiak
• [Asterisk-Users] ISDN Eicon Diva Server V-BRI   Bartosz Jozwiak
• [Asterisk-Users] channel.c: Avoided deadlock for '0x91a8b20', 10 retries!   Bartosz Jozwiak
• [Asterisk-Users] channel.c: Avoided deadlock for '0x91a8b20', 10 retries!   Bartosz Jozwiak
• [Asterisk-Users] Internal Server Error   Bartosz Jozwiak
• [Asterisk-Users] Bridge Calls with G()   Prakash Rao Kanthi
• [Asterisk-Users] alternative to realtime?   Roy Sigurd Karlsbakk
• [Asterisk-Users] alternative to realtime?   Roy Sigurd Karlsbakk
• [Asterisk-Users] Instant Messaging: with SIP or XMPP   Nicholas Kathmann
• [Asterisk-Users] IAXModem/Hylafax problem   Nicholas Kathmann
• [Asterisk-Users] OT: VoIP over bonded link   Nicholas Kathmann
• [Asterisk-Users] White Noise on TDM22B FXO Channels (newbie)   John Kelly
• [Asterisk-Users] Which Quad Port FXO is Best?   John Kelly
• [Asterisk-Users] interface to dpnss   Steve Kennedy
• [Asterisk-Users] codec negotiation with SPA-3K   Steve Kennedy
• [Asterisk-Users] g.729 woes   Steve Kennedy
• [Asterisk-Users] g.729 woes   Steve Kennedy
• [Asterisk-Users] Asterisk on Solaris 10 (AMD Opteron, SunFire X2100)   Steve Kennedy
• [Asterisk-Users] q931 85   Steve Kennedy
• [Asterisk-Users] Fwd: Asterisk on Solaris 10 (AMD Opteron, Sun Fire X2100)   Steve Kennedy
• [Asterisk-Users] Fwd: Asterisk on Solaris 10 (AMD Opteron, Sun Fire X2100)   Steve Kennedy
• [Asterisk-Users] GPS-enabled cell phone/PDA   Steve Kennedy
• [Asterisk-Users] GSM phone reception range extendor   Steve Kennedy
• [Asterisk-Users] Delaying media stream by short period after 183 is sent   Mark van Kerkwyk
• [Asterisk-Users] Faint background noise/crackle on FXS port on TDM400P   Mark van Kerkwyk
• [Asterisk-Users] Faint background noise/crackle on FXS port on TDM400P   Mark van Kerkwyk
• [Asterisk-Users] ruby-agi-1.1.1 released !   Mohammad Khan
• [Asterisk-Users] Asterisk native sounds now available!   Kristian Kielhofner
• [Asterisk-Users] Asterisk native sounds now available!   Kristian Kielhofner
• [Asterisk-Users] Asterisk native sounds now available!   Kristian Kielhofner
• [Asterisk-Users] Asterisk native sounds now available!   Kristian Kielhofner
• [Asterisk-Users] Asterisk native sounds now available!   Kristian Kielhofner
• [Asterisk-Users] Asterisk native sounds now available!   Kristian Kielhofner
• [Asterisk-Users] Asterisk native sounds now available!   Kristian Kielhofner
• [Asterisk-Users] Asterisk native sounds now available!   Kristian Kielhofner
• [Asterisk-Users] Asterisk native sounds now available!   Kristian Kielhofner
• [Asterisk-Users] Asterisk Native Sounds re-release   Kristian Kielhofner
• [Asterisk-Users] MOH from RCA jack?   Kristian Kielhofner
• [Asterisk-Users] MOH from RCA jack?   Kristian Kielhofner
• [Asterisk-Users] MOH from RCA jack?   Kristian Kielhofner
• [Asterisk-Users] What business IP phone to use   Kristian Kielhofner
• [Asterisk-Users] A room full of Cisco 7960s behind NAT   Kristian Kielhofner
• [Asterisk-Users] How hard to create Asterisk for Compact Flash?   Kristian Kielhofner
• [Asterisk-Users] Asterisk + Avaya DTMF problem   Matt King
• [Asterisk-Users] Queues and On Hold   Matt King
• [Asterisk-Users] Missing 31 DTMF tones over ZAP   Matt King
• [Asterisk-Users] Re: Missing 31 DTMF tones over ZAP   Matt King
• [Asterisk-Users] Planet VoIP Phones   Andrew Kirch
• [Asterisk-Users] Asterisk compile error   Andrew D Kirch
• [Asterisk-Users] voicemail files in Asterisk have rights 600 , I need 644   Ben Klang
• [Asterisk-Users] any doc/example for app_sms.so ?   Philipp von Klitzing
• AW: [Asterisk-Users] MP3player Problem   Rene Kluwen
• [Asterisk-Users] help on dial plan   Wooi Koay
• [Asterisk-Users] authenticate problem   Wooi Koay
• [Asterisk-Users] Anyone know a good ITSP in Canada that supports *?   Andrew Kohlsmith
• [Asterisk-Users] CallerID popup   Andrew Kohlsmith
• [Asterisk-Users] CallerID popup   Andrew Kohlsmith
• [Asterisk-Users] more cpu intensive echo cancellers ?   Andrew Kohlsmith
• [Asterisk-Users] Polycom IP501 with Asterisk - distinctive ring?   Andrew Kohlsmith
• SOLVED: Re: [Asterisk-Users] Polycom IP501 with Asterisk - distinctive ring?   Andrew Kohlsmith
• [Asterisk-Users] More Polycom IP501 questions   Andrew Kohlsmith
• [Asterisk-Users] Re: More Polycom IP501 questions   Andrew Kohlsmith
• [Asterisk-Users] Setting up Polycom 501 with 2 Different Extensions   Andrew Kohlsmith
• [Asterisk-Users] Setting up Polycom 501 with 2 Different Extensions   Andrew Kohlsmith
• [Asterisk-Users] Rough Two Days   Andrew Kohlsmith
• [Asterisk-Users] Channel bleedover?   Andrew Kohlsmith
• [Asterisk-Users] RE: Random Disconnects - or ARE they?   Andrew Kohlsmith
• [Asterisk-Users] TE411P Really Bad Echo   Andrew Kohlsmith
• [Asterisk-Users] RE: Random Disconnects - or ARE they?   Andrew Kohlsmith
• [Asterisk-Users] RE: AGI Flakyness *sigh*   Andrew Kohlsmith
• [Asterisk-Users] [slackware 10.2 and TE205P] Unknown signalling method 'pri_cpe'   Andrew Kohlsmith
• [Asterisk-Users] Analyzer for Milliwatt   Andrew Kohlsmith
• [Asterisk-Users] OT: VoIP over bonded link   Andrew Kohlsmith
• [Asterisk-Users] attended call transfer   Koopmann, Jan-Peter
• [Asterisk-Users] Asterisk and Snom 360   Koopmann, Jan-Peter
• [Asterisk-Users] SIP and firewalls?   Koopmann, Jan-Peter
• [Asterisk-Users] how to add stun functionality in asterisk   Koopmann, Jan-Peter
• [Asterisk-Users] HELP, SPA-2002 - SPA-2002 singleside sound   Bernard van de Koppel
• [Asterisk-Users] Oh323 channel problem   Reto Kortas
• [Asterisk-Users] Sipura SPA-2002 rings randomly   Jeremy Koski
• [Asterisk-Users] Sipura SPA-2002 rings randomly   Jeremy Koski
• [Asterisk-Users] Cisco 79xx and SIP 7.5 Problems   Jeremy Koski
• [Asterisk-Users] international calling via POTS in Russia   Balint Kovacs
• [Asterisk-Users] polycom and its minibrowser   Anton Krall
• [Asterisk-Users] Re: Polycom boot times/XML files.   Anton Krall
• [Asterisk-Users] Hardware recommendations   Anton Krall
• [Asterisk-Users] cannot change distinctive ring polycom phones   Anton Krall
• [Asterisk-Users] Re: Polycom IP501 with Asterisk - distinctive   Anton Krall
• [Asterisk-Users] Re: More Polycom IP501 questions   Anton Krall
• [Asterisk-Users] Polycom 501 ACDlogin   Anton Krall
• [Asterisk-Users] fax receive using TDM400P   Anton Krall
• [Asterisk-Users] spandsp debug or logging   Anton Krall
• [Asterisk-Users] spandsp debug or logging   Anton Krall
• [Asterisk-Users] fax receive using TDM400P   Anton Krall
• [Asterisk-Users] fax receive using TDM400P   Anton Krall
• [Asterisk-Users] Polycom Default Ring Volume   Anton Krall
• [Asterisk-Users] fax receive using TDM400P   Anton Krall
• [Asterisk-Users] fax receive using TDM400P   Anton Krall
• [Asterisk-Users] always on agents and queues   Anton Krall
• [Asterisk-Users] fax receive using TDM400P   Anton Krall
• [Asterisk-Users] fax receive using TDM400P   Anton Krall
• [Asterisk-Users] fax receive using TDM400P   Anton Krall
• [Asterisk-Users] fax receive using TDM400P   Anton Krall
• [Asterisk-Users] always on agents and queues   Anton Krall
• [Asterisk-Users] fax receive using TDM400P   Anton Krall
• [Asterisk-Users] fax receive using TDM400P   Anton Krall
• [Asterisk-Users] fax receive using TDM400P   Anton Krall
• [Asterisk-Users] fax receive using TDM400P   Anton Krall
• [Asterisk-Users] fax receive using TDM400P   Anton Krall
• [Asterisk-Users] Internal Server Error   Anton Krall
• [Asterisk-Users] Internal Server Error   Anton Krall
• [Asterisk-Users] Internal Server Error   Anton Krall
• [Asterisk-Users] How to query a table from the keypad?   Anton Krall
• [Asterisk-Users] Internal Server Error   Anton Krall
• [Asterisk-Users] Polycom Default Ring Volume   Anton Krall
• [Asterisk-Users] fax receive using TDM400P   Anton Krall
• [Asterisk-Users] Polycom Default Ring Volume   Anton Krall
• [Asterisk-Users] fax receive using TDM400P   Anton Krall
• [Asterisk-Users] fax receive using TDM400P   Anton Krall
• [Asterisk-Users] fax receive using TDM400P   Anton Krall
• [Asterisk-Users] fax receive using TDM400P   Anton Krall
• [Asterisk-Users] Re: Polycom Default Ring Volume [OT]   Anton Krall
• [Asterisk-Users] fax receive using TDM400P   Anton Krall
• [Asterisk-Users] fax receive using TDM400P   Anton Krall
• [Asterisk-Users] Polycom Echo   Anton Krall
• [Asterisk-Users] Re: Polycom Default Ring Volume [OT]   Anton Krall
• [Asterisk-Users] Polycom Echo   Anton Krall
• [Asterisk-Users] Fax transmission interrupt on ISDN network   Olivier Krief
• [Asterisk-Users] Asterisk and Snom 360   Olivier Krief
• [Asterisk-Users] Asterisk and Snom 360   Olivier Krief
• [Asterisk-Users] Asterisk and Snom 360   Olivier Krief
• [Asterisk-Users] automatically start application from the command prompt   Arjan Kroon
• [Asterisk-Users] automatically start application from thecommandprompt   Arjan Kroon
• [Asterisk-Users] dual TE410, both span 3 is broken   Josh Krueger
• [Asterisk-Users] h323 configuration   Patricio Ku
• [Asterisk-Users] RE: Cisco 7960 upgrade to SIP   Kaleb L. Kunzler
• [Asterisk-Users] FW: 7960-tones.xml (Schochet, Wes)   Kaleb L. Kunzler
• [Asterisk-Users] Anyway to do this?   Andy Kuo
• [Asterisk-Users] lists problem, Gmail????????   Andy Kuo
• [Asterisk-Users] RE: ZAP extension, DTMF?   Andy Kuo
• [Asterisk-Users] Asterisk and T38 Fax   Andy Kuo
• [Asterisk-Users] Re: [Serusers] Deploying VoIP on a WAN   Jiri Kuthan
• [Asterisk-Users] Remapping Polycom IP501 buttons   Henry Kwan
• [Asterisk-Users] Re: Remapping Polycom IP501 buttons   Henry Kwan
• [Asterisk-Users] What business IP phone to use   Henry Kwan
• [Asterisk-Users] TDM400P digium card   Henry Kwan
• [Asterisk-Users] Re: Polycom Default Ring Volume [OT]   Henry Kwan
• [Asterisk-Users] Asterisk, SIP phone , NAT   Paul Lacatus
• [Asterisk-Users] Asterisk on laptop connected to POTS line   Mr. James W. Laferriere
• [Asterisk-Users] Regarding cdr_manager.conf   Edwin Lam
• [Asterisk-Users] French and German translations?   Philippe Lang
• [Asterisk-Users] Changes to sip.conf in 1.2.x ?   John Lange
• [Asterisk-Users] Meetme echo cancellation   Steven Langley
• [Asterisk-Users] 5,000 concurrent calls system rollout question   Kristian Larsson
• [Asterisk-Users] 5,000 concurrent calls system rollout question   Kristian Larsson
• [Asterisk-Users] Codec Selection   Abdul Lateef
• [Asterisk-Users] Codec Selection   Abdul Lateef
• [Asterisk-Users] Codec Selection   Abdul Lateef
• [Asterisk-Users] Codec Selection   Abdul Lateef
• [Asterisk-Users] OOH323 Configuration   Abdul Lateef
• [Asterisk-Users] OH323 Peer   Abdul Lateef
• [Asterisk-Users] OH323 Peer   Abdul Lateef
• [Asterisk-Users] Next Montreal meeting - the 21st - featuring a conference call with Mark Spencer   Adrien Laurent
• [Asterisk-Users] Calls fading in and out   Nora Lavelle
• [Asterisk-Users] Calls fading in and out   Nora Lavelle
• [Asterisk-Users] Asterisk vs. Traditional PBX   Nora Lavelle
• [Asterisk-Users] Asterisk vs. Traditional PBX   Nora Lavelle
• [Asterisk-Users] Asterisk vs. Traditional PBX   Nora Lavelle
• [Asterisk-Users] Asterisk vs. Traditional PBX   Nora Lavelle
• [Asterisk-Users] TDM400P digium card   Nora Lavelle
• [Asterisk-Users] TDM400P digium card   Nora Lavelle
• [Asterisk-Users] TDM400P digium card   Nora Lavelle
• [Asterisk-Users] TDM400P digium card   Nora Lavelle
• [Asterisk-Users] Sipura SPA-2002 rings randomly   Benjamin Lawetz
• [Asterisk-Users] SIP question   Benjamin Lawetz
• [Asterisk-Users] Dialing multiple phones with Macro-exten-vm   Jason Lee
• [Asterisk-Users] Dialing multiple phones with Macro-exten-vm   Jason Lee
• [Asterisk-Users] Dialing multiple phones with Macro-exten-vm   Jason Lee
• [Asterisk-Users] Voicemail Problem   Sam Lee
• [Asterisk-Users] Voicemailmain() refusing connection problem   Sam Lee
• [Asterisk-Users] SPA-3000 VOIP-PSTN gateway - longdelaybetweenanswering and ringing   Sam Lee
• [Asterisk-Users] Voicemail Problem   Sam Lee
• [Asterisk-Users] Voicemailmain() refusing connection problem   Sam Lee
• [Asterisk-Users] Voicemail Problem   Sam Lee
• [Asterisk-Users] How come I don't have the MeetMe applicationregistered?   Sam Lee
• [Asterisk-Users] Voicemail Problem   Sam Lee
• [Asterisk-Users] Asterisk video conference   Shain Lee
• [Asterisk-Users] a recipe for compiling asterisk 1.2.4 with h.323 support   Lenz
• [Asterisk-Users] attended call transfer   Lenz
• [Asterisk-Users] Developing a call centre app. Communication with asterisk?   Lenz
• [Asterisk-Users] Problems with voicemail   Roger Lewau
• SV: [Asterisk-Users] Problems with voicemail   Roger Lewau
• SV: [Asterisk-Users] Problems with voicemail   Roger Lewau
• [Asterisk-Users] Forwarding any number issue   Leon Yong Li
• [Asterisk-Users] Opinions needed on call quality vs network latency   Michael J. Liberatore
• [Asterisk-Users] Help on queues   Michael J. Liberatore
• [Asterisk-Users] intel 536 ep as fxo -> possible?   Michael J. Liberatore
• [Asterisk-Users] Grandstream GXP-2000   Michael J. Liberatore
• [Asterisk-Users] Bridged line appearance   Michael J. Liberatore
• [Asterisk-Users] snom 360 incorrect US indications   Michael J. Liberatore
• [Asterisk-Users] Grandstream GXP-2000   Michael J. Liberatore
• [Asterisk-Users] Fwd: Which ATA device do you recommend?   Michael J. Liberatore
• [Asterisk-Users] Re: SPA-941 stutter tone   Michael J. Liberatore
• [Asterisk-Users] SPA-941 & hint   Michael J. Liberatore
• [Asterisk-Users] one way / irratic voice over iax and g729   Michael J. Liberatore
• [Asterisk-Users] Asterisk and Snom 360   Michael J. Liberatore
• [Asterisk-Users] Good VoIP providers that support Asterisk PBX's   Michael J. Liberatore
• [Asterisk-Users] Asterisk and Snom 360   Michael J. Liberatore
• [Asterisk-Users] Incoming Calls Getting Crossed - Weird   Michael J. Liberatore
• [Asterisk-Users] Incoming Calls Getting Crossed - Weird   Michael J. Liberatore
• [Asterisk-Users] Incoming Calls Getting Crossed - Weird   Michael J. Liberatore
• [Asterisk-Users] Asterisk 1.2.4 IAX2 New Jitterbuffer Tuning   Michael J. Liberatore
• [Asterisk-Users] Re: Incoming Calls Getting Crossed - Weird   Michael J. Liberatore
• [Asterisk-Users] Good VoIP providers that support Asterisk PBX's   Michael J. Liberatore
• [Asterisk-Users] Update to the latest zaptel driver - Congestion gone, but scary write errors replaced it   Keiron Liddle
• [Asterisk-Users] Skilled API consultant required - preferably with Salesforce.com intergration   Paul Liew
• [Asterisk-Users] TE411P Really Bad Echo   Isaac Xiao (KVB Kunlun Pty Limited)
• [Asterisk-Users] strange problem with asterisk in media proxy mode   VoIP Linux
• [Asterisk-Users] Line Dropouts on E405P   Asterisk - Mailing List
• [Asterisk-Users] Line Dropouts on E405P   Asterisk - Mailing List
• [Asterisk-Users] Outbound Routing does not use Multiple Trunks   Nate List
• [Asterisk-Users] Outbound Routing does not use Multiple Trunks   Nate List
• [Asterisk-Users] COMMPARTNERS Resellers   Lists
• [Asterisk-Users] (newby) IAX Trunk on low bandwidth connection   Rob Lith
• [Asterisk-Users] (newby) IAX Trunk on low bandwidth connection   Rob Lith
• [Asterisk-Users] Handset phone to replace Flash Operator Panel   Rob Lith
• [Asterisk-Users] PRI Bridging and Recording   Rob Lith
• [Asterisk-Users] stable ISDN BRI card for asterisk   Rob Lith
• [Asterisk-Users] Re: How many TDM2400P's will a server take?   Rob Lith
• [Asterisk-Users] TE411P Really Bad Echo   Rob Lith
• [Asterisk-Users] TE411P Really Bad Echo   Rob Lith
• [Asterisk-Users] TE411P Really Bad Echo   Rob Lith
• [Asterisk-Users] TE411P Really Bad Echo   Rob Lith
• [Asterisk-Users] attended call transfer   Rob Lith
• [Asterisk-Users] Zap, Caller ID problem   Rob Lith
• [Asterisk-Users] IP phone with many speed dial buttons   Rob Lith
• [Asterisk-Users] TE411P Really Bad Echo   Rob Lith
• [Asterisk-Users] asterisk silence suppression?   Rob Lith
• [Asterisk-Users] Problem with ARI and seeing voicemail...   Dan Littlejohn
• [Asterisk-Users] ARI 0.06   Dan Littlejohn
• [Asterisk-Users] ARI 0.06   Dan Littlejohn
• [Asterisk-Users] What to know for installing ARI   Dan Littlejohn
• [Asterisk-Users] Asterisk Web-Based Voicemail?   Dan Littlejohn
• [SPAM-Score-10.0] [Asterisk-Users] Asterisk native sounds now available!   Tim Litwiller
• [SPAM-Score-10.0] [Asterisk-Users] Asterisk native sounds now available!   Tim Litwiller
• [SPAM-Score-10.0] [Asterisk-Users] Asterisk native sounds now available!   Tim Litwiller
• [Asterisk-Users] Asterisk native sounds now available!   Tim Litwiller
• [Asterisk-Users] Session Media 183 and Ringing Tone 180 Passing To SIP At the Same Time   David Liu
• [Asterisk-Users] Zhone channel Banks   William Lloyd
• [Asterisk-Users] 5,000 concurrent calls system rollout question   Michael Loftis
• [Asterisk-Users] 5,000 concurrent calls system rollout question   Michael Loftis
• [Asterisk-Users] 5,000 concurrent calls system rollout question   Michael Loftis
• [Asterisk-Users] Use one sip account for multiple sipura   Reli Loin
• [Asterisk-Users] Use one sip account for multiple sipura   Reli Loin
• [Asterisk-Users] change languages from an IVR   Marcel Eric Loiselle
• [Asterisk-Users] Clipcomm product feedback required   Marcel Eric Loiselle
• [Asterisk-Users] Anyone in or around Redmond, WA?   Nilesh Londhe
• [Asterisk-Users] click to talk   Nilesh Londhe
• [Asterisk-Users] Asterisk with rhinobell.net   Nilesh Londhe
• [Asterisk-Users] Skype-to-Asterisk(SIP): progress   Nilesh Londhe
• [Asterisk-Users] Asterisk Follow Me   Nilesh Londhe
• [Asterisk-Users] Asterisk and Snom 360   Darrell Long
• [Asterisk-Users] Best ATA for general residential deployment??   Darrell Long
• [Asterisk-Users] Best ATA for general residential deployment??   Darrell Long
• [Asterisk-Users] Blocked Callerid   Alexander Lopez
• [Asterisk-Users] Blocked Callerid   Alexander Lopez
• [Asterisk-Users] Asterisk on laptop connected to POTS line   Alexander Lopez
• [Asterisk-Users] TE411P or TE406P   Alexander Lopez
• [Asterisk-Users] Streaming MOH   Alexander Lopez
• [Asterisk-Users] Rewind MusicOnHold?   Alexander Lopez
• [Asterisk-Users] Callerid Name   Alexander Lopez
• [Asterisk-Users] Rewind MusicOnHold?   Alexander Lopez
• [Asterisk-Users] virtual extension per user ?   Alexander Lopez
• [Asterisk-Users] SetCallerID and CDR   Alexander Lopez
• [Asterisk-Users] PRI to PRI not passing callerid   Alexander Lopez
• [Asterisk-Users] PRI to PRI not passing callerid   Alexander Lopez
• [Asterisk-Users] Connecting to live calls   Alexander Lopez
• [Asterisk-Users] Two Lines, Two Businesses   Alexander Lopez
• [Asterisk-Users] Connecting to live calls   Alexander Lopez
• [Asterisk-Users] TDMoE   Alexander Lopez
• [Asterisk-Users] Re: Asterisk-Users Digest, Vol 19, Issue 58   Alexander Lopez
• [Asterisk-Users] Two Lines, Two Businesses   Alexander Lopez
• [Asterisk-Users] IP Authorization   Alexander Lopez
• [Asterisk-Users] IP Authorization   Alexander Lopez
• [Asterisk-Users] RE: ex-girlfriend (ex-boyfriend)   Alexander Lopez
• [Asterisk-Users] T1 Channel splitting PRI/data not working   Alexander Lopez
• [Asterisk-Users] Re: asterisk logger - urgent!!!   Alexander Lopez
• [Asterisk-Users] consult about Digium Card   Alexander Lopez
• [Asterisk-Users] RE: Random Disconnects - or ARE they?   Alexander Lopez
• [Asterisk-Users] DID's   Alexander Lopez
• [Asterisk-Users] Cisco 7960 won't register   Alexander Lopez
• [Asterisk-Users] Aasterisk large-scale deployment w/analog phones   Alexander Lopez
• [Asterisk-Users] MOH from RCA jack?   Alexander Lopez
• [Asterisk-Users] g.729 woes   Alexander Lopez
• [Asterisk-Users] Linear Queues Strategies for 3rd Party Application   Alexander Lopez
• [Asterisk-Users] Incoming Calls Getting Crossed - Weird   Alexander Lopez
• [Asterisk-Users] Incoming Calls Getting Crossed - Weird   Alexander Lopez
• [Asterisk-Users] Cisco upgrade to SIP was: Covad anyone ...   Alexander Lopez
• [Asterisk-Users] Re: Asterisk Question   Alexander Lopez
• [Asterisk-Users] A room full of Cisco 7960s behind NAT   Alexander Lopez
• [Asterisk-Users] consult about Digium Card   Luz Lopez
• [Asterisk-Users] consult about Digium Card   Luz Lopez
• [Asterisk-Users] consult about Digium Card   Luz Lopez
• [Asterisk-Users] consult about Digium Card   Luz Lopez
• [Asterisk-Users] simple iaxmoden configuration   Christian Lox
• [Asterisk-Users] Asterisk as MGCP User Agent   David Lublink
• [Asterisk-Users] Asterisk as MGCP User Agent   David Lublink
• [Asterisk-Users] Asterisk as MGCP User Agent   David Lublink
• [Asterisk-Users] register => 2345:password@sip_proxy doesn't care about port   Bruno De Luca
• [Asterisk-Users] Monitor a call in progress.   Fernando Lujan
• [Asterisk-Users] Monitor a call in progress.   Fernando Lujan
• [Asterisk-Users] Fast AGI performance question   Eric Lyons
• [Asterisk-Users] BAD/GOOD Echo Cancel   Doug Lytle
• [Asterisk-Users] BAD/GOOD Echo Cancel   Doug Lytle
• [Asterisk-Users] BAD/GOOD Echo Cancel   Doug Lytle
• [Asterisk-Users] BAD/GOOD Echo Cancel   Doug Lytle
• [Asterisk-Users] two tellabs 2572 echo board in a 253c mounting assembly?   Doug Lytle
• [Asterisk-Users] BAD/GOOD Echo Cancel   Doug Lytle
• [Asterisk-Users] Re: two tellabs 2572 echo board in a 253c mounting   Doug Lytle
• [Asterisk-Users] IVR Menu   Doug Lytle
• [Asterisk-Users] Re: two tellabs 2572 echo board in a 253c mounting   Doug Lytle
• [Asterisk-Users] Polycom dialplan restriction   Doug Lytle
• [Asterisk-Users] STUPID question? Tellabs echo can cards and PSTN?   Doug Lytle
• [Asterisk-Users] Cisco 79XX firmware 7.5   Doug Lytle
• [Asterisk-Users] TE411P Really Bad Echo ORION   Doug Lytle
• [Asterisk-Users] Asterisk 1.2.4 and IAX MOH   Doug Lytle
• [Asterisk-Users] Asterisk 1.2.4 and IAX MOH   Doug Lytle
• [Asterisk-Users] MOH broke with 1.2.4 .. ?   Doug Lytle
• [Asterisk-Users] Software E.C. Along with Tellabs   Doug Lytle
• [Asterisk-Users] Software E.C. Along with Tellabs   Doug Lytle
• [Asterisk-Users] Random Disconnects - or ARE they?   Doug Lytle
• [Asterisk-Users] is there a web interface to this mailing list?   Doug Lytle
• [Asterisk-Users] Channel bank woes - no outbound calls   Doug Lytle
• [Asterisk-Users] Software E.C. Along with Tellabs   Doug Lytle
• [Asterisk-Users] Channel bank woes - no outbound calls   Doug Lytle
• [Asterisk-Users] Random Disconnects - or ARE they?   Doug Lytle
• [Asterisk-Users] Viking CPC-Disconnect   Doug Lytle
• [Asterisk-Users] Loops and Variables   Doug Lytle
• [Asterisk-Users] Loops and Variables   Doug Lytle
• [Asterisk-Users] Loops and Variables   Doug Lytle
• [Asterisk-Users] Loops and Variables   Doug Lytle
• [Asterisk-Users] asterisk error   Doug Lytle
• [Asterisk-Users] BAD/GOOD Echo Cancel   Doug Lytle
• [Asterisk-Users] mpg123 alternative?   Doug Lytle
• [Asterisk-Users] How can I debug spandsp?   Doug Lytle
• [Asterisk-Users] Problem with T1 installation   Doug Lytle
• [Asterisk-Users] Problem with T1 installation   Doug Lytle
• [Asterisk-Users] Call quality problems   Doug Lytle
• [Asterisk-Users] Cisco 7960 (SIP) with Asterisk: how to get # towork during a call   Doug Lytle
• [Asterisk-Users] Call quality problems   Doug Lytle
• [Asterisk-Users] Call quality problems   Doug Lytle
• [Asterisk-Users] Internal Server Error   Doug Lytle
• [Asterisk-Users] Internal Server Error   Doug Lytle
• [Asterisk-Users] mpg123 alternative?   Doug Lytle
• [Asterisk-Users] Re: How can I debug spandsp?   Doug Lytle
• [Asterisk-Users] A room full of Cisco 7960s behind NAT   Doug Lytle
• [Asterisk-Users] A room full of Cisco 7960s behind NAT   Doug Lytle
• [Asterisk-Users] (newby) IAX Trunk on low bandwidth connection   Guillermo Salas M
• [Asterisk-Users] No path to translate from Zap to SIP   Guillermo Salas M
• [Asterisk-Users] No path to translate from Zap to SIP   Guillermo Salas M
• [Asterisk-Users] calling from SIP to a h.323 device with oh323   Guillermo Salas M
• [Asterisk-Users] calling from SIP to a h.323 device with oh323   Guillermo Salas M
• [Asterisk-Users] Problema calling from elesign h.323 to iax   Guillermo Salas M
• [Asterisk-Users] Re: a2billing without IVR   Guillermo Salas M
• [Asterisk-Users] Skype vs. an Xlite registered to Asterisk   Guillermo Salas M
• [Asterisk-Users] Anyone using LG LIP-100 ip phone   Guillermo Salas M
• [Asterisk-Users] Problema calling from elesign h.323 to iax device   Guillermo Salas M.
• [Asterisk-Users] Problems with gnugk, asterisk, and ooh323   Guillermo Salas M.
• [Asterisk-Users] help with oh323   Guillermo Salas M.
• [Asterisk-Users] IAXModem/Hylafax problem   Julian J. M.
• [Asterisk-Users] Zap channels Deactivated with Bristuff-0.3.x after upgrade from 0.2.0   Olivier MONNET
• [Asterisk-Users] DID's   JOAO CARLOS MOURA
• [Asterisk-Users] DID's   JOAO CARLOS MOURA
• [Asterisk-Users] Dundi Examples   Leif Madsen
• [Asterisk-Users] Re: Voicemail - direct call   Mikael Magnusson
• [Asterisk-Users] 5,000 concurrent calls system rollout question   Paul Mahler
• [Asterisk-Users] Intro to Asterisk & VoIP telephony course - March 21st London seats still available   Paul Mahler
• [Asterisk-Users] Secure voicemail passwords?   Scott Maier
• [Asterisk-Users] Polycom IP 601 Buddy Watch problems   Marco Maiolini
• [Asterisk-Users] Polycom IP 601 Buddy Watch doesn't work after Asterisk reload   Marco Maiolini
• [Asterisk-Users] BLF not working after reload   Marco Maiolini
• [Asterisk-Users] BLF not working after reload   Marco Maiolini
• [Asterisk-Users] Cisco 7905 can't register   Jeremy Malcolm
• [Asterisk-Users] RE: Cisco 7905 can't register   Jeremy Malcolm
• [Asterisk-Users] RE: Cisco 7905 can't register   Jeremy Malcolm
• [Asterisk-Users] RE: Cisco 7905 can't register - SOLVED   Jeremy Malcolm
• [Asterisk-Users] Explain Yellow Alarm in a Legacy Integration   Geoff Manning
• [Asterisk-Users] Cisco 79xx and SIP 7.5 Problems   Geoff Manning
• [Asterisk-Users] Re: Explain Yellow Alarm in a Legacy Integration   Geoff Manning
• [Asterisk-Users] H.323 ( HW PBX to *)   Pedro Mansilla
• [Asterisk-Users] RE: 5, 000 concurrent calls system rollout question   Rafael Marconi
• [Asterisk-Users] Setting MSN for outgoing ISDN calls   Henry Margies
• [Asterisk-Users] Setting MSN for outgoing ISDN calls   Henry Margies
• [Asterisk-Users] Outbound Caller ID number on E1   Henry Margies
• [Asterisk-Users] TDM 400 FXO FXS Test   Jaco Maritz
• [Asterisk-Users] (newby) IAX Trunk on low bandwidth connection   Master_PE
• [Asterisk-Users] TE411P or TE406P   Matt
• [Asterisk-Users] Call completes but then drops?   Matt
• [Asterisk-Users] Call completes but then drops?   Matt
• [Asterisk-Users] Call completes but then drops?   Matt
• [Asterisk-Users] BAD/GOOD Echo Cancel   Matt
• [Asterisk-Users] TE411P Really Bad Echo   Matt
• [Asterisk-Users] Yuck! Asterisk Crash...   Matt
• [Asterisk-Users] TE411P Really Bad Echo   Matt
• [Asterisk-Users] Asterisk Crash   Matt
• [Asterisk-Users] Softphone and 911   Matt
• [Asterisk-Users] Softphone and 911   Matt
• [Asterisk-Users] G723 error   Matt
• [Asterisk-Users] G723 error   Matt
• [Asterisk-Users] Intrado / VoIP E911   Matt
• [Asterisk-Users] how to add stun functionality in asterisk   Matt
• [Asterisk-Users] g.729 woes   Matt
• [Asterisk-Users] G723 error   Matt
• [Asterisk-Users] how to add stun functionality in asterisk   Matt
• [Asterisk-Users] how to add stun functionality in asterisk   Matt
• [Asterisk-Users] Best ATA for general residential deployment??   Matt
• [Asterisk-Users] Best ATA for general residential deployment??   Matt
• [Asterisk-Users] chanspy instability   Matt
• [Asterisk-Users] chanspy instability   Matt
• [Asterisk-Users] Aasterisk large-scale deployment w/analog phones   Joshua McAdam
• [Asterisk-Users] Aasterisk large-scale deployment w/analog phones   Joshua McAdam
• [Asterisk-Users] Asterisk Topology   Ron McCarthy
• [Asterisk-Users] Asterisk Topology   Ron McCarthy
• [Asterisk-Users] Re: [asterisk-biz] iDEFISK (mac iax2 softphone) release   trixter aka Bret McDanel
• [Asterisk-Users] can asterisk to say chinese like say english   trixter aka Bret McDanel
• [Asterisk-Users] g729 license question   trixter aka Bret McDanel
• [Asterisk-Users] g729 license question   trixter aka Bret McDanel
• [Asterisk-Users] g729 license question   trixter aka Bret McDanel
• [Asterisk-Users] G729a Pass-Through and Recording/Monitoring   trixter aka Bret McDanel
• [Asterisk-Users] G729a Pass-Through and Recording/Monitoring   trixter aka Bret McDanel
• [Asterisk-Users] BAD/GOOD Echo Cancel   trixter aka Bret McDanel
• [Asterisk-Users] BAD/GOOD Echo Cancel   trixter aka Bret McDanel
• [Asterisk-Users] BAD/GOOD Echo Cancel   trixter aka Bret McDanel
• [Asterisk-Users] Different Voice Prompts at Different Times   trixter aka Bret McDanel
• [Asterisk-Users] audio cuts out   trixter aka Bret McDanel
• [Asterisk-Users] Fax to Email with Asterisk and Lucent TNT   trixter aka Bret McDanel
• [Asterisk-Users] Fax to Email with Asterisk and Lucent TNT   trixter aka Bret McDanel
• [Asterisk-Users] iax2 trunking known problems?   trixter aka Bret McDanel
• [Asterisk-Users] iax2 trunking known problems?   trixter aka Bret McDanel
• [Asterisk-Users] iax2 trunking known problems?   trixter aka Bret McDanel
• [Asterisk-Users] iax2 trunking known problems?   trixter aka Bret McDanel
• [Asterisk-Users] iax2 trunking known problems?   trixter aka Bret McDanel
• [Asterisk-Users] iax2 trunking known problems?   trixter aka Bret McDanel
• [Asterisk-Users] show calls   trixter aka Bret McDanel
• [Asterisk-Users] free tollfree termination   trixter aka Bret McDanel
• [Asterisk-Users] sipdiscount   trixter aka Bret McDanel
• [Asterisk-Users] Loops and Variables   trixter aka Bret McDanel
• [Asterisk-Users] Loops and Variables   trixter aka Bret McDanel
• [Asterisk-Users] Streaming Music On Hold   trixter aka Bret McDanel
• [Asterisk-Users] username as extension   trixter aka Bret McDanel
• [Asterisk-Users] Double ring   Bob McDowell
• [Asterisk-Users] Double ring   Bob McDowell
• [Asterisk-Users] Aastra phones and common directory?   Bob McDowell
• [Asterisk-Users] Aastra phones and common directory?   Bob McDowell
• [Asterisk-Users] TAPI Recommendations   Bob McDowell
• [Asterisk-Users] TAPI Recommendations   Bob McDowell
• [Asterisk-Users] TAPI Recommendations   Bob McDowell
• [Asterisk-Users] Podget or Similar   Bob McDowell
• [Asterisk-Users] Podget or Similar   Bob McDowell
• [Asterisk-Users] Podget or Similar   Bob McDowell
• [Asterisk-Users] RE: Channel bleedover?   Bob McDowell
• [Asterisk-Users] Random Disconnects - or ARE they?   Bob McDowell
• [Asterisk-Users] Podget or Similar   Bob McDowell
• [Asterisk-Users] 79xx's and call queues   Bob McDowell
• [Asterisk-Users] Playing sound File using GotoifTime function   Bob McDowell
• [Asterisk-Users] MOH from RCA jack?   Bob McDowell
• [Asterisk-Users] Download "Asterisk: The Future Of Telephony" [More Info]   Bob McDowell
• [Asterisk-Users] What business IP phone to use   Bob McDowell
• [Asterisk-Users] IAXModem/Hylafax problem   Bob McDowell
• [Asterisk-Users] IAXModem/Hylafax problem   Bob McDowell
• [Asterisk-Users] RE: IAXModem/Hylafax problem   Bob McDowell
• [Asterisk-Users] Output from "Show Queues" cmd   Callum McGillivray
• [Asterisk-Users] Best ATA for general residential deployment??   S McGowan
• [Asterisk-Users] Best ATA for general residential deployment??   S McGowan
• [Asterisk-Users] Re: Asterisk-Users Digest, Vol 19, Issue 34   patty McHenry
• [Asterisk-Users] digium TE405P and intel motherboard   patty McHenry
• [Asterisk-Users] What business IP phone to use   Raymond McKay
• [Asterisk-Users] Re: How many TDM2400P's will a server take?   Rob McKrill
• [Asterisk-Users] Calls fading in and out   Mike McMullen
• [Asterisk-Users] alternative to realtime?   Jeremy McNamara
• [Asterisk-Users] alternative to realtime?   Jeremy McNamara
• [Asterisk-Users] PIKA Technologies Inc. Announces Support for Open Source Asterisk PBX   Jeremy McNamara
• [Asterisk-Users] Linksys WIP300 WiFi Phone   Me
• [Asterisk-Users] Linksys WIP300 WiFi Phone   Me
• [Asterisk-Users] Asterisk and T38 Fax   Jorge Mendoza
• [Asterisk-Users] Re: BRI Newbie - What Hardware, PCI, in the US?   Doug Meredith
• [Asterisk-Users] Re: SPA-941 stutter tone   Doug Meredith
• [Asterisk-Users] Re: SPA-941 stutter tone   Doug Meredith
• [Asterisk-Users] CallerID popup   Giovanni Miano
• [Asterisk-Users] Events when the target of the call answer   Giovanni Miano
• [Asterisk-Users] re: Polycom IP501 with Asterisk - distinctive ring   Bill Michaelson
• [Asterisk-Users] I must be missing something zimple...   Bill Michaelson
• [Asterisk-Users] GPS-enabled cell phone/PDA   Bill Michaelson
• [Asterisk-Users] limit sip sessions   Miguel
• [Asterisk-Users] limit sip sessions   Miguel
• [Asterisk-Users] ztdummy on gentoo 2005.1   Miguel
• [Asterisk-Users] ztdummy on gentoo 2005.1   Miguel
• [Asterisk-Users] ztdummy on gentoo 2005.1   Miguel
• [Asterisk-Users] ztdummy on gentoo 2005.1 [SOLVED]   Miguel
• [Asterisk-Users] meetme application   Miguel
• [Asterisk-Users] Unable to create channel of type 'SIP'   Miguel
• [Asterisk-Users] incoming peer register problem   Miguel
• [Asterisk-Users] Sound quality issue in one direction and wctdm problem with APIC enabled kernel   Chris Miller
• [Asterisk-Users] ZAP <--> sip(polycom301) can not hear each other   Noah Miller
• [Asterisk-Users] Re: Polycomm IP600 continues to ring   Noah Miller
• [Asterisk-Users] Re: Remapping Polycom IP501 buttons   Noah Miller
• [Asterisk-Users] Re: Polycom IP501 with Asterisk - distinctive   Noah Miller
• [Asterisk-Users] Polycom remapping SpeedDials   Noah Miller
• [Asterisk-Users] Re: More Polycom IP501 questions   Noah Miller
• [Asterisk-Users] Re: auto provision of IP501 polycom   Noah Miller
• [Asterisk-Users] FW: auto provision of IP501 polycom   Noah Miller
• [Asterisk-Users] Re: Asterisk 1.2.4 IAX2 New Jitterbuffer Tuning   Noah Miller
• [Asterisk-Users] Re: Polycom Default Ring Volume [OT]   Noah Miller
• [Asterisk-Users] Re: auto provision of IP501 polycom   Noah I. Miller
• [Asterisk-Users] Re: Polycom Default Ring Volume [OT]   Noah I. Miller
• [Asterisk-Users] Direct pickup   Mimmus
• [Asterisk-Users] Directed Call Pickup   Mimmus
• [Asterisk-Users] Directed Call Pickup   Mimmus
• [Asterisk-Users] CallerID popup   Mimmus
• [Asterisk-Users] CallerID popup   Mimmus
• [Asterisk-Users] CallerID popup   Mimmus
• [Asterisk-Users] CallerID popup   Mimmus
• [Asterisk-Users] Asterisk - Brooktrout   Mimmus
• [Asterisk-Users] Brief pauses during calls   Mimmus
• [Asterisk-Users] Alcatel 4200 series pbx   Mimmus
• [Asterisk-Users] Switch statement   Mimmus
• [Asterisk-Users] RE: virtual extension per user ?   Mimmus
• [Asterisk-Users] Status UNKNOWN   Mimmus
• [Asterisk-Users] RE: RE: virtual extension per user ?   Mimmus
• [Asterisk-Users] Cheap BRI card   Mimmus
• [Asterisk-Users] Grandstream GXP-2000   Mimmus
• [Asterisk-Users] Set CallerIDNum for outgoing calls on a PRI+DDI line   Mimmus
• [Asterisk-Users] Outbound Routing does not use Multiple Trunks   Mimmus
• [Asterisk-Users] What business IP phone to use   Mimmus
• [Asterisk-Users] ISDN30E + T1 crossover cable woes   Mimmus
• [Asterisk-Users] Set CallerIDNum on a PRI   Mimmus
• [Asterisk-Users] DTMF level   Virmones Pereira T. Miranda
• [Asterisk-Users] Grandstream BT-101 POS Error   Keith Mitchell
• [Asterisk-Users] NEED COMMENT ON USING FEDORA CORE 3   Keith Mitchell
• [Asterisk-Users] Cisco 7920 wi-phone firmware   Louis-David Mitterrand
• [Asterisk-Users] Possible Bug in SIP Stack.   Chris Modesitt
• [Asterisk-Users] RE: [asterisk-dev] Possible Bug in SIP Stack.   Chris Modesitt
• [Asterisk-Users] RE: [asterisk-dev] Possible Bug in SIP Stack.   Chris Modesitt
• [Asterisk-Users] RE: [asterisk-dev] Possible Bug in SIP Stack.   Chris Modesitt
• [Asterisk-Users] RE: [asterisk-dev] Possible Bug in SIP Stack.   Chris Modesitt
• [Asterisk-Users] Best ATA for general residential deployment??   Adam Moffett
• [Asterisk-Users] Best ATA for general residential deployment??   Adam Moffett
• [Asterisk-Users] Configure DID   Adam Moffett
• [Asterisk-Users] Dundi key Problem   Jesus Mogollon
• [Asterisk-Users] Voicemail - direct call   Jesus Mogollon
• [Asterisk-Users] Zaptel 1.2.3 with Asterisk 1.0.9   Mojo with Horan & Company, LLC
• [Asterisk-Users] More Polycom IP501 questions   Mojo with Horan & Company, LLC
• [Asterisk-Users] Polycom remapping SpeedDials   Mojo with Horan & Company, LLC
• [Asterisk-Users] FW: auto provision of IP501 polycom   Mojo with Horan & Company, LLC
• [Asterisk-Users] chan_bluetooth - concurrent calls?   Peter Molnar
• [Asterisk-Users] chan_bluetooth - concurrent calls?   Peter Molnar
• [Asterisk-Users] IE Display in SETUP (pri_cpe)   Markus Monka
• [Asterisk-Users] DID's   Vladimir Montealegre
• [Asterisk-Users] RE: ex-girlfriend (ex-boyfriend)   Ricardo Monteiro
• [Asterisk-Users] IVR Menu   Alex Montoanelli
• [Asterisk-Users] MFC/R2 in Brazil   Melcon Moraes
• [Asterisk-Users] Where can I get the tar.gz sources of libnewt?   Melcon Moraes
• [Asterisk-Users] Is Asterisk a PBX?   Melcon Moraes
• [Asterisk-Users] RE: Teliax - Codec Preference effective?   Dan Morin
• [Asterisk-Users] ~1 sec delay from callee answering to call established on dialout   John Morris
• [Asterisk-Users] Possible AGI Bug in Asterisk?   Shad Mortazavi
• [Asterisk-Users] RE: Possible AGI Bug in Asterisk?   Shad Mortazavi
• [Asterisk-Users] Asterisk and T38 Fax   Carey Mould
• [Asterisk-Users] Re: DeadAGI variables confusion   Tony Mountifield
• [Asterisk-Users] SIP on IP aliases   Tony Mountifield
• [Asterisk-Users] Re: Meetme echo cancellation   Tony Mountifield
• [Asterisk-Users] Re: Bridge Calls with G()   Tony Mountifield
• [Asterisk-Users] Re: Download "Asterisk: The Future Of Telephony"   Tony Mountifield
• [Asterisk-Users] RE: X100P help required   Marco Mouta
• [Asterisk-Users] Asterisk running on DMZ (no NAT) PROBLEMS- OPTION message is out of State   Marco Mouta
• [Asterisk-Users] which ATA SIP is better with asterisk   Marco Mouta
• [Asterisk-Users] Fwd: Which ATA device do you recommend?   Marco Mouta
• [Asterisk-Users] Asterisk running on DMZ (no NAT) PROBLEMS- OPTION message is out of State   Marco Mouta
• [Asterisk-Users] FaxToEmail for diferent Channels and different Mail accounts?   Marco Mouta
• [Asterisk-Users] TDM400P FXS problem   Sergio Garcia Murillo
• [Asterisk-Users] Corrupt CDR records in Asterisk 1.2.x   Sergio Garcia Murillo
• [Asterisk-Users] delaying "answer" for a number of rings or an amount of time   Brian J. Murrell
• SV: [Asterisk-Users] delaying "answer" for a number of rings or anamount of time   Brian J. Murrell
• [Asterisk-Users] Re: delaying "answer" for a number of rings or an amount of time   Brian J. Murrell
• [Asterisk-Users] Re: delaying "answer" for a number of ringsor an amount of time   Brian J. Murrell
• SV: SV: [Asterisk-Users] delaying "answer" for a number of ringsor anamount of time   Brian J. Murrell
• [Asterisk-Users] Re: delaying "answer" for a number of ringsor an amount of time   Brian J. Murrell
• [Asterisk-Users] Re: delaying "answer" for a number of rings or an amount   Brian J. Murrell
• SV: SV: [Asterisk-Users] delaying "answer" for a number of ringsor anamount of time   Brian J. Murrell
• [Asterisk-Users] Re: delaying "answer" for a number of ring or an amount of time   Brian J. Murrell
• SV: SV: [Asterisk-Users] delaying "answer" for a number of ringsor anamount of time   Brian J. Murrell
• [Asterisk-Users] Re: delaying "answer" for a number of rings or an amount   Brian J. Murrell
• [Asterisk-Users] dummy Technology/resource for Dial   Brian J. Murrell
• [Asterisk-Users] Asterisk native sounds now available!   Brian J. Murrell
• [Asterisk-Users] How to determine duration call when is used Attended Transfer   Miroslav Nachev
• [Asterisk-Users] Re: [asterisk-biz] iDEFISK (mac iax2 softphone) release   Dinesh Nair
• [Asterisk-Users] file.c:509 ast_openstream_full: File 100 does not exist in any format   Dinesh Nair
• [Asterisk-Users] Queue() with timeout=0   Dinesh Nair
• [Asterisk-Users] 5,000 concurrent calls system rollout question   Dinesh Nair
• [Asterisk-Users] 5,000 concurrent calls system rollout question   Dinesh Nair
• [Asterisk-Users] Re: [asterisk-biz] iDEFISK (mac iax2 softphone) release   Dinesh Nair
• [Asterisk-Users] 5,000 concurrent calls system rollout question   Dinesh Nair
• [Asterisk-Users] (newby) Is PING a good indicator of latency?   Dinesh Nair
• [Asterisk-Users] Queue() with timeout=0   Dinesh Nair
• [Asterisk-Users] Welltech USA? and Wellgate Products?   Dinesh Nair
• [Asterisk-Users] attended call transfer   Dinesh Nair
• [Asterisk-Users] Bridge Calls with G()   Dinesh Nair
• [Asterisk-Users] Anyway to pass CIC in sip header   Dinesh Nair
• [Asterisk-Users] SOLVED - Channel bank woes - no outbound calls   Dinesh Nair
• [Asterisk-Users] Re: Bridge Calls with G()   Dinesh Nair
• [Asterisk-Users] Asterisk Follow Me   Dinesh Nair
• SV: [Asterisk-Users] Problems with voicemail   Dinesh Nair
• [Asterisk-Users] mpg123 alternative?   Dinesh Nair
• [Asterisk-Users] Asterisk Follow Me   Dinesh Nair
• [Asterisk-Users] Monitor a call in progress.   Dinesh Nair
• [Asterisk-Users] virtual extension per user ?   Rajeev Natarajan
• [Asterisk-Users] ChanIsAvail   Jayson Navitsky
• [Asterisk-Users] ChanIsAvail   Jayson Navitsky
• [Asterisk-Users] Unknown RTP codec 100 received   Nedi
• [Asterisk-Users] voipstunt can't get call in asterisk   Nedi
• [Asterisk-Users] Asterisk as MGCP User Agent   Stewart Nelson
• [Asterisk-Users] No Ringing Sound & No periodic-announce   Rene at Just For You Networks
• [Asterisk-Users] Ringing Delay   chan (Alpha Trilogies Networls)
• [Asterisk-Users] Ringing Delay   chan (Alpha Trilogies Networls)
• [Asterisk-Users] touch tones too fast ?   Eldon Neustaeter
• [Asterisk-Users] touch tones too fast ?   Eldon Neustaeter
• [Asterisk-Users] touch tones too fast ?   Eldon Neustaeter
• [Asterisk-Users] Alcatel 4200 series pbx   Igor Neves
• [Asterisk-Users] Cisco 7960 won't register   Mike Newton
• [Asterisk-Users] Cisco 7960 won't register   Mike Newton
• [Asterisk-Users] RE: Cisco 7905 can't register   Mike Newton
• [Asterisk-Users] forward to gateway   Nhadie
• [Asterisk-Users] error on AMP route   Nhadie
• [Asterisk-Users] 911 and ISDN PRI   Darren Nickerson
• [Asterisk-Users] Sirrix PC140 Quad card   Marnus van Niekerk
• [Asterisk-Users] touch tones too fast ?   John Novack
• [Asterisk-Users] attended call transfer   John Novack
• [Asterisk-Users] PrivacyManager Broken?   John Novack
• [Asterisk-Users] international calling via POTS in Russia   John Novack
• [Asterisk-Users] is there a web interface to this mailing list?   John Novack
• [Asterisk-Users] Bridged line appearance   John Novack
• [Asterisk-Users] Sendmail with exchange   Jordan Novak
• [Asterisk-Users] Virtual Extensions   Jordan Novak
• [Asterisk-Users] Conference bridge dimensioning   Jordan Novak
• [Asterisk-Users] Leading 0 on caller ID with internal S0 (HFC)   Pedro Nunes
• [Asterisk-Users] ZAP Digit Timeout   Jarrod O'Connell
• [Asterisk-Users] Remapping Polycom IP501 buttons   Matthew T. O'Connor
• [Asterisk-Users] Re: Remapping Polycom IP501 buttons   Matthew T. O'Connor
• [Asterisk-Users] Re: Remapping Polycom IP501 buttons   Matthew T. O'Connor
• [Asterisk-Users] FW: auto provision of IP501 polycom   Matthew T. O'Connor
• [Asterisk-Users] Ottawa Asterisk Users Group   Richard OSS
• [Asterisk-Users] co-location providers in Ottawa, Canada   Richard OSS
• [Asterisk-Users] co-location providers in Ottawa, Canada   Richard OSS
• [Asterisk-Users] Dell PowerEdge 2850   Richard OSS
• [Asterisk-Users] Dell PowerEdge 2850   Richard OSS
• [Asterisk-Users] Voice conferencing server capacity   Richard OSS
• [Asterisk-Users] Voice conferencing server capacity   Richard OSS
• [Asterisk-Users] Conference bridge dimensioning   Richard OSS
• [Asterisk-Users] How to check if transcoding is setup to work properly   Obelix
• [Asterisk-Users] Hunting for DIDs in Kenya/Nigeria   Alphonse Ogulla
• [Asterisk-Users] (newby) IAX Trunk on low bandwidth connection   Greg Oliver
• [Asterisk-Users] OT- Rwanda DSL growth   Greg Oliver
• [Asterisk-Users] OT: VoIP over bonded link   Greg Oliver
• [Asterisk-Users] asterisk 1.2.4,addons 1.2.1, ooh323 and freebsd   Olivier.taylor
• [Asterisk-Users] Asterisk 1.2.x + oh323 on Debian Sarge.   Olivier.taylor
• [Asterisk-Users] Dialing part of the extension   Olivier.taylor
• [Asterisk-Users] How to Get SIP Header : To Field ?   Olivier.taylor
• [Asterisk-Users] Oh323, opengk and asterisk   Olivier.taylor
• [Asterisk-Users] lists problem, Gmail????????   Olivier.taylor
• [Asterisk-Users] Beer meeting at Fosdem   Olivier.taylor
• [Asterisk-Users] Cheapest provider for Philippine route   Olivier.taylor
• [Asterisk-Users] changing cisco 7940/7960 standard menus ?   Alex Ongena
• [Asterisk-Users] virtual extension per user ?   Alex Ongena
• [Asterisk-Users] virtual extension per user ?   Alex Ongena
• [Asterisk-Users] virtual extension per user ?   Alex Ongena
• [Asterisk-Users] Sip + Cisco 7940/7960 + Panel + DND + queues   Alex Ongena
• Using *RT for HA purposes was: [Asterisk-Users]RealtimeMultipleAsterisk boxes, iaxusers   Chad Osmond
• [Asterisk-Users] Asterisk vs. Traditional PBX   Chad Osmond
• [Asterisk-Users] Caller stuck in MoH after being answered by a phone that was forwarded to.   Chad Osmond
• [Asterisk-Users] co-location providers in Ottawa, Canada   Chad Osmond
• [Asterisk-Users] Sip One way audio   Paul Oster
• [Asterisk-Users] How to handle "provider UNREACHABLE" in the dialplan?   Florian Overkamp
• [Asterisk-Users] How to handle "provider UNREACHABLE" in the dialplan?   Florian Overkamp
• [Asterisk-Users] Codec negotiation   Florian Overkamp
• [Asterisk-Users] Codec negotiation   Florian Overkamp
• [Asterisk-Users] Set CallerIDNum on a PRI   Florian Overkamp
• [Asterisk-Users] Firmware version 1.3.1 released for Aastra IP phones   Gareth Owen
• [Asterisk-Users] RE: aastra v1.3.1 firmware   Gareth Owen
• [Asterisk-Users] RE: Firmware version 1.3.1 released for Aastra IP phones   Gareth Owen
• [Asterisk-Users] RE: Firmware version 1.3.1 released for AastraIPphones   Gareth Owen
• [Asterisk-Users] Asterisk Registering with SER question   Ryan Pagquil
• [Asterisk-Users] Echo PSTN Asterisk@Home 2.0 Digium TDM11B & DSL   David K Parker
• [Asterisk-Users] Echo PSTN Asterisk@Home 2.0 Digium TDM11B & DSL   David K Parker
• [Asterisk-Users] Good VoIP providers that support Asterisk PBX's   David K Parker
• [Asterisk-Users] help on dial plan   Patrick
• [Asterisk-Users] Grandstream GXP-2000   Patrick
• [Asterisk-Users] Linksys rt31p2   Thomas Patterson
• [Asterisk-Users] Anyway to do this?   Paul
• [Asterisk-Users] Anyway to do this?   Paul
• [Asterisk-Users] NuFone   Paul
• [Asterisk-Users] Fedora Core 3 or Fedora Core 4? yum update ornot?   Paul
• [Asterisk-Users] Fedora Core 3 or Fedora Core 4? yum update ornot?   Paul
• [Asterisk-Users] What ATA should I buy?   Paul
• [Asterisk-Users] is there a web interface to this mailing list?   Paul
• [Asterisk-Users] Analyzer for Milliwatt   Paul
• [Asterisk-Users] What business IP phone to use   Paul
• [Asterisk-Users] Analyzer for Milliwatt   Paul
• [Asterisk-Users] Is Asterisk a PBX?   Paul
• [Asterisk-Users] Asterisk Web-Based Voicemail?   Paul
• [Asterisk-Users] VOIP provider iristel, setup account   Cristian Paun
• [Asterisk-Users] Outbound problem sip chanel   Cristian Paun
• [Asterisk-Users] Strata DK280 + Asterisk@Home   Chris Peacock
• [Asterisk-Users] No Audio on Local Machine, Remote works fine   Hadar Pedhazur
• [Asterisk-Users] No Audio on Local Machine, Remote works fine   Hadar Pedhazur
• [Asterisk-Users] problems while dailing outside   David Peer
• [Asterisk-Users] Deploying VoIP on a WAN   Joao Pereira
• [Asterisk-Users] Asterisk on Solaris 10 (AMD Opteron, SunFire X2100)   Roberto Pereyra
• [Asterisk-Users] attended call transfer   Richard Perini
• [Asterisk-Users] Re: asterisk logger - urgent!!!   Olivier Perrin
• [Asterisk-Users] Obtaining billsecs in the dialplan after a call?   Olivier Perrin
• [Asterisk-Users] Generating Polarity Reversal on FXS line   Infobox Peru
• [Asterisk-Users] How to determine the power draw on TDM2400P?   Infobox Peru
• [Asterisk-Users] Hold and Call Waiting - Budgetone 100   Dan Peters
• [Asterisk-Users] Hold and Call Waiting - Budgetone 100   Dan Peters
• [Asterisk-Users] MOH sourced from a sound card?   Mark Phillips
• [Asterisk-Users] MOH sourced from a sound card?   Mark Phillips
• [Asterisk-Users] OT O'Reilly Asterisk TFOT   Mark Phillips
• [Asterisk-Users] OT O'Reilly Asterisk TFOT   Mark Phillips
• [Asterisk-Users] change languages from an IVR   Mark Phillips
• [Asterisk-Users] 911 and ISDN PRI   Mark Phillips
• [Asterisk-Users] Asterisk native sounds now available!   Mark Phillips
• [Asterisk-Users] asterisk to FWD   Mark Phillips
• [Asterisk-Users] Asterisk native sounds now available!   Mark Phillips
• [Asterisk-Users] Better i18n for Asterisk?   Mark Phillips
• [Asterisk-Users] change languages from an IVR   Mark Phillips
• [Asterisk-Users] change languages from an IVR   Mark Phillips
• [Asterisk-Users] Free IAX login   Mark Phillips
• [Asterisk-Users] asterisk to FWD   Mark Phillips
• [Asterisk-Users] change languages from an IVR   Mark Phillips
• [Asterisk-Users] Asterisk native sounds now available!   Mark Phillips
• [Asterisk-Users] Asterisk native sounds now available!   Mark Phillips
• [Asterisk-Users] Asterisk native sounds now available!   Mark Phillips
• [Asterisk-Users] Asterisk native sounds now available!   Mark Phillips
• [Asterisk-Users] How do I install speex for asterisk?   Mark Phillips
• [Asterisk-Users] How do I install speex for asterisk?   Mark Phillips
• [Asterisk-Users] How do I install speex for asterisk?   Mark Phillips
• [Asterisk-Users] How can I force Asterisk t not override my codec order?   Mark Phillips
• [Asterisk-Users] double ringing tone on asterisk 1.2 (workaround)   Matteo Piazza
• [Asterisk-Users] SPA-941 & hint   Matteo Piazza
• [Asterisk-Users] Hangup Detection   Aaron Picht
• [Asterisk-Users] (newby) Is PING a good indicator of latency?   Wilson Pickett
• [Asterisk-Users] OT O'Reilly Asterisk TFOT   Wilson Pickett
• [Asterisk-Users] Re: delaying "answer" for a number of ringsor an amount of time   Wilson Pickett
• [Asterisk-Users] TDM400 and Phone does not 'ring'   Wilson Pickett
• [Asterisk-Users] g729 license question   Wilson Pickett
• [Asterisk-Users] Asterisk native sounds now available!   Wilson Pickett
• [Asterisk-Users] Codec issue with my iaxy   Wilson Pickett
• [Asterisk-Users] Codec issue with my iaxy   Wilson Pickett
• [Asterisk-Users] Re: Polycom boot times/XML files.   Wilson Pickett
• [Asterisk-Users] Polycom Default Ring Volume   Wilson Pickett
• [Asterisk-Users]how to send fax using Spandsp   Bartosz Piec
• [Asterisk-Users] Receiving faxes with spandsp - strange problem   Bartosz Piec
• [Asterisk-Users] Receiving faxes with spandsp - strange problem   Bartosz Piec
• [Asterisk-Users] Receiving faxes with spandsp - strange problem   Bartosz Piec
• [Asterisk-Users] Receiving faxes with spandsp - strange problem   Bartosz Piec
• [Asterisk-Users] Receiving faxes with spandsp - strange problem   Bartosz Piec
• [Asterisk-Users] Receiving faxes with spandsp - strange problem   Bartosz Piec
• [Asterisk-Users] Broken faxes when other call disconnects   Bartosz Piec
• [Asterisk-Users] voicemail recording format   Bartosz Piec
• [Asterisk-Users] Problem with receiving faxes with spandsp - full log included (long)   Bartosz Piec
• [Asterisk-Users] spandsp debug or logging   Bartosz Piec
• [Asterisk-Users] How can I debug spandsp?   Bartosz Piec
• [Asterisk-Users] Zaptel 1.2.4 Released!   Pikoro
• [Asterisk-Users] Realtime queues with Firebird SQL through unixodbc   Eric Piros
• [Asterisk-Users] Incoming ISDN DATA calls answered by asterisk IVR! - How to stop that?   Pisac
• [Asterisk-Users] Incoming ISDN DATA calls answered by asterisk IVR! - How to stop that?   Pisac
• [Asterisk-Users] DATA calls answered by IVR, but I don't want that   Pisac
• [Asterisk-Users] why incoming DATA CALLS are answered as VOICE by asterisk IVR?   Pisac
• [Asterisk-Users] Voicemail 0 for operator call routing   Don Pobanz
• [Asterisk-Users] Intro and first questions   Tom Poe
• [Asterisk-Users] FC4 and yum install; how to configure questions   Tom Poe
• [Asterisk-Users] Voicemail Problem   Mike Pollitt
• [Asterisk-Users] odd 'digital' sound artifacts   Mike Pollitt
• [Asterisk-Users] No Voice when canreinvite=no   Mike Pollitt
• [Asterisk-Users] TE411P Really Bad Echo   Mike Pollitt
• [Asterisk-Users] sip expire 60   Mike Pollitt
• [Asterisk-Users] Polycom buddy watch limit of 7   Mike Pollitt
• [Asterisk-Users] Speex echo cancellation   Mike Pollitt
• Looping through variables, or sort of but not really arrays [was, bizarrely, RE: [Asterisk-Users] is there a web interface to this mailing list?]   Mike Pollitt
• [Asterisk-Users] show calls   Mike Pollitt
• [Asterisk-Users] How do I install speex for asterisk?   Mike Pollitt
• [Asterisk-Users] DID's   Mike Pollitt
• [Asterisk-Users] How do I install speex for asterisk?   Mike Pollitt
• [Asterisk-Users] Call forward on unavailable timer issues   Mike Pollitt
• [Asterisk-Users] Dial timeouts and SIP 302 redirects   Mike Pollitt
• [Asterisk-Users] Asterisk hints   Mike Pollitt
• [Asterisk-Users] Dial timeouts and SIP 302 redirects   Mike Pollitt
• [Asterisk-Users] Asterisk hints   Mike Pollitt
• [Asterisk-Users] username as extension   Mike Pollitt
• [Asterisk-Users] auto provision of IP501 polycom   Mike Pollitt
• [Asterisk-Users] How to query a table from the keypad?   Mike Pollitt
• [Asterisk-Users] meetme announcement   Barry Porch
• [Asterisk-Users] click to talk   Graziano Poretti
• [Asterisk-Users] Zhone channel Banks   Jon Pounder
• [Asterisk-Users] Application Faxing using SIP   J Poz
• [Asterisk-Users] Application Faxing using SIP   J Poz
• [Asterisk-Users] Application Faxing using SIP   J Poz
• [Asterisk-Users] Application Faxing using SIP   J Poz
• [Asterisk-Users] chan_capi and Eicon Diva   Paolo Prandini
• [Asterisk-Users] Fedora Core 3 or Fedora Core 4? yum update or not?   Russ Price
• [Asterisk-Users] Channel bleedover?   Paul A. Pringle
• [Asterisk-Users] (newby) IAX Trunk on low bandwidth connection   Cosmin Prund
• [Asterisk-Users] (newby) EURO-ISDN line question   Cosmin Prund
• [Asterisk-Users] (newby) Is PING a good indicator of latency?   Cosmin Prund
• [Asterisk-Users] (newby) EURO-ISDN line question   Cosmin Prund
• [Asterisk-Users] (newby) Is PING a good indicator of latency?   Cosmin Prund
• [Asterisk-Users] (newby) IAX Trunk on low bandwidth connection   Cosmin Prund
• [Asterisk-Users] (newby) Is PING a good indicator of latency?   Cosmin Prund
• [Asterisk-Users] (newby) IAX Trunk on low bandwidth connection   Cosmin Prund
• [Asterisk-Users] (newby) IAX Trunk on low bandwidth connection   Cosmin Prund
• [Asterisk-Users] (newby) IAX Trunk on low bandwidth connection   Cosmin Prund
• [Asterisk-Users] (newby) Asterisk on the open internet & security   Cosmin Prund
• [Asterisk-Users] (newby) Asterisk on the open internet & security   Cosmin Prund
• [Asterisk-Users] (newby) Asterisk on the open internet & security   Cosmin Prund
• [Asterisk-Users] Asterisk with USB   Cosmin Prund
• [Asterisk-Users] Problem with CLI output on Asterisk@Home   Cosmin Prund
• [Asterisk-Users] Help with dialplan   Cosmin Prund
• [Asterisk-Users] Dialing part of the extension   Cosmin Prund
• [Asterisk-Users] Dialing part of the extension   Cosmin Prund
• [Asterisk-Users] FXO port on TDM400P hangs   Cosmin Prund
• [Asterisk-Users] FXO port on TDM400P hangs!!   Cosmin Prund
• [Asterisk-Users] FXO port on TDM400P hangs!!   Cosmin Prund
• [Asterisk-Users] FXO port on TDM400P hangs!!   Cosmin Prund
• [Asterisk-Users] FXO port on TDM400P hangs!!   Cosmin Prund
• [Asterisk-Users] Intro and first questions   Cosmin Prund
• [Asterisk-Users] Detecting disconnect on TDM400P with 3 FXO ports and 1 FXS port   Cosmin Prund
• [Asterisk-Users] fax receive using TDM400P   Cosmin Prund
• [Asterisk-Users] fax receive using TDM400P   Cosmin Prund
• [Asterisk-Users] fax receive using TDM400P   Cosmin Prund
• [Asterisk-Users] transferring 3000 SIP calls   Cosmin Prund
• [Asterisk-Users] fax receive using TDM400P   Cosmin Prund
• [Asterisk-Users] Blocked Callerid   Joe Pukepail
• [Asterisk-Users] New version of snom soft phone   Joe Pukepail
• [Asterisk-Users] 911 and ISDN PRI   Joe Pukepail
• [Asterisk-Users] 911 and ISDN PRI   Joe Pukepail
• [Asterisk-Users] 911 and ISDN PRI   Joe Pukepail
• [Asterisk-Users] Traffic prioritization and 'class of service' for SIP   Joe Pukepail
• [Asterisk-Users] [Announce] Web-MeetMe v2.0.0   Joe Pukepail
• [Asterisk-Users] [Announce] Web-MeetMe v2.0.0   Joe Pukepail
• [Asterisk-Users] What business IP phone to use   Joe Pukepail
• [Asterisk-Users] need help   Dirgan Putra
• [Asterisk-Users] [Announce] Web-MeetMe v2.0.0   Ben Q
• [Asterisk-Users] [Announce] Web-MeetMe v2.0.0   Ben Q
• [Asterisk-Users] [Announce] Web-MeetMe v2.0.0   Ben Q
• [Asterisk-Users] Re: CallerID Problem   Bromont Quebec
• [Asterisk-Users] Re: delaying "answer" for a number of rings or an amount of time   Bromont Quebec
• [Asterisk-Users] Re: delaying "answer" for a number of rings or an amount   Bromont Quebec
• [Asterisk-Users] Re: Sipura SPA-2002 rings randomly   Bromont Quebec
• [Asterisk-Users] Re: delaying "answer" for a number of ring or an amount of time   Bromont Quebec
• [Asterisk-Users] Re: Will not authenticate incoming VOIP provider   Bromont Quebec
• [Asterisk-Users] Re: Why is asterisk ignoring my context?   Bromont Quebec
• [Asterisk-Users] Asterisk at SCALE 4x   Ilan Rabinovitch
• [Asterisk-Users] dtmfmode=auto, but doesn't work   Aryanto Rachmad
• [Asterisk-Users] Zap, Caller ID problem   Aryanto Rachmad
• [Asterisk-Users] dtmfmode=auto, but doesn't work   Aryanto Rachmad
• [Asterisk-Users] Fedora Core 3 or Fedora Core 4? yum update or not?   RandyW
• [Asterisk-Users] Asterisk-Netsec Ranch Networks   Vikram Rangnekar
• [Asterisk-Users] Problems mixing audio in queues and playing queue positions   Faris Raouf
• [Asterisk-Users] chan_capi-cm 0.6.4 random outgoing MSN problem   Faris Raouf
• [Asterisk-Users] chan_capi-cm 0.6.4 random outgoing MSN problem   Faris Raouf
• [Asterisk-Users] mpg123 alternative?   Faris Raouf
• [Asterisk-Users] chan_capi-cm 0.6.4 random outgoing MSN problem   Faris Raouf
• [Asterisk-Users] mpg123 alternative?   Faris Raouf
• [Asterisk-Users] SIP Problem Fedora Core 4 and Asterisk 1.2.4   Abhimanyu Rapria
• [Asterisk-Users] Codec issue with my iaxy   Mark Ratering
• [Asterisk-Users] Codec issue with my iaxy   Mark Ratering
• [Asterisk-Users] Agents, queues and zombies   Steve Rawlings
• [Asterisk-Users] Fw: Agents, queues and zombies   Steve Rawlings
• [Asterisk-Users] PRI Bridging and Recording   Steve Rawlings
• [Asterisk-Users] (newby) IAX Trunk on low bandwidth connection   Jolly M. Recto
• [Asterisk-Users] attended call transfer   Paul Redstone
• [Asterisk-Users] What ATA should I buy?   Tele Cost Price Reducer
• [Asterisk-Users] Best ATA for general residential deployment??   Tele Cost Price Reducer
• [Asterisk-Users] Configure DID   Tele Cost Price Reducer
• [Asterisk-Users] Codec order sent wrong from Asterisk   Tele Cost Price Reducer
• [Asterisk-Users] Asterisk and Hipath interconnections   Tele Cost Price Reducer
• [Asterisk-Users] Cheapest provider for Philippine route   Tele Cost Price Reducer
• [Asterisk-Users] sip to oh323 converter converts sip uri to h.323 number and not h.323 url   Oliver Rehak
• [Asterisk-Users] sip to oh323 converter converts sip uri to h.323 number and not h.323 url   Oliver Rehak
• [Asterisk-Users] setup default console debug levels   Oliver Rehak
• [Asterisk-Users] Dell PowerEdge 2850   Darren Reilly
• [Asterisk-Users] Cisco 2620 as PRI gateway   Tim Reimers
• [Asterisk-Users] Cisco 2620 as PRI gateway   Tim Reimers
• [Asterisk-Users] Fedora Core 3 or Fedora Core 4? yum update ornot?   Tim Reimers
• [Asterisk-Users] Cisco 2620 as PRI gateway   Tim Reimers
• [Asterisk-Users] Cisco 2620 as PRI gateway   Tim Reimers
• [Asterisk-Users] Cisco 2620 as PRI gateway   Tim Reimers
• [Asterisk-Users] Cisco 2620 as PRI gateway   Tim Reimers
• [Asterisk-Users] How to query a table from the keypad?   Richard Reina
• [Asterisk-Users] How to query a table from the keypad?   Richard Reina
• [Asterisk-Users] sip channel status - how?   Stefan Reuter
• [Asterisk-Users] Teliax - Codec Preference effective?   John Reynolds
• [Asterisk-Users] Call over SIP channel becomes a zombie   Stephane Ricard
• [Asterisk-Users] Sipura SPA 3000 logic   Hadley Rich
• [Asterisk-Users] Sipura SPA 3000 logic   Hadley Rich
• [Asterisk-Users] How hard to create Asterisk for Compact Flash?   Hadley Rich
• [Asterisk-Users] CallerID Problem   Gary Richardson
• [Asterisk-Users] CallerID Problem   Gary Richardson
• [Asterisk-Users] Re: CallerID Problem   Gary Richardson
• [Asterisk-Users] Re: CallerID Problem   Gary Richardson
• [Asterisk-Users] moh about twice as fast   Gary Richardson
• [Asterisk-Users] Cisco 2620 as PRI gateway   Gary Richardson
• [Asterisk-Users] Cisco 2620 as PRI gateway   Gary Richardson
• [Asterisk-Users] 79xx's and call queues   Gary Richardson
• [Asterisk-Users] Call queue design issues and suggestions   Gary Richardson
• [Asterisk-Users] Re: Call queue design issues and suggestions   Gary Richardson
• [Asterisk-Users] ImportVar Syntax   Steven Ringwald
• [Asterisk-Users] any doc/example for app_sms.so ?   Luigi Rizzo
• [Asterisk-Users] Repeating Zap Message   Adam Robins
• [Asterisk-Users] Asterisk 1.2.4 Quality Issues   Adam Robins
• [Asterisk-Users] Asterisk 1.2.4 IAX2 New Jitterbuffer Tuning   Adam Robins
• [Asterisk-Users] Asterisk 1.2.4 IAX2 New Jitterbuffer Tuning   Adam Robins
• [Asterisk-Users] Asterisk 1.2.4 IAX2 New Jitterbuffer Tuning   Adam Robins
• [Asterisk-Users] Asterisk 1.2.4 IAX2 New Jitterbuffer Tuning   Adam Robins
• [Asterisk-Users] Asterisk 1.2.4 IAX2 New Jitterbuffer Tuning   Adam Robins
• [Asterisk-Users] Asterisk 1.2.4 IAX2 New Jitterbuffer Tuning   Adam Robins
• [Asterisk-Users] Asterisk 1.2.4 IAX2 New Jitterbuffer Tuning   Adam Robins
• [Asterisk-Users] Asterisk 1.2.4 IAX2 New Jitterbuffer Tuning   Adam Robins
• [Asterisk-Users] Asterisk 1.2.4 IAX2 New Jitterbuffer Tuning   Adam Robins
• [Asterisk-Users] Re: Asterisk 1.2.4 IAX2 New Jitterbuffer Tuning   Adam Robins
• [Asterisk-Users] Asterisk compile error   Adam Robins
• [Asterisk-Users] Good VoIP providers that support Asterisk PBX's   Jim Robinson
• [Asterisk-Users] Corrupt CDR records in Asterisk 1.2.x   Tim Robinson
• [Asterisk-Users] UK X100P installation help   Tim Robinson
• [Asterisk-Users] UK X100P installation help   Tim Robinson
• [Asterisk-Users] ISDN30E + T1 crossover cable woes   Tim Robinson
• [Asterisk-Users] Asterisk configuration for h323 calls   Aing Roda
• [Asterisk-Users] SIP and firewalls?   Hagen Rode
• [Asterisk-Users] A good SIP VB6.0 component to use?   Hagen Rode
• [Asterisk-Users] Polycom IP301: Pass-through ethernet portunusable?   Anthony Rodgers
• [Asterisk-Users] Re: Contents of Asterisk-Users digest...   Anthony Rodgers
• [Asterisk-Users] MWI on Polycom 501.   Anthony Rodgers
• [Asterisk-Users] MWI on Polycom 501.   Anthony Rodgers
• [Asterisk-Users] TE411P Really Bad Echo   Anthony Rodgers
• [Asterisk-Users] Nortel Meridian Opt 81C and PRI   Anthony Rodgers
• [Asterisk-Users] SPA-3000 VOIP-PSTN gateway - long delay between answering and ringing   Anthony Rodgers
• [Asterisk-Users] Polycom IP501 MWI goes off periodically   Anthony Rodgers
• [Asterisk-Users] SPA-3000 VOIP-PSTN gateway - long delay betweenanswering and ringing   Anthony Rodgers
• [Asterisk-Users] Polycom IP501 MWI goes off periodically   Anthony Rodgers
• [Asterisk-Users] odd 'digital' sound artifacts   Anthony Rodgers
• [Asterisk-Users] is there a web interface to this mailing list?   Anthony Rodgers
• [Asterisk-Users] Problem with T1 installation   Anthony Rodgers
• [Asterisk-Users] Help Asterisk with Phoneserve   Carlos Rojas
• [Asterisk-Users] OT O'Reilly Asterisk TFOT   James Ronald
• [Asterisk-Users] Fedora Core 3 or Fedora Core 4? yum update ornot?   Matt Roth
• [Asterisk-Users] Disabling SELinux in FC3 - good or bad   Matt Roth
• [Asterisk-Users] How do I install speex for asterisk?   Matt Roth
• [Asterisk-Users] mpg123 alternative?   Matt Roth
• [Asterisk-Users] Analyzer for Milliwatt   Matt Roth
• [Asterisk-Users] Swapping lines using dtmf   Joseph Rothstein
• [Asterisk-Users] IAX registration expiration   Joseph Rothstein
• [Asterisk-Users] Can I escape queue with a '*'?   Joseph Rothstein
• [Asterisk-Users] Problem with T1 installation   Brian Roy
• [Asterisk-Users] What SW/HW phones support sendtext feature (trying to send speech recognition results back to user)?   Robert Rozman
• [Asterisk-Users] Reading sound in eagi script and recognizing DTMF sounds at thesame time ?   Robert Rozman
• [Asterisk-Users] Asterisk and MOH for Queues   Waldo Rubinstein
• [Asterisk-Users] What business IP phone to use   Waldo Rubinstein
• [Asterisk-Users] Best ATA for general residential deployment??   Marc Rys
• [Asterisk-Users] Queue Messages now playing when caller is inside queue   Rajkumar S
• [Asterisk-Users] Queue Messages not playing when caller is inside queue   Rajkumar S
• [Asterisk-Users] Queue Messages not playing when caller is inside queue   Rajkumar S
• [Asterisk-Users] No sound on 10% of incoming calls   Jerome SOUCANY
• [Asterisk-Users] No sound on 10% of incoming calls   Jerome SOUCANY
• [Asterisk-Users] Busy problem   Jerome SOUCANY
• [Asterisk-Users] Dell PowerEdge 1800 and TE410P   Jerome SOUCANY
• [Asterisk-Users] Linksys WIP300 WiFi Phone   Omar A. Sabek
• [Asterisk-Users] Linksys WIP300 WiFi Phone   Omar A. Sabek
• [Asterisk-Users] OH323 Peer   Alberto Sagredo
• [Asterisk-Users] Limiting Sip Calls ?   Alberto Sagredo
• [Asterisk-Users] Cisco 2620 as PRI gateway   Juan Salas
• [Asterisk-Users] Cisco 2620 as PRI gateway   Juan Salas
• [Asterisk-Users] asterisk silence suppression?   Juan Salas
• [Asterisk-Users] anyway to a2billing without IVR   Asterisk Sales
• [Asterisk-Users] a2billing without IVR   Asterisk Sales
• [Asterisk-Users] How to Get SIP Header : To Field ?   Jean-Marc Salsa
• [Asterisk-Users] SIP Header VIA when behind NAT   Jean-Marc Salsa
• [Asterisk-Users] ARI 0.06   Jean-Marc Salsa
• [Asterisk-Users] ARI 0.06   Jean-Marc Salsa
• [Asterisk-Users] DTMF Mode supported by VoiceMail Application   Jean-Marc Salsa
• [Asterisk-Users] DTMF Mode supported by VoiceMail Application   Jean-Marc Salsa
• [Asterisk-Users] Asterisk hints   Jean-Marc Salsa
• [Asterisk-Users] IP PAX gateway to PSTN   Sam
• [Asterisk-Users] IP PAX gateway to PSTN   Sam
• [Asterisk-Users] IP PAX gateway to PSTN   Sam
• [Asterisk-Users] IP PAX gateway to PSTN   Sam
• [Asterisk-Users] IP PAX gateway to PSTN   Sam
• [Asterisk-Users] IP PAX gateway to PSTN   Sam
• [Asterisk-Users] echo cancel from telco   Michael Sampson
• [Asterisk-Users] What ATA should I buy?   Michael Sampson
• [Asterisk-Users] Automated wake up call   Michael Sampson
• [Asterisk-Users] Automated wake up call   Michael Sampson
• [Asterisk-Users] Random Disconnects - or ARE they?   Michael Sampson
• [Asterisk-Users] Quintum Tenor AX 24 Port SIP FXS "Unsupported Media Type"   Michael Sampson
• [Asterisk-Users] odd 'digital' sound artifacts   Gerard Saraber
• [Asterisk-Users] more cpu intensive echo cancellers ?   Gerard Saraber
• [Asterisk-Users] more cpu intensive echo cancellers ?   Gerard Saraber
• [Asterisk-Users] more cpu intensive echo cancellers ?   Gerard Saraber
• [Asterisk-Users] ztdummy on gentoo 2005.1   Gerard Saraber
• [Asterisk-Users] more cpu intensive echo cancellers ?   Gerard Saraber
• [Asterisk-Users] more cpu intensive echo cancellers ?   Gerard Saraber
• [Asterisk-Users] odd 'digital' sound artifacts   Gerard Saraber
• [Asterisk-Users] more cpu intensive echo cancellers ?   Gerard Saraber
• [Asterisk-Users] odd 'digital' sound artifacts   Gerard Saraber
• [Asterisk-Users] more cpu intensive echo cancellers ?   Gerard Saraber
• [Asterisk-Users] odd 'digital' sound artifacts   Gerard Saraber
• [Asterisk-Users] odd 'digital' sound artifacts [1 card = no artifacts]   Gerard Saraber
• [Asterisk-Users] odd 'digital' sound artifacts [solved]   Gerard Saraber
• [Asterisk-Users] MOH from RCA jack?   Gerard Saraber
• [Asterisk-Users] MOH from RCA jack?   Gerard Saraber
• [Asterisk-Users] A unique 'click to call' project - Could use some advice   Gerard Saraber
• [Asterisk-Users] Multiple TDM400P's in a single machine   Gerard Saraber
• [Asterisk-Users] Cisco 79xx firmware   Gerard Saraber
• [Asterisk-Users] Calls not going through   Gerard Saraber
• [Asterisk-Users] Cisco 79xx and SIP 7.5 Problems   Gerard Saraber
• [Asterisk-Users] segmentation fault   Sascha
• [Asterisk-Users] RE: Euro-ISDN   Armin Schindler
• [Asterisk-Users] RE: Euro-ISDN   Armin Schindler
• [Asterisk-Users] ISDN Eicon Diva Server V-BRI   Armin Schindler
• [Asterisk-Users] ISDN Eicon Diva Server V-BRI   Armin Schindler
• [Asterisk-Users] ISDN Eicon Diva Server V-BRI   Armin Schindler
• [Asterisk-Users] chan_sccp availability?   Armin Schindler
• [Asterisk-Users] Routing Calls via chan_capi with AVM FritzCard   Armin Schindler
• [Asterisk-Users] Routing Calls via chan_capi with AVM FritzCard   Armin Schindler
• [Asterisk-Users] FXS with v.90 modem support?   Armin Schindler
• [Asterisk-Users] stable ISDN BRI card for asterisk   Armin Schindler
• [Asterisk-Users] Problem with Wait() and chan_capi-cm?   Armin Schindler
• [Asterisk-Users] Problem with Wait() and chan_capi-cm?   Armin Schindler
• [Asterisk-Users] chan_capi setting ${DNIS}   Armin Schindler
• [Asterisk-Users] Setting up an EICON CARD with CAPI   Armin Schindler
• [Asterisk-Users] Re: chan_capi-cm-0.6.4   Armin Schindler
• [Asterisk-Users] chan_capi-cm 0.6.4 random outgoing MSN problem   Armin Schindler
• [Asterisk-Users] Re: chan_capi-cm-0.6.4   Armin Schindler
• [Asterisk-Users] Problem with chan-capi: outgoing calls on two lines   Armin Schindler
• [Asterisk-Users] Problem with chan-capi: outgoing calls on two lines   Armin Schindler
• [Asterisk-Users] Re: chan_capi-cm-0.6.4   Armin Schindler
• [Asterisk-Users] chan_capi and Eicon Diva   Armin Schindler
• [Asterisk-Users] Re: chan_capi-cm-0.6.4   Armin Schindler
• [Asterisk-Users] Re: chan_capi-cm-0.6.4   Armin Schindler
• [Asterisk-Users] Re: chan_capi-cm-0.6.4   Armin Schindler
• [Asterisk-Users] Austria isdn p2p empty DID   Armin Schindler
• [Asterisk-Users] Re: chan_capi-cm-0.6.4   Armin Schindler
• [Asterisk-Users] chan_capi-cm-0.6.4   Ralf Schlatterbeck
• [Asterisk-Users] Re: chan_capi-cm-0.6.4   Ralf Schlatterbeck
• [Asterisk-Users] Re: chan_capi-cm-0.6.4   Ralf Schlatterbeck
• [Asterisk-Users] Re: chan_capi-cm-0.6.4   Ralf Schlatterbeck
• [Asterisk-Users] Re: chan_capi-cm-0.6.4   Ralf Schlatterbeck
• [Asterisk-Users] Q: Status of feature Call Deflection / Partial Rerouing in chan-capi and zaphfc   Ralf Schlatterbeck
• [Asterisk-Users] Re: chan_capi-cm-0.6.4   Ralf Schlatterbeck
• [Asterisk-Users] Re: chan_capi-cm-0.6.4   Ralf Schlatterbeck
• [Asterisk-Users] Austria isdn p2p empty DID   Ralf Schlatterbeck
• [Asterisk-Users] Re: chan_capi-cm-0.6.4   Ralf Schlatterbeck
• [Asterisk-Users] Re: chan_capi-cm-0.6.4   Ralf Schlatterbeck
• [Asterisk-Users] Routing Calls via chan_capi with AVM FritzCard   Christian Schmidt
• [Asterisk-Users] Routing Calls via chan_capi with AVM FritzCard   Christian Schmidt
• [Asterisk-Users] Routing Calls via chan_capi with AVM FritzCard   Christian Schmidt
• [Solved] [Asterisk-Users] Routing Calls via chan_capi with AVM FritzCard   Christian Schmidt
• [Solved] [Asterisk-Users] Routing Calls via chan_capi with AVM FritzCard   Christian Schmidt
• [Asterisk-Users] Fromstring when sending e-mail on recievedvoicemail   Christian Schmidt
• [Asterisk-Users] Routing Calls via chan_capi with AVM FritzCard   Peer Oliver Schmidt
• [Asterisk-Users] Fritz card technology & German *   Peer Oliver Schmidt
• [Asterisk-Users] 1 ISDN BRI to IAX2/SIP... (*) best tool or?...   Peer Oliver Schmidt
• [Asterisk-Users] Application Faxing using SIP   Peer Oliver Schmidt
• [Asterisk-Users] Grandstream BT-101 POS Error   Peer Oliver Schmidt
• [Asterisk-Users] Cisco 2620 as PRI gateway   Schochet, Wes
• [Asterisk-Users] Waiting for your help...   Schochet, Wes
• [Asterisk-Users] chan_sccp .conf changes   Schochet, Wes
• [Asterisk-Users] Cisco 7960 dialing trouble   Schochet, Wes
• [Asterisk-Users] MWI   Schochet, Wes
• [Asterisk-Users] 7960-tones.xml   Schochet, Wes
• [Asterisk-Users] How to reset Digium card while asterisk is running?   Roger Schreiter
• [Asterisk-Users] Analyzer for Milliwatt   Roger Schreiter
• [Asterisk-Users] Analyzer for Milliwatt   Roger Schreiter
• [Asterisk-Users] Matching '*'   Roger Schreiter
• [Asterisk-Users] AGI Channel Status   Roger Schreiter
• [Asterisk-Users] AGI Scripts Terminate too Soon   Roger Schreiter
• [Asterisk-Users] ArtDio gateways   Richard Schroeder
• [Asterisk-Users] What ATA should I buy?   Richard Schroeder
• [Asterisk-Users] Optimizing Linux to run Asterisk   Matt Schulte
• [Asterisk-Users] Performance differences 64-bit vs 32-bit   Richard Scobie
• [Asterisk-Users] Events when the target answer   Ezequiel A. Sculli
• [Asterisk-Users] Events when the target of the call answer   Ezequiel A. Sculli
• [Asterisk-Users] Events when the target of the call   Ezequiel A. Sculli
• [Asterisk-Users] Is anyone using hinting?   Stephan Seitz
• [Asterisk-Users] sipdiscount   Roshan Sembacuttiaratchy
• [Asterisk-Users] Do we need a QOS switch ?   Ron Senykoff
• [Asterisk-Users] RE: Teliax - Codec Preference effective?   Ron Senykoff
• [Asterisk-Users] What ATA should I buy?   Ron Senykoff
• [Asterisk-Users] Solution for 1 time blast of 200, 000 recorded calls   Ron Senykoff
• [Asterisk-Users] Solution for 1 time blast of 200, 000 recorded calls   Ron Senykoff
• [Asterisk-Users] Solution for 1 time blast of 200, 000 recorded calls   Ron Senykoff
• [Asterisk-Users] Solution for 1 time blast of 200, 000 recorded calls   Ron Senykoff
• [Asterisk-Users] automatically detecting failed registration   Ron Senykoff
• [Asterisk-Users] Call AGI when agent answers call in queue... ?   Ron Senykoff
• [Asterisk-Users] Polycom IP601 Question   Ron Senykoff
• [Asterisk-Users] Re: Asterisk 1.2.4 IAX2 New Jitterbuffer Tuning   Ron Senykoff
• [Asterisk-Users] Re: delaying "answer" for a number of ringsor an amount   Gonzalo Servat
• [Asterisk-Users] Re: delaying "answer" for a number of ringsor an amount   Gonzalo Servat
• [Asterisk-Users] change languages from an IVR   Gonzalo Servat
• [Asterisk-Users] GotoIf number exists in file. How can i do this?   Gonzalo Servat
• [Asterisk-Users] Best ATA for general residential deployment??   Gonzalo Servat
• [Asterisk-Users] Best ATA for general residential deployment??   Gonzalo Servat
• [Asterisk-Users] Best ATA for general residential deployment??   Gonzalo Servat
• [Asterisk-Users] SPA-941 Selective DND   Gonzalo Servat
• [Asterisk-Users] Need to retrieve Call-ID from dialed SIP channel (w/o CDRs)   Darren Sessions
• [Asterisk-Users] Re: Need to retrieve Call-ID from dialed SIP channel (w/o CDRs)   Darren Sessions
• [Asterisk-Users] Re: Need to retrieve Call-ID from dialed number   Darren Sessions
• Using *RT for HA purposes was: [Asterisk-Users]Realtime MultipleAsterisk boxes, iaxusers   Rusty Shackleford
• [Asterisk-Users] Asterisk video conference   Rusty Shackleford
• [Asterisk-Users] Do we need a QOS switch ?   Rusty Shackleford
• [Asterisk-Users] Solution for 1 time blast of 200, 000 recorded calls   Rusty Shackleford
• [Asterisk-Users] g729 quality at GSM bitrates   Rusty Shackleford
• [Asterisk-Users] RE: X100P help required   Tejas Shah
• [Asterisk-Users] Configuring Meeting Room from Asterisk Manager API   Somesh S Shanbhag
• [Asterisk-Users] Re: Configuring Meeting Room from Asterisk Manager API   Somesh S Shanbhag
• [Asterisk-Users] Re: Configuring Meeting Room from Asterisk Manager API   Somesh S Shanbhag
• [Asterisk-Users] MeetMe - Party's are not exchanging Audio - Is this BUG?   Somesh S Shanbhag
• [Asterisk-Users] Re: MeetMe - Party's are not exchanging Audio - Is this BUG?   Somesh S Shanbhag
• [Asterisk-Users] Manager cmd: originate without picking up the fone?!   Ravi Shankar
• [Asterisk-Users] Asterisk and SER   Sharon
• [Asterisk-Users] SER ,Asterisk and MWI   Sharon
• [Asterisk-Users] more cpu intensive echo cancellers ?   Clint Sharp
• [Asterisk-Users] Grandstream GXP-2000   Clint Sharp
• [Asterisk-Users] Grandstream GXP-2000   Clint Sharp
• [Asterisk-Users] Bridged line appearance   Clint Sharp
• [Asterisk-Users] Asterisk on SPA-3000 rings extensions only after third ring   Clint Sharp
• [Asterisk-Users] What business IP phone to use   Clint Sharp
• [Asterisk-Users] What business IP phone to use   Clint Sharp
• [Asterisk-Users] TE411P Really Bad Echo   Stagg Shelton
• [Asterisk-Users] TE411P Really Bad Echo   Stagg Shelton
• [Asterisk-Users] TE411P Really Bad Echo   Stagg Shelton
• [Asterisk-Users] TE411P Really Bad Echo   Stagg Shelton
• [Asterisk-Users] TE411P Really Bad Echo   Stagg Shelton
• [Asterisk-Users] TE411P Really Bad Echo   Stagg Shelton
• [Asterisk-Users] TE411P Really Bad Echo   Stagg Shelton
• [Asterisk-Users] TE411P Really Bad Echo   Stagg Shelton
• [Asterisk-Users] TE411P Really Bad Echo   Stagg Shelton
• [Asterisk-Users] Re: asterisk t.38 pass   Stagg Shelton
• [Asterisk-Users] Recommended rack-mountable server anyone?   Stagg Shelton
• [Asterisk-Users] Recommended rack-mountable server anyone?   Stagg Shelton
• [Asterisk-Users] Re: auto provision of IP501 polycom   Stagg Shelton
• [Asterisk-Users] Voice Over WiFi   Stagg Shelton
• [Asterisk-Users] Some articles   Shidan
• [Asterisk-Users] Re: Some articles   Shidan
• [Asterisk-Users] SPA-941 stutter tone   Jock W. Shirey
• [Asterisk-Users] Fwd: Asterisk on Solaris 10 (AMD Opteron, Sun Fire X2100)   Andreas Sikkema
• [Asterisk-Users] sniffing sip password/uri/host info   Andreas Sikkema
• [Asterisk-Users] Cisco 7960 (SIP) with Asterisk: how to get # towork during a call   Mahilal Silva
• [Asterisk-Users] Regarding cdr_manager.conf   Moises Silva
• [Asterisk-Users] return code from AGI   Moises Silva
• [Asterisk-Users] Fast AGI performance question   Moises Silva
• [Asterisk-Users] attended call transfer   Moises Silva
• [Asterisk-Users] How can I send DTMF from the console?   Moises Silva
• [Asterisk-Users] attended call transfer   Moises Silva
• [Asterisk-Users] Dial command to connect two channels and bypassasterisk server   Moises Silva
• [Asterisk-Users] asterisk silence suppression?   Moises Silva
• [Asterisk-Users] asterisk silence suppression?   Moises Silva
• [Asterisk-Users] Keep getting message in logs that pbx.c cannot find extension context 'default'   Moises Silva
• [Asterisk-Users] Is setting the variable _TRANSFER_CONTEXT required in features.conf?   Moises Silva
• [Asterisk-Users] Is setting the variable _TRANSFER_CONTEXT required in features.conf?   Moises Silva
• [Asterisk-Users] Is setting the variable _TRANSFER_CONTEXT required in features.conf?   Moises Silva
• [Asterisk-Users] Problems dialing to another Asterisk server   Moises Silva
• [Asterisk-Users] BAD/GOOD Echo Cancel   David Yat Sin
• [Asterisk-Users] (newby) IAX Trunk on low bandwidth connection   Garth van Sittert
• [Asterisk-Users] (newby) Is PING a good indicator of latency?   Garth van Sittert
• [Asterisk-Users] fax possibilities   Garth van Sittert
• [Asterisk-Users] Outbound Caller ID number on E1   Garth van Sittert
• [Asterisk-Users] Outbound Caller ID number on E1   Garth van Sittert
• [Asterisk-Users] Directed Call Pickup   Garth van Sittert
• [Asterisk-Users] Rewind MusicOnHold?   Garth van Sittert
• [Asterisk-Users] Directed Call Pickup   Garth van Sittert
• [Asterisk-Users] Directed Call Pickup   Garth van Sittert
• [Asterisk-Users] Directed Call Pickup   Garth van Sittert
• [Asterisk-Users] Codec Selection   Garth van Sittert
• [Asterisk-Users] ATA's and faxing   Garth van Sittert
• [Asterisk-Users] ATA's and faxing   Garth van Sittert
• [Asterisk-Users] Handset phone to replace Flash Operator Panel   Garth van Sittert
• [Asterisk-Users] PRI Bridging and Recording   Garth van Sittert
• [Asterisk-Users] Asterisk and Snom 360   Garth van Sittert
• [Asterisk-Users] Hint priority   Garth van Sittert
• [Asterisk-Users] asterisk silence suppression?   Garth van Sittert
• [Asterisk-Users] MixMonitor and command   Garth van Sittert
• [Asterisk-Users] Handset phone to replace Flash Operator Pane l   Garth van Sittert
• [Asterisk-Users] Asterisk hints   Garth van Sittert
• [Asterisk-Users] need help   Garth van Sittert
• [Asterisk-Users] Asterisk hints   Garth van Sittert
• [Asterisk-Users] Asterisk hints   Garth van Sittert
• [Asterisk-Users] Asterisk hints   Garth van Sittert
• [Asterisk-Users] Asterisk - Brooktrout   Tertius Smit
• [Asterisk-Users] BRI Newbie - What Hardware, PCI, in the US?   Tertius Smit
• [Asterisk-Users] Zaptel problem on 4 Processor Opteron SMP system   Tertius Smit
• [Asterisk-Users] Call Waiting x100P and Cisco IP Phone   Chuck Smith
• [Asterisk-Users] Call Waiting x100P and Cisco IP Phone   Chuck Smith
• [Asterisk-Users] Any Digium Supplier/reseller accepts Paypal ?   Garrett Smith
• [Asterisk-Users] Polycom IP601 Question   Kevin Smith
• [Asterisk-Users] Polycom IP601 Question   Kevin Smith
• [Asterisk-Users] Incoming/Outgoing call question   Kevin Smith
• [Asterisk-Users] Sound issue   Kevin Smith
• [Asterisk-Users] Sound Issue   Kevin Smith
• [Asterisk-Users] UK X100P installation help   Paul J. Smith
• [Asterisk-Users] UK X100P installation help   Paul J. Smith
• [Asterisk-Users] UK X100P installation help   Paul J. Smith
• [Asterisk-Users] Sipura SPA 3000 logic   Richard Smith
• [Asterisk-Users] Re: Sipura SPA 3000 logic   Richard Smith
• [Asterisk-Users] Traffic prioritization and 'class of service' for SIP   Rick Smith
• [Asterisk-Users] good voip   Cyber Source
• [Asterisk-Users] Asterisk 1.2.x + oh323 on Debian Sarge.   Craig Southeren
• [Asterisk-Users] realtime sip_buddies does not store ip address   Andrea Cristofanini - Gedam Europe Srl
• [Asterisk-Users] RE: Asterisk-Users Digest, Vol 19, Issue 10   Kevin Steil
• [Asterisk-Users] POTS lines vs. using T1 to connect phone services?? HELP   Kevin Steil
• [Asterisk-Users] POTS lines vs. using T1 to connect phoneservices?? HELP   Kevin Steil
• [Asterisk-Users] Outbound ZAP Dialing   Kevin Steil
• [Asterisk-Users] Fwd: Which ATA device do you recommend?   Kevin Steil
• [Asterisk-Users] ATA's and faxing   Johann Steinwendtner
• [Asterisk-Users] ATA's and faxing   Johann Steinwendtner
• [Asterisk-Users] QSIG error -- can somebody explain?   Johann Steinwendtner
• [Asterisk-Users] SPA-3000 VOIP-PSTN gateway - long delaybetweenanswering and ringing   Chris Stenton
• [Asterisk-Users] SPA-3000 VOIP-PSTN gateway -longdelaybetweenanswering and ringing   Chris Stenton
• [Asterisk-Users] mpg123 alternative?   Chris Stenton
• [Asterisk-Users] Meetme echo cancellation   SteveK
• [Asterisk-Users] CDR for Inbound Calls   James Steven
• [Asterisk-Users] CDR for Inbound Calls   James Steven
• [Asterisk-Users] Call Detail Records for Inbound Calls   James Steven
• [Asterisk-Users] TDM400P digium card   Dewey Straughn
• [Asterisk-Users] TDM400P digium card   Dewey Straughn
• [Asterisk-Users] TDM400P digium card   Dewey Straughn
• [Asterisk-Users] New version of snom soft phone   Christian Stredicke
• [Asterisk-Users] Asterisk and Snom 360   Christian Stredicke
• [Asterisk-Users] What business IP phone to use   Christian Stredicke
• [Asterisk-Users] Dial and Congestion   Massimiliano Stucchi
• [Asterisk-Users] BAD/GOOD Echo Cancel   David Stude
• [Asterisk-Users] 2wav2mp3, monitor, mixmonitor, mpg123, queues   James Sturges
• [Asterisk-Users] Problem with two cards Digium   Bartosz Supczinski
• [Asterisk-Users] Dell server   Paolo Supino
• [Asterisk-Users] winnipeg canada   Technical Support
• [Asterisk-Users] Anyone know a good ITSP in Canada that supports*?   Technical Support
• [Asterisk-Users] User web portal for Asterisk   Technical Support
• [Asterisk-Users] Fedora Core 3 or Fedora Core 4? yum update or not?   Technical Support
• [Asterisk-Users] Fedora Core 3 or Fedora Core 4? yum update ornot?   Technical Support
• [Asterisk-Users] Fedora Core 3 or Fedora Core 4? yum update ornot?   Technical Support
• [Asterisk-Users] sipura 3000 and other probs   Technical Support
• [Asterisk-Users] Fax to Email with Asterisk and Lucent TNT   Technical Support
• [Asterisk-Users] SIP Problem Fedora Core 4 and Asterisk 1.2.4   Technical Support
• [Asterisk-Users] Application Faxing using SIP   Technical Support
• [Asterisk-Users] Application Faxing using SIP   Technical Support
• [Asterisk-Users] Intro to Asterisk & VoIP telephony course - March21st London seats still available   Technical Support
• [Asterisk-Users] NEED COMMENT ON USING FEDORA CORE 3   Technical Support
• [Asterisk-Users] automon not working for analogue phone   Technical Support
• [Asterisk-Users] SPA-941 Selective DND   Technical Support
• [Asterisk-Users] Aastra phones and common directory?   I T
• [Asterisk-Users] Sip - no peer or user found on incoming call   Administrator TOOTAI
• [Asterisk-Users] OT O'Reilly Asterisk TFOT   Administrator TOOTAI
• [Asterisk-Users] how to add stun functionality in asterisk   Administrator TOOTAI
• [Asterisk-Users] fax receive using TDM400P   Administrator TOOTAI
• [Asterisk-Users] Polycomm IP600 continues to ring   Matt TPI
• [Asterisk-Users] Re: delaying "answer" for a number of ringsor anamount   Joe Tahan
• [Asterisk-Users] Detecting Hangups and CDR records   Joe Tahan
• [Asterisk-Users] Re: delaying "answer" for a number of ringsoranamount   Joe Tahan
• [Asterisk-Users] PGSQL asterisk command   Joe Tahan
• [Asterisk-Users] Asterisk with USB   Joe Tahan
• [Asterisk-Users] virtual extension per user ?   Joe Tahan
• [Asterisk-Users] asterisk and week-ends   Joe Tahan
• [Asterisk-Users] extension h and DeadAGI   Joe Tahan
• [Asterisk-Users] No sound on 10% of incoming calls   Joe Tahan
• [Asterisk-Users] No sound on 10% of incoming calls   Joe Tahan
• [Asterisk-Users] Asterisk native sounds now available!   Joe Tahan
• [Asterisk-Users] New GSM 1-8 ports Gateway / Terminal for sale (with SMS Feature and Many more) £99 per unit   Sam Tam
• [Asterisk-Users] GSM Gateway / Terminal for sale   Sam Tam
• [Asterisk-Users] What ATA should I buy?   Sam Tam
• [Asterisk-Users] IP Authorization   Sam Tam
• [Asterisk-Users] IP Authorization   Sam Tam
• [Asterisk-Users] IP Authorization   Sam Tam
• [Asterisk-Users] IP Authorization   Sam Tam
• [Asterisk-Users] IP Authorization   Sam Tam
• [Asterisk-Users] GSM GATEWAY   Sam Tam
• [Asterisk-Users] GSM GATEWAY   Sam Tam
• [Asterisk-Users] $ for an hr of asterisk support   Sam Tam
• [Asterisk-Users] Best ATA for general residential deployment??   Sam Tam
• [Asterisk-Users] Anyone using the GSM gateway from CyberTelecom ?   Sam Tam
• [Asterisk-Users] Anyone using the GSM gateway from CyberTelecom ?   Sam Tam
• [Asterisk-Users] Cheapest provider for Philippine route   Sam Tam
• [Asterisk-Users] GSM phone reception range extendor   Sam Tam
• [Asterisk-Users] Optimizing Linux to run Asterisk   Tamas
• [Asterisk-Users] Optimizing Linux to run Asterisk   Tamas
• [Asterisk-Users] How to handle "provider UNREACHABLE" in the dialplan?   Joseph Tanner
• [Asterisk-Users] callback script?   Joseph Tanner
• [Asterisk-Users] Re: delaying "answer" for a number of rings or an amount   Joseph Tanner
• [Asterisk-Users] callback script?   Joseph Tanner
• [Asterisk-Users] callback script?   Joseph Tanner
• [Asterisk-Users] Some feedback and issues on using chan_bluetooth   Joseph Tanner
• [Asterisk-Users] Re: delaying "answer" for a number of ringsor an amount   Joseph Tanner
• [Asterisk-Users] Re: delaying "answer" for a number of ringsor an amount   Joseph Tanner
• [Asterisk-Users] Re: delaying "answer" for a number of ringsor anamount   Joseph Tanner
• [Asterisk-Users] Re: delaying "answer" for a number of ringsor an amount   Joseph Tanner
• [Asterisk-Users] asterisk and week-ends   Joseph Tanner
• [Asterisk-Users] chan_bluetooth - concurrent calls?   Joseph Tanner
• [Asterisk-Users] chan_bluetooth - concurrent calls?   Joseph Tanner
• [Asterisk-Users] Asterisk with USB   Joseph Tanner
• [Asterisk-Users] touch tones too fast ?   Joseph Tanner
• [Asterisk-Users] Asterisk with USB   Joseph Tanner
• [Asterisk-Users] Asterisk with USB   Joseph Tanner
• [Asterisk-Users] Asterisk with USB   Joseph Tanner
• [Asterisk-Users] Fedora Core 3 or Fedora Core 4? yum update ornot?   Joseph Tanner
• [Asterisk-Users] Chan_BT question WAS: Asterisk with USB   Joseph Tanner
• [Asterisk-Users] Welltech USA? and Wellgate Products?   Joseph Tanner
• [Asterisk-Users] Re: Why is asterisk ignoring my context?   Joseph Tanner
• [Asterisk-Users] lists problem, Gmail????????   Joseph Tanner
• [Asterisk-Users] ChanIsAvail   Joseph Tanner
• [Asterisk-Users] ChanIsAvail   Joseph Tanner
• [Asterisk-Users] Software E.C. Along with Tellabs   Joseph Tanner
• [Asterisk-Users] Good VoIP providers that support Asterisk PBX's   Joseph Tanner
• [Asterisk-Users] Asterisk on SPA-3000 rings extensions only after third ring   Joseph Tanner
• [Asterisk-Users] Asterisk on SPA-3000 rings extensions only afterthird ring   Joseph Tanner
• SV: [Asterisk-Users] Problems with voicemail   Joseph Tanner
• [Asterisk-Users] Grandstream GXP-2000   Soner Tari
• [Asterisk-Users] Grandstream GXP-2000   Soner Tari
• [Asterisk-Users] mpg123 alternative?   Soner Tari
• [Asterisk-Users] mpg123 alternative?   Soner Tari
• [Asterisk-Users] mpg123 alternative?   Soner Tari
• [Asterisk-Users] Asterisk with HT 488 FXO   Soner Tari
• [Asterisk-Users] Polycom bootrom and SIP software   AR Tarzi
• [Asterisk-Users] Tormenta 2 and channel bank   Viktor Tatianin
• [Asterisk-Users] Tormenta CAS signaling   Viktor Tatianin
• [Asterisk-Users] Tormenta CAS signaling   Viktor Tatianin
• [Asterisk-Users] Tormenta CAS signaling   Viktor Tatianin
• [Asterisk-Users] Tormenta CAS signaling   Viktor Tatianin
• [Asterisk-Users] Tormenta CAS signaling   Viktor Tatianin
• [Asterisk-Users] Asterisk and Hipath interconnections   Viktor Tatianin
• [Asterisk-Users] Asterisk and Hipath interconnections   Viktor Tatianin
• [Asterisk-Users] Codec Selection   Olivier Taylor
• [Asterisk-Users] Codec Selection   Olivier Taylor
• [Asterisk-Users] New issue tracker for handling licensing issues for Asterisk, Zaptel and related projects   Asterisk Development Team
• [Asterisk-Users] Zaptel 1.2.4 Released!   The Asterisk Development Team
• [Asterisk-Users] [asterisk-dev] Zaptel 1.2.4 Released!   The Asterisk Development Team
• [Asterisk-Users] Asterisk compile error   MBIT Technologies
• [Asterisk-Users] Zaptel problem on 4 Processor Opteron SMP system   Chris Teesdale
• [Fwd: Re: [Asterisk-Users] Asterisk for Call Center (missing reference)]   Rodrigo P. Telles
• [Asterisk-Users] How to determine the power draw on TDM2400P?   Melisa Teoh
• [Asterisk-Users] How to determine the power draw on TDM2400P?   Melisa Teoh
• [Asterisk-Users] Channel bank woes - no outbound calls   James Texter
• [Asterisk-Users] SOLVED - Channel bank woes - no outbound calls   James Texter
• [Asterisk-Users] Asterisk with T1 card on laptop   James Texter
• [Asterisk-Users] FW: Re: Delay on Phone ringing   Ash Thakrar
• [Asterisk-Users] FW: Re: Delay on Phone ringing   Ash Thakrar
• [Asterisk-Users] Random Hangups/Disconnects   Thczv F. Thczv
• [Asterisk-Users] voicemail files in Asterisk have rights 600 , I need 644   Thomas
• [Asterisk-Users] voicemail files in Asterisk have rights 600 , I need 644   Thomas
• [Asterisk-Users] Re: chan_sip.c:5262 sip_reg_timeout Probably a DNS error for registration   Thomas
• [Asterisk-Users] sipgate.de question   Thomas
• [Asterisk-Users] DTMF Sporadicaly Being Generated   Rob Thomas
• [Asterisk-Users] An array of extensions in my lab   Rob Thomas
• [Asterisk-Users] Weird DTMF issue   Joshua M Thompson
• [Asterisk-Users] Weird DTMF issue   Joshua M Thompson
• [Asterisk-Users] Weird DTMF issue   Joshua M Thompson
• [Asterisk-Users] Voicemail 0 for operator call routing   Paul Tinsley
• [Asterisk-Users] Voicemail 0 for operator call routing   Paul Tinsley
• [Asterisk-Users] Skype-to-Asterisk(SIP): progress   John Todd
• [Asterisk-Users] Skype-to-Asterisk(SIP): progress   John Todd
• [Asterisk-Users] RE: 5, 000 concurrent calls system rollout question   John Todd
• [Asterisk-Users] RE: 5,000 concurrent calls system rollout question   John Todd
• [Asterisk-Users] Asterisk native sounds now available!   John Todd
• [Asterisk-Users] Uninstall Asterisk   Tom
• [Asterisk-Users] Re: Uninstall Asterisk   Tom
• [Asterisk-Users] RE: Teliax - Codec Preference effective?   Brent Torrenga
• [Asterisk-Users] Re: Asterisk hangs on 1.2.1   Brent Torrenga
• [Asterisk-Users] RE: Rewind MusicOnHold?   Brent Torrenga
• [Asterisk-Users] RE: Rewind MusicOnHold?   Brent Torrenga
• [Asterisk-Users] BRI Newbie - What Hardware, PCI, in the US?   Brent Torrenga
• [Asterisk-Users] Random Disconnects - or ARE they?   Brent Torrenga
• [Asterisk-Users] RE: Random Disconnects - or ARE they?   Brent Torrenga
• [Asterisk-Users] RE: Random Disconnects - or ARE they?   Brent Torrenga
• [Asterisk-Users] RE: Random Disconnects - or ARE they?   Brent Torrenga
• [Asterisk-Users] Random Hangups/Disconnects   Brent Torrenga
• [Asterisk-Users] RE: Cisco 79xx and SIP 7.5 Problems   Brent Torrenga
• [Asterisk-Users] Echo on PRI/BRI?   Brent Torrenga
• [Asterisk-Users] Re: Echo and other reasons to migrate to BRI from POTS? Was (Echo on PRI/BRI?)   Brent Torrenga
• [Asterisk-Users] Re: Web interface   Steve Totaro
• [Asterisk-Users] G729 Commercial Licenses.   Steve Totaro
• [Asterisk-Users] G729a Pass-Through and Recording/Monitoring   Steve Totaro
• [Asterisk-Users] Monitoring   Steve Totaro
• [Asterisk-Users] G729a Pass-Through and Recording/Monitoring   Steve Totaro
• [Asterisk-Users] List Broken Again?   Steve Totaro
• [Asterisk-Users] List Broken Again?   Steve Totaro
• [Asterisk-Users] BAD/GOOD Echo Cancel   Steve Totaro
• [Asterisk-Users] Problems with Cepstral and Asterisk   Steve Totaro
• [Asterisk-Users] Sendmail with exchange   Steve Totaro
• [Asterisk-Users] TE411P Really Bad Echo   Steve Totaro
• [Asterisk-Users] Quintum Tenor AX 24 Port SIP FXS "Unsupported Media Type"   Steve Totaro
• [Asterisk-Users] Quintum Tenor AX 24 Port SIP FXS "UnsupportedMedia Type"   Steve Totaro
• [Asterisk-Users] Linear Queues Strategies for 3rd Party Application   Steve Totaro
• [Asterisk-Users] Monitor a call in progress.   Steve Totaro
• [Asterisk-Users] Work in Ukraine   Wojciech Tryc
• [Asterisk-Users] Voicemail Problem   Wojciech Tryc
• [Asterisk-Users] Voicemail Problem   Wojciech Tryc
• [Asterisk-Users] PIKA Technologies Inc. Announces Support for Open Source Asterisk PBX   Wojciech Tryc
• [Asterisk-Users] A unique 'click to call' project - Could use someadvice   Wojciech Tryc
• [Asterisk-Users] A unique 'click to call' project - Could usesomeadvice   Wojciech Tryc
• [Asterisk-Users] auto provision of IP501 polycom   Wojciech Tryc
• [Asterisk-Users] auto provision of IP501 polycom   Wojciech Tryc
• [Asterisk-Users] segmentation fault   Justin Tunney
• [Asterisk-Users] hardware and network requirements   Alyed Tzompa
• [Asterisk-Users] help with oh323   Hussain Umair
• [Asterisk-Users] Outbound Caller ID number on E1   Steve Underwood
• [Asterisk-Users] varion card   Steve Underwood
• [Asterisk-Users] BAD/GOOD Echo Cancel   Steve Underwood
• [Asterisk-Users] BAD/GOOD Echo Cancel   Steve Underwood
• [Asterisk-Users] BAD/GOOD Echo Cancel   Steve Underwood
• [Asterisk-Users] FXS with v.90 modem support?   Steve Underwood
• [Asterisk-Users] fax <-> misdn <-> zap <-> fax // fax <-> misdn <-> ATA <-> fax   Steve Underwood
• [Asterisk-Users] Tormenta CAS signaling   Steve Underwood
• [Asterisk-Users] Problem win Unicall   Steve Underwood
• [Asterisk-Users] more cpu intensive echo cancellers ?   Steve Underwood
• [Asterisk-Users] more cpu intensive echo cancellers ?   Steve Underwood
• [Asterisk-Users] TE411P Really Bad Echo   Steve Underwood
• [Asterisk-Users] Tormenta CAS signaling   Steve Underwood
• [Asterisk-Users] Tormenta CAS signaling   Steve Underwood
• [Asterisk-Users] Tormenta CAS signaling   Steve Underwood
• [Asterisk-Users] fax receive using TDM400P   Steve Underwood
• [Asterisk-Users] fax receive using TDM400P   Steve Underwood
• [Asterisk-Users] fax receive using TDM400P   Steve Underwood
• [Asterisk-Users] Unistim Packet Decoder   Polycom User
• [Asterisk-Users] Unistim Packet Decoder   Polycom User
• [Asterisk-Users] Channel SS7   VOICEIN at aol.com
• [Asterisk-Users] ISDN Eicon Diva Server V-BRI   Jens Vagelpohl
• [Asterisk-Users] Fedora Core 3 or Fedora Core 4? yum update ornot?   Jens Vagelpohl
• [Asterisk-Users] automon not working for analogue phone   Ondrej Valousek
• [Asterisk-Users] Queue - check agent   Joel Vandal
• [Asterisk-Users] Sangoma analog cards?   Joel Vandal
• [Asterisk-Users] mpg123 alternative?   Joel Vandal
• [Asterisk-Users] sipdiscount   Alejandro Vargas
• [Asterisk-Users] sipdiscount   Alejandro Vargas
• [Asterisk-Users] one way / irratic voice over iax and g729   Alejandro Vargas
• [Asterisk-Users] sipdiscount   Alejandro Vargas
• [Asterisk-Users] sipdiscount   Alejandro Vargas
• [Asterisk-Users] sipdiscount   Alejandro Vargas
• [Asterisk-Users] spa3000   Alejandro Vargas
• [Asterisk-Users] spa3000   Alejandro Vargas
• [Asterisk-Users] immediate pick up in "s"   Alejandro Vargas
• [Asterisk-Users] Re: immediate pick up in "s"   Alejandro Vargas
• [Asterisk-Users] spa3000   Alejandro Vargas
• [Asterisk-Users] did from sip trunk   Alejandro Vargas
• [Asterisk-Users] did from sip trunk   Alejandro Vargas
• [Asterisk-Users] debugging asterisk configuration   Alejandro Vargas
• [Asterisk-Users] Skype vs. an Xlite registered to Asterisk   Alejandro Vargas
• [Asterisk-Users] variables internas   Alejandro Vargas
• [Asterisk-Users] Manager cmd: originate without picking up the fone?!   Arnd Vehling
• [Asterisk-Users] Re: Contents of Asterisk-Users digest...   Will Velez
• [Asterisk-Users] Grandstream Budgetone mass deployment?   Ezio Vernacotola
• [Asterisk-Users] how to configure my asterisk@home 1.0.9 to do call forwarding ?   Maxim Vexler
• [Asterisk-Users] 5,000 concurrent calls system rollout question   Vic
• [Asterisk-Users] 5,000 concurrent calls system rollout question   Vic
• [Asterisk-Users] transferring 3000 SIP calls   Vic
• [Asterisk-Users] TE210P + MicroITX as E1 to TDMoE appliance?   Christian Victor
• [Asterisk-Users] digium TE405P and intel motherboard   Christian Victor
• [Asterisk-Users] POTS lines vs. using T1 to connect phone services?? HELP   Tom Vile
• [Asterisk-Users] Grandstream hold one way audio -URGENT   Tom Vile
• [Asterisk-Users] ztdummy configuration issues   Vinod
• [Asterisk-Users] co-location providers in Ottawa, Canada   VirTERM
• [Asterisk-Users] XLite dtmf issue?   Vladyslav
• [Asterisk-Users] 911 and ISDN PRI   Adam Vocks
• [Asterisk-Users] Codec negotiation   Ronald Voermans
• [Asterisk-Users] Codec negotiation   Ronald Voermans
• [Asterisk-Users] Codec negotiation   Ronald Voermans
• [Asterisk-Users] Grandstream hold one way audio -URGENT   Ronald Voermans
• [Asterisk-Users] Static problems with Asterisk + Polycom phones   Roman Volf
• [Asterisk-Users] Polycom dialplan restriction   Roman Volf
• Using *RT for HA purposes was: [Asterisk-Users]Realtime MultipleAsterisk boxes, iaxusers   Charles Wang
• [Asterisk-Users] Re: [Serusers] high-availibility setup using f5 bigip   Charles Wang
• [Asterisk-Users] TDM 400P in Malaysia   Darryl Ware
• [Asterisk-Users] 911 and ISDN PRI   Watkins, Bradley
• [Asterisk-Users] Nortel Meridian Opt 81C and PRI   Watkins, Bradley
• [Asterisk-Users] Multiple AGI Issues   Watkins, Bradley
• [Asterisk-Users] What business IP phone to use   Watkins, Bradley
• [Asterisk-Users] Queue - check agent   David Waugh
• [Asterisk-Users] RE: Queue - check agent   David Waugh
• [Asterisk-Users] BRI Newbie - What Hardware, PCI, in the US?   David Waugh
• [Asterisk-Users] chan_capi and Eicon Diva   David Waugh
• [Asterisk-Users] playing hold time announcement without queue position announcement   Franklin Webb
• [Asterisk-Users] Newbie question   Robert Webb
• [Asterisk-Users] [OT] List messages and end user outages   Robert Webb
• [Asterisk-Users] chan_sccp availability?   Andy Webster
• [Asterisk-Users] AMP 1.10.010 Config Problem   Mark Welch
• [Asterisk-Users] AMP 1.10.010 Config Problem   Mark Welch
• [Asterisk-Users] Re: asterisk logger - urgent!!!   Ron Wellsted
• [Asterisk-Users] GPS-enabled cell phone/PDA   Michael Welter
• [Asterisk-Users] Call quality problems   Michael Welter
• [Asterisk-Users] Call quality problems   Michael Welter
• [Asterisk-Users] Call quality problems   Michael Welter
• [Asterisk-Users] Call quality problems   Michael Welter
• [Asterisk-Users] Problem with chan-capi: outgoing calls on two lines   Karsten Wemheuer
• [Asterisk-Users] Problem with chan-capi: outgoing calls on two lines   Karsten Wemheuer
• [Asterisk-Users] Problem with chan-capi: outgoing calls on two lines   Karsten Wemheuer
• [Asterisk-Users] Q: Status of feature Call Deflection / Partial Rerouing in chan-capi and zaphfc   Karsten Wemheuer
• [Asterisk-Users] FXS with v.90 modem support?   Dustin Wenz
• [Asterisk-Users] FXS with v.90 modem support?   Dustin Wenz
• [Asterisk-Users] Sipura SPA-2002 rings randomly   BJ Weschke
• [Asterisk-Users] 64bit processor and 32 bit digium card   BJ Weschke
• [Asterisk-Users] Polycom dialplan restriction   BJ Weschke
• [Asterisk-Users] 4 TE411P in one server installation   BJ Weschke
• [Asterisk-Users] More Polycom IP501 questions   BJ Weschke
• [Asterisk-Users] Safely editing voicemail.conf   BJ Weschke
• [Asterisk-Users] Anyway to pass CIC in sip header   BJ Weschke
• [Asterisk-Users] A unique 'click to call' project - Could use some advice   BJ Weschke
• [Asterisk-Users] MixMonitor and command   BJ Weschke
• [Asterisk-Users] Bridged line appearance   BJ Weschke
• [Asterisk-Users] call parking "hint"   BJ Weschke
• [Asterisk-Users] Voice conferencing server capacity   BJ Weschke
• [Asterisk-Users] Polycom 501 ACDlogin   BJ Weschke
• [Asterisk-Users] Polycom 501 ACDlogin   BJ Weschke
• [Asterisk-Users] Polycom IP 601 Buddy Watch doesn't work after Asterisk reload   BJ Weschke
• [Asterisk-Users] Linksys WIP300 WiFi Phone   BJ Weschke
• [Asterisk-Users] Asterisk Follow Me   BJ Weschke
• [Asterisk-Users] Linksys WIP300 WiFi Phone   BJ Weschke
• [Asterisk-Users] BLF not working after reload   BJ Weschke
• [Asterisk-Users] Capturing DIALSTATUS on a PARTICULAR channel if multiple-dialling?   BJ Weschke
• [Asterisk-Users] change languages from an IVR   Derek Whitten
• [Asterisk-Users] IP Authorization   Darren Wiebe
• [Asterisk-Users] IP Authorization   Darren Wiebe
• [Asterisk-Users] Billing inbound calls per minute   Darren Wiebe
• [Asterisk-Users] IP Authorization   Darren Wiebe
• [Asterisk-Users] Solution for 1 time blast of 200, 000 recorded calls   Darren Wiebe
• [Asterisk-Users] Billing inbound calls per minute   Darren Wiebe
• [Asterisk-Users] mysql phone number pattern match query   Darren Wiebe
• [Asterisk-Users] AGI Scripts Terminate too Soon   Darren Wiebe
• [Asterisk-Users] billing - different tarif per phone   Darren Wiebe
• [Asterisk-Users] Re: Asterisk Question   Darren Wiebe
• [Asterisk-Users] BAD/GOOD Echo Cancel   Eric "ManxPower" Wieling
• [Asterisk-Users] TE411P Really Bad Echo   Eric "ManxPower" Wieling
• [Asterisk-Users] attended call transfer   Eric "ManxPower" Wieling
• [Asterisk-Users] Best quad-port fxo solution with EC?   Eric "ManxPower" Wieling
• [Asterisk-Users] Use one sip account for multiple sipura   Eric "ManxPower" Wieling
• [Asterisk-Users] ChanIsAvail   Eric "ManxPower" Wieling
• [Asterisk-Users] Loops and Variables   Eric "ManxPower" Wieling
• [Asterisk-Users] Re: Call centre - * hang's up   Eric "ManxPower" Wieling
• [Asterisk-Users] Dial from AGI = no ring back ??   Eric "ManxPower" Wieling
• [Asterisk-Users] What business IP phone to use   Eric "ManxPower" Wieling
• [Asterisk-Users] What business IP phone to use   Eric "ManxPower" Wieling
• [Asterisk-Users] Anyone using the GSM gateway from CyberTelecom ?   Eric "ManxPower" Wieling
• [Asterisk-Users] Polycom Echo   Eric "ManxPower" Wieling
• [Asterisk-Users] Polycom Echo   Eric "ManxPower" Wieling
• [Asterisk-Users] incoming calls dropout on PRI over TE110p   Eric "ManxPower" Wieling
• [Asterisk-Users] 5,000 concurrent calls system rollout question   Dustin Wildes
• [Asterisk-Users] Need to Hire PHP Programmer(s)   Dustin Wildes
• [Asterisk-Users] Detect answer and hangup   Wile
• [Asterisk-Users] Asterisk vs. Traditional PBX   William
• [Asterisk-Users] Asterisk 1.2.4 IAX2 New Jitterbuffer Tuning   Mark Willis
• [Asterisk-Users] Dumb Dialout Question   Ronald Wiplinger
• [Asterisk-Users] How to handle "provider UNREACHABLE" in the dialplan?   Ronald Wiplinger
• [Asterisk-Users] username not stabled? * DO NOT USE USERNAME for locally attached phones!!!   Ronald Wiplinger
• [Asterisk-Users] How to handle "provider UNREACHABLE" in the dialplan?   Ronald Wiplinger
• [Asterisk-Users] mysql problems   Ronald Wiplinger
• [Asterisk-Users] IAX2 through Shorewall rpoblem   Manny A. Wise
• [Asterisk-Users] wrong dell   Hans Witvliet
• [Asterisk-Users] TDM400p   Hans Witvliet
• [Asterisk-Users] Re: How many TDM2400P's will a server take?   Hans Witvliet
• [Asterisk-Users] TDM2400P FXS Only vs. T1/E1 to FXS Channel Banks   Hans Witvliet
• [Asterisk-Users] authenticate problem   Wojtek
• [Asterisk-Users] Connecting two phones with different codecs   Lisa Wolf
• [Asterisk-Users] Connecting two phones with different codecs   Lisa Wolf
• [Asterisk-Users] Cisco Gateway and Context Issues   Sum Ding Wong
• [Asterisk-Users] Cisco Gateway and Context Issues   Sum Ding Wong
• [Asterisk-Users] username not stabled? * DO NOT USE USERNAME for locally attached phones!!!   Sum Ding Wong
• [Asterisk-Users] XLite dtmf issue?   Sum Ding Wong
• [Asterisk-Users] Cisco Gateway and Context Issues   Sum Ding Wong
• [Asterisk-Users] Pattern Match - 0 or 1 digit   Sum Ding Wong
• [Asterisk-Users] Pattern Match - 0 or 1 digit   Sum Ding Wong
• [Asterisk-Users] Lockups since upgrade 1.2.3 - anyone else? Any ideas?   Sum Ding Wong
• [Asterisk-Users] Cisco Gateway and Context Issues   Sum Ding Wong
• [Asterisk-Users] bug in bristuff?   Conrad Wood
• [Asterisk-Users] bug in bristuff?   Conrad Wood
• [Asterisk-Users] GotoIf number exists in file. How can i do this?   Conrad Wood
• [Asterisk-Users] bug in bristuff?   Conrad Wood
• [Asterisk-Users] BRI Newbie - What Hardware, PCI, in the US?   Conrad Wood
• [Asterisk-Users] attended call transfer   Conrad Wood
• [Asterisk-Users] Anyone using the GSMgateway from CyberTelecom ?   Conrad Wood
• [Asterisk-Users] MOH from RCA jack?   Conrad Wood
• [Asterisk-Users] Cheap BRI card   Conrad Wood
• [Asterisk-Users] What business IP phone to use   Conrad Wood
• [Asterisk-Users] TFTP server for GrandStream BT phones / need testing   Conrad Wood
• [Asterisk-Users] TFTP server for GrandStream BT phones / need testing   Conrad Wood
• [Asterisk-Users] What business IP phone to use   Conrad Wood
• [Asterisk-Users] Pickup call on Hold   Conrad Wood
• [Asterisk-Users] What business IP phone to use   Conrad Wood
• [Asterisk-Users] Incoming/Outgoing call question   Conrad Wood
• [Asterisk-Users] What business IP phone to use   Conrad Wood
• [Asterisk-Users] mysql problems   Conrad Wood
• [Asterisk-Users] What business IP phone to use   Conrad Wood
• [Asterisk-Users] TE411P Really Bad Echo ORION   Darren Wright
• [Asterisk-Users] TE411P Really Bad Echo   Darren Wright
• [Asterisk-Users] TE411P Really Bad Echo   Darren Wright
• [Asterisk-Users] Software E.C. Along with Tellabs   Darren Wright
• [Asterisk-Users] RE: 5, 000 concurrent calls system rollout question   Wai Wu
• [Asterisk-Users] RE: 5, 000 concurrent calls system rolloutquestion   Wai Wu
• [Asterisk-Users] RE: 5, 000 concurrent calls system rolloutquestion   Wai Wu
• [Asterisk-Users] can asterisk to say chinese like say english   Wai Wu
• [Asterisk-Users] g729 license question   Wai Wu
• [Asterisk-Users] Connecting to live calls   Wai Wu
• [Asterisk-Users] Connecting to live calls   Wai Wu
• [Asterisk-Users] Connecting to live calls   Wai Wu
• [Asterisk-Users] Manager API 'Redirect' is not working for both end of a call.   Wai Wu
• [Asterisk-Users] Connecting to live calls   Wai Wu
• [Asterisk-Users] Virtual Extensions   Wai Wu
• [Asterisk-Users] automatically start application from the commandprompt   Wai Wu
• [Asterisk-Users] Manager cmd: originate without picking up thefone?!   Wai Wu
• [Asterisk-Users] Dial command to connect two channels and bypassasterisk server   Wai Wu
• [Asterisk-Users] Dial command to connect two channelsand bypassasterisk server   Wai Wu
• [Asterisk-Users] Solution for 1 time blast of 200, 000 recorded calls   Wai Wu
• [Asterisk-Users] Supported protocols in pri   Wai Wu
• [Asterisk-Users] Polycom IP 601 Buddy Watch problems   Isaac Xiao
• [Asterisk-Users] Polycom IP 601 Buddy Watch problems   Isaac Xiao
• [Asterisk-Users] Asterisk and Hipath interconnections   Isaac Xiao
• [Asterisk-Users] alternative to realtime?   Tielin Xu
• [Asterisk-Users] Trunk calls ring internal analog phone   Steven Yelton
• [Asterisk-Users] Welltech USA? and Wellgate Products?   Vahan Yerkanian
• [Asterisk-Users] Sipura SPA-3000 and PSTN dtmf   Vahan Yerkanian
• [Asterisk-Users] stable ISDN BRI card for asterisk   Peng Yong
• [Asterisk-Users] L option of Dial does not work properly   Peng Yong
• [Asterisk-Users] DTMF Sporadicaly Being Generated   Michael L. Young
• [Asterisk-Users] Alarmreceiver   Shane Young
• [Asterisk-Users] php+agi   Ever Zalazar
• [Asterisk-Users] asterisk and S.E.R.   Ever Zalazar
• [Asterisk-Users] Re: Can't send DTMF transfer code from called SIP phone   Milan Zamazal
• [Asterisk-Users] asterisk still tries native bridging   Igor Zamocky
• [Asterisk-Users] asterisk still tries native bridging   Igor Zamocky
• [Asterisk-Users] Double ring   Edward de Zeeuw
• [Asterisk-Users] How do I install speex for asterisk?   Jesus E Zepeda
• [Asterisk-Users] How do I install speex for asterisk?   Jesus E Zepeda
• [Asterisk-Users] How do I install speex for asterisk?   Jesus E Zepeda
• [Asterisk-Users] How do I install speex for asterisk?   Jesus E Zepeda
• [Asterisk-Users] Re: How do I install speex for asterisk?   Jesus E Zepeda
• [Asterisk-Users] Asterisk 1.2.4 IAX2 New Jitterbuffer Tuning   Jesus E Zepeda
• [Asterisk-Users] re: questions about sip requests to asterisk 1.2   C. Zerbo
• [Asterisk-Users] early media   Jiang Zhou
• 答复: [Asterisk-Users] Search for Links for "Communicating PC to PC inthe same lan through Asterisk "   Jiang Zhou
• [Asterisk-Users] Routing Calls via chan_capi with AVM FritzCard   Juergen K. Zick
• [Asterisk-Users] GPS-enabled cell phone/PDA   Juergen K. Zick
• [Asterisk-Users] GPS-enabled cell phone/PDA   Juergen K. Zick
• [Asterisk-Users] Voice Over WiFi   Juergen K. Zick
• [Asterisk-Users] iax2 trunking known problems?   Zoa
• [Asterisk-Users] iax2 trunking known problems?   Zoa
• [Asterisk-Users] iax2 trunking known problems?   Zoa
• [Asterisk-Users] iax2 trunking known problems?   Zoa
• [Asterisk-Users] Looking for Q.Sig success story   Patrick Zwahlen
• [Asterisk-Users] Corrupt CDR records in Asterisk 1.2.x   Jeroen Zwarts
• [Asterisk-Users] Corrupt CDR records in Asterisk 1.2.x   Jeroen Zwarts
• [Asterisk-Users] QSIG error -- can somebody explain?   Wolfgang Zweimueller
• [Asterisk-Users] QSIG error -- can somebody explain?   Wolfgang Zweimueller
• [Asterisk-Users] Alcatel 4200 series pbx   Wolfgang Zweimueller
• [Asterisk-Users] Austria isdn p2p empty DID   Wolfgang Zweimueller
• [Asterisk-Users] Problem win Unicall   acriollo
• [Asterisk-Users] Caller ID patches - updated   adam at net-europa.com
• [Asterisk-Users] TAPI Recommendations   adam at net-europa.com
• [Asterisk-Users] sipdiscount   adibar
• [Asterisk-Users] Anyone using the GSMgateway from CyberTelecom ?   adibar
• [Asterisk-Users] Anyone using the GSMgateway from CyberTelecom ?   adibar
• [Asterisk-Users] Sorry for the multiple-posts... I had a mailserver-hickup   adibar
• [Asterisk-Users] sipdiscount   adibar
• [Asterisk-Users] sipdiscount   adibar
• [Asterisk-Users] Instant Messaging: with SIP or XMPP   roswel ajf
• [Asterisk-Users] is there a web interface to this mailing list?   roswel ajf
• [Asterisk-Users] Alarmreceiver   andrutto
• [Asterisk-Users] advanced options access problem   arun arora
• [Asterisk-Users] Skype vs. an Xlite registered to Asterisk   asterisk
• [Asterisk-Users] CallerID popup   asterisk183
• [Asterisk-Users] problem with Zaptel   asterisk183
• [Asterisk-Users] echo problem   asterisk183
• [Asterisk-Users] Strange echo phenomenon (double tandem)   asterisk at anime.net
• [Asterisk-Users] 5,000 concurrent calls system rollout question   asterisk at anime.net
• [Asterisk-Users] 5,000 concurrent calls system rollout question   asterisk at anime.net
• [Asterisk-Users] Asterisk on laptop connected to POTS line   asterisk at anime.net
• [Asterisk-Users] RE: 5, 000 concurrent calls system rolloutquestion   asterisk at anime.net
• [Asterisk-Users] BAD/GOOD Echo Cancel   asterisk at anime.net
• [Asterisk-Users] thomson speedtouch ST2030   asterisk at anime.net
• [Asterisk-Users] BAD/GOOD Echo Cancel   asterisk at anime.net
• [Asterisk-Users] TE411P Really Bad Echo   asterisk at anime.net
• [Asterisk-Users] Qwest disconnect supervision?   asterisk at anime.net
• [Asterisk-Users] TE411P Really Bad Echo   asterisk at anime.net
• [Asterisk-Users] TE411P Really Bad Echo   asterisk at anime.net
• [Asterisk-Users] Grandstream Budgetone mass deployment?   asterisk at anime.net
• [Asterisk-Users] Grandstream Budgetone mass deployment?   asterisk at anime.net
• [Asterisk-Users] Best quad-port fxo solution with EC?   asterisk at anime.net
• [Asterisk-Users] Asterisk Televantage integration   asterisk at anime.net
• [Asterisk-Users] Solution for 1 time blast of 200, 000 recorded calls   asterisk at anime.net
• [Asterisk-Users] Solution for 1 time blast of 200, 000 recorded calls   asterisk at anime.net
• [Asterisk-Users] Fax to Email with Asterisk and Lucent TNT   asterisk at anime.net
• [Asterisk-Users] Grandstream hold one way audio -URGENT   asterisk at anime.net
• [Asterisk-Users] Fax to Email with Asterisk and Lucent TNT   asterisk at anime.net
• [Asterisk-Users] Asterisk and Snom 360   asterisk at anime.net
• [Asterisk-Users] zoom FXS/FXO gateways   asterisk at anime.net
• [Asterisk-Users] zoom FXS/FXO gateways   asterisk at anime.net
• [Asterisk-Users] Grandstream GXP-2000   asterisk at anime.net
• [Asterisk-Users] Grandstream GXP-2000   asterisk at anime.net
• [Asterisk-Users] snom 360 incorrect US indications   asterisk at anime.net
• [Asterisk-Users] Grandstream GXP-2000   asterisk at anime.net
• [Asterisk-Users] Grandstream GXP-2000   asterisk at anime.net
• [Asterisk-Users] Asterisk and Snom 360   asterisk at anime.net
• [Asterisk-Users] Grandstream GXP-2000   asterisk at anime.net
• [Asterisk-Users] Grandstream GXP-2000   asterisk at anime.net
• [Asterisk-Users] Grandstream GXP-2000   asterisk at anime.net
• [Asterisk-Users] Re: Call centre - * hang's up   asterisk at anime.net
• [Asterisk-Users] Grandstream GXP-2000   asterisk at anime.net
• [Asterisk-Users] Grandstream GXP-2000   asterisk at anime.net
• [Asterisk-Users] What business IP phone to use   asterisk at anime.net
• [Asterisk-Users] What business IP phone to use   asterisk at anime.net
• [Asterisk-Users] how to add stun functionality in asterisk   asterisk at anime.net
• [Asterisk-Users] What business IP phone to use   asterisk at anime.net
• [Asterisk-Users] What business IP phone to use   asterisk at anime.net
• [Asterisk-Users] What business IP phone to use   asterisk at anime.net
• [Asterisk-Users] Best ATA for general residential deployment??   asterisk at anime.net
• [Asterisk-Users] snom 360 problem - only one call works after reboot   asterisk at anime.net
• [Asterisk-Users] Best ATA for general residential deployment??   asterisk at anime.net
• [Asterisk-Users] Polycom IP 601 Buddy Watch problems   asterisk at anime.net
• [Asterisk-Users] Newbie config help? Wellgate 3701a   asterisk at anime.net
• [Asterisk-Users] Zoom 5801 problems with *   asterisk at anime.net
• [Asterisk-Users] Zoom 5801 problems with *   asterisk at anime.net
• [Asterisk-Users] MOH sourced from a sound card?   asterisk at b-great.net
• [Asterisk-Users] international calling via POTS in Russia   asterisk at b-great.net
• [Asterisk-Users] php agi configuration issue   asterisk at frameweb.it
• R: [Asterisk-Users] php agi configuration issue   asterisk at frameweb.it
• [Asterisk-Users] php agi configuration issue SOLVED   asterisk at frameweb.it
• [Asterisk-Users] clid and src fields wrong in cdr   asterisk at frameweb.it
• [Asterisk-Users] clid and src fields wrong in cdr SOLVED   asterisk at frameweb.it
• [Asterisk-Users] Corrupt CDR records in Asterisk 1.2.x   asterisk at frameweb.it
• [Asterisk-Users] Corrupt CDR records in Asterisk 1.2.x   asterisk at frameweb.it
• [kpj@junghanns.net: Re: [asterisk@frameweb.it: RE: [Asterisk-Users] Corrupt CDR records in Asterisk 1.2.x]]   asterisk at frameweb.it
• [Asterisk-Users] Bug in AMP 1.10.010 in sip outbound callerid   asterisk at frameweb.it
• [Asterisk-Users] Bug in AMP 1.10.010 in sip outbound callerid   asterisk at frameweb.it
• [Asterisk-Users] problem with SU100   asterisk at frameweb.it
• [Asterisk-Users] S100U and TigerJet   asterisk at frameweb.it
• [Asterisk-Users] S100U and TigerJet   asterisk at frameweb.it
• [Asterisk-Users] Big problems with Voicemails ODBC Storage   asterisk at mazone.info
• [Asterisk-Users] TE411P Really Bad Echo   asterisk at oneringnetworks.com
• [Asterisk-Users] MINNESOTA: TwinCities Asterisk Users Group - Saturday 02/11/2006   asterisk_help at iwishi.nu
• [Asterisk-Users] interface to dpnss   bails
• [Asterisk-Users] using AMP custom extensions   bails
• [Asterisk-Users] [Fwd: using AMP custom extensions]   bails
• [Asterisk-Users] q931 85   bails
• [Asterisk-Users] international calling via POTS in Russia   balint.kovacs at moveonerelo.com
• [Asterisk-Users] Dundi key Problem   bbench at mail.bg
• [Asterisk-Users] callback script?   bbench at mail.bg
• [Asterisk-Users] Billing inbound calls per minute   bbench at mail.bg
• [Asterisk-Users] Billing inbound calls per minute   bbench at mail.bg
• [Asterisk-Users] Billing inbound calls per minute   bbench at mail.bg
• [Asterisk-Users] Billing inbound calls per minute   bbench at mail.bg
• [Asterisk-Users] Billing inbound calls per minute   bbench at mail.bg
• [Asterisk-Users] TDM04B/TDM2401E Card   bbench at mail.bg
• [Asterisk-Users] Asterisk returning 403 Forbidden response   bharat.sarvan at wipro.com
• [Asterisk-Users] Delay after first digit - dial plan   brett at websmyths.com
• [Asterisk-Users] Voicemail Problem   brett at websmyths.com
• [Asterisk-Users] setup default console debug levels   brett at websmyths.com
• [Asterisk-Users] Sendmail with exchange   brian at txshirts.com
• [Asterisk-Users] MOH from RCA jack?   brian at txshirts.com
• [Asterisk-Users] Softphones and other VOIP PBX's   brian at txshirts.com
• [Asterisk-Users] Best ATA for general residential deployment??   broadbandvoice at comcast.net
• [Asterisk-Users] seg fault when skinny phone answers   btb
• [Asterisk-Users] context being ignored by inbound sip call   btb
• [Asterisk-Users] context being ignored by inbound sip call   btb
• [Asterisk-Users] context being ignored by inbound sip call   btb
• [Asterisk-Users] voicemail folders   btb
• [Asterisk-Users] incoming call release after 1 ring   leonimar cape
• [Asterisk-Users] incoming call release after 1 ring   leonimar cape
• [Asterisk-Users] asterisk h323   leonimar cape
• [Asterisk-Users] asterisk h323   leonimar cape
• [Asterisk-Users] asterisk h323   leonimar cape
• [Asterisk-Users] asterisk nagios plugin   leonimar cape
• [Asterisk-Users] digium TE405P and intel motherboard   leonimar cape
• [Asterisk-Users] unicall channel on asteriskathome 2.5   leonimar cape
• [Asterisk-Users] Comedian Mail Wont Take Password   casasterisk at valnet.com
• [Asterisk-Users] Polycom IP501 Endless Loop   casasterisk at valnet.com
• [Asterisk-Users] Dumb Dialout Question   casasterisk at valnet.com
• [Asterisk-Users] Multiple Asterisk Server Question   casasterisk at valnet.com
• [Asterisk-Users] IAX Extension Dialing Question   casasterisk at valnet.com
• [Asterisk-Users] fax pass-through   marek cervenka
• [Asterisk-Users] fax pass-through   marek cervenka
• [Asterisk-Users] SIP and firewalls?   chentschel at arnet.com.ar
• [Asterisk-Users] SIP and firewalls?   chentschel at arnet.com.ar
• [Asterisk-Users] TE411P Really Bad Echo   chip at innovates.com
• [Asterisk-Users] Dial command exits non-zero   cmould
• [Asterisk-Users] mpg123 alternative?   sean cook
• [Asterisk-Users] Good VoIP providers that support Asterisk PBX's   John covici
• [Asterisk-Users] meetme feature request (or maybe its there already?)   John covici
• [Asterisk-Users] Problem transferring call to a meetme conference   John covici
• [Asterisk-Users] Problem calling a ZAP channel with svn 10842   John covici
• [Asterisk-Users] CISCO 1760 with 1 BRI   Jean-Louis curty
• [Asterisk-Users] CISCO 1760 with 1 BRI   Jean-Louis curty
• [Asterisk-Users] Help with sip setup because can't receive calls!!!!!!   abc def
• [Asterisk-Users] Help with sip setup because can't receive calls!!!!!!   abc def
• [Asterisk-Users] Help with sip setup because can't receive calls!!!!!!   abc def
• [Asterisk-Users] Help with sip setup because can't receive calls!!!!!!   abc def
• [Asterisk-Users] asterisk and week-ends   demigor
• [Asterisk-Users] Asterisk start errors with TDM2413E   duane.pudenz at shastaindustries.com
• [Asterisk-Users] Asterisk start errors with TDM2413E   duane.pudenz at shastaindustries.com
• [Asterisk-Users] Asterisk start errors with TDM2413E   duane.pudenz at shastaindustries.com
• [Asterisk-Users] Calls not going through   duane.pudenz at shastaindustries.com
• [Asterisk-Users] OT O'Reilly Asterisk TFOT   f6hqz-m at hamwlan.net
• [Asterisk-Users] OT O'Reilly Asterisk TFOT   f6hqz-m at hamwlan.net
• [Asterisk-Users] OT O'Reilly Asterisk TFOT   f6hqz-m at hamwlan.net
• [Asterisk-Users] BAD/GOOD Echo Cancel   f6hqz-m at hamwlan.net
• [Asterisk-Users] To connect between more than 2 asterisk server [links needed ]   f6hqz-m at hamwlan.net
• [Asterisk-Users] Best quad-port fxo solution with EC?   f6hqz-m at hamwlan.net
• [Asterisk-Users] Asterisk start errors with TDM2413E   f6hqz-m at hamwlan.net
• [Asterisk-Users] Detecting disconnect on TDM400P with 3 FXO portsand 1 FXS port   f6hqz-m at hamwlan.net
• [Asterisk-Users] IAX2 through Shorewall rpoblem   f6hqz-m at hamwlan.net
• [Asterisk-Users] Ringing Delay   f6hqz-m at hamwlan.net
• [Asterisk-Users] [help] warning 4246   fabrizio
• [Asterisk-Users] 2wav2mp3, monitor, mixmonitor, mpg123, queues   g.delduca at webster.it
• [Asterisk-Users] ZAP <--> sip(polycom301) can not hear each other   sdgesa gaeharth
• [Asterisk-Users] ZAP <--> sip(polycom301) can not hear each other   sdgesa gaeharth
• [Asterisk-Users] Sirrix BRI errors   garth at bitco.co.za
• [Asterisk-Users] Asterisk register ip phone   al gav
• [Asterisk-Users] Cisco 7960 Register Problem   al gav
• [Asterisk-Users] ISDN busy line   francesco giuliani
• [Asterisk-Users] ISDN busy line   francesco giuliani
• [Asterisk-Users] Free IAX login   Asterisk guy
• [Asterisk-Users] Free IAX login   Asterisk guy
• [Asterisk-Users] Free IAX login   Asterisk guy
• [Asterisk-Users] Free IAX login   Asterisk guy
• [Asterisk-Users] Error running iaxcomm   ast guy
• [Asterisk-Users] routing question: multipath routing for SIP   gw at adcomcorp.com
• [Asterisk-Users] Re: delaying "answer" for a number of ringsor an amount of time   gw at adcomcorp.com
• [Asterisk-Users] Chan capi failing post build 8015, possible causes?   gw at adcomcorp.com
• [Asterisk-Users] Problem with Wait() and chan_capi-cm?   gw at adcomcorp.com
• [Asterisk-Users] Problem with Wait() and chan_capi-cm?   gw at adcomcorp.com
• [Asterisk-Users] Best quad-port fxo solution with EC?   gw at adcomcorp.com
• [Asterisk-Users] asterisk + door opener   gw at adcomcorp.com
• [Asterisk-Users] Best quad-port fxo solution with EC?   gw at adcomcorp.com
• [Asterisk-Users] Best quad-port fxo solution with EC?   gw at adcomcorp.com
• [Asterisk-Users] sip registration fails with 404   wendell hamilton
• [Asterisk-Users] show calls   jonny hashem
• [Asterisk-Users] Polycom 501 ACDlogin   hgaillac-sip at yahoo.fr
• [Asterisk-Users] query about Three way calling   himanshu at tataelxsi.co.in
• [Asterisk-Users] Anyone know a good ITSP in Canada that supports *?   hugolivude
• [Asterisk-Users] Skype vs. an Xlite registered to Asterisk   hugolivude
• [Asterisk-Users] Skype vs. an Xlite registered to Asterisk   hugolivude
• [Asterisk-Users] EDGE-CORE SIP PHONE   listas iPfone
• [Asterisk-Users] Called party number   imata at plenty.co.jp
• [Asterisk-Users] Any Digium Supplier/reseller accepts Paypal ?   isamar at isamarmaia.org
• [Asterisk-Users] Pound to Hangup an ongoing call   isamar at isamarmaia.org
• [Asterisk-Users] Analog with channel bank - Inbound works, outbound doesn't   james.texter at cox.net
• [Asterisk-Users] Analog with channel bank - Inbound works, outbound doesn't   james.texter at cox.net
• [Asterisk-Users] Analog with channel bank - Inbound works, outbound doesn't   james.texter at cox.net
• [Asterisk-Users] Analog with channel bank - Inbound works, outbound doesn't   james.texter at cox.net
• [Asterisk-Users] Channel bank woes - no outbound calls   james.texter at cox.net
• SV: [Asterisk-Users] Re: CallerID Problem   jan.sarin at securia.se
• SV: [Asterisk-Users] Re: CallerID Problem   jan.sarin at securia.se
• SV: [Asterisk-Users] Outbound Caller ID number on E1   jan.sarin at securia.se
• SV: [Asterisk-Users] delaying "answer" for a number of rings or anamount of time   jan.sarin at securia.se
• SV: SV: [Asterisk-Users] delaying "answer" for a number of ringsor anamount of time   jan.sarin at securia.se
• SV: [Asterisk-Users] BAD/GOOD Echo Cancel   jan.sarin at securia.se
• SV: [Asterisk-Users] Help on queues   jan.sarin at securia.se
• [Asterisk-Users] Polycom 501 ACDlogin   jan.sarin at securia.se
• SV: [Asterisk-Users] Polycom 501 ACDlogin   jan.sarin at securia.se
• SV: [Asterisk-Users] Polycom 501 ACDlogin   jan.sarin at securia.se
• [Asterisk-Users] BAD/GOOD Echo Cancel   janvb at caselaboratories.com
• [Asterisk-Users] AVAYA H.323 IP phone account and Asterisk   janvb at caselaboratories.com
• [Asterisk-Users] asterisk related job offer in Florida   jarnaud at prosodiemail.com
• [Asterisk-Users] Zap tuning for echo/gain   jerry at voiptower.com
• [Asterisk-Users] TDD   jltaylor
• [Asterisk-Users] TE411P Really Bad Echo   jltaylor
• [Asterisk-Users] Looking for Asterisk Platform for DID   john
• [Asterisk-Users] sipura 3000 and other probs   john at cusick.ws
• [Asterisk-Users] sipura 3000 and other probs   john at cusick.ws
• [Asterisk-Users] Grandstream Budgetone BT-101 audio problems   jurgen
• [Asterisk-Users] Compiling Intel G729 error   atik khan
• [Asterisk-Users] register => 2345:password@sip_proxy doesn't care about port   kleis-asterisk-dev at tiscali.it
• [Asterisk-Users] queues   kurtz
• [Asterisk-Users] SIP to H.323 Native bridging ...   pack3t l0ss
• [Asterisk-Users] Call centre - * hang's up   lenz
• [Asterisk-Users] XLite dtmf issue?   kevin ling
• [Asterisk-Users] Call Waiting x100P and Cisco IP Phone   kevin ling
• [Asterisk-Users] French and German translations?   kevin ling
• [Asterisk-Users] Welltech USA? and Wellgate Products?   kevin ling
• [Asterisk-Users] Free IAX login   kevin ling
• [Asterisk-Users] TDM04B FXO Asterisk@Home   kevin ling
• [Asterisk-Users] Welltech USA? and Wellgate Products?   kevin ling
• [Asterisk-Users] Asterisk native sounds now available!   kevin ling
• [Asterisk-Users] Welltech USA? and Wellgate Products?   kevin ling
• [Asterisk-Users] Web based SIP client   kevin ling
• [Asterisk-Users] cisco 7940 firmware upgrade   kevin ling
• [Asterisk-Users] festival-script.pl... howto change language?   kevin ling
• [Asterisk-Users] attended call transfer   kevin ling
• [Asterisk-Users] Sendmail with exchange   kevin ling
• [Asterisk-Users] Possible for Asterisk to output CLID to invoke 3rd party app?   lists.digium.com at al-najjar.co.uk
• [Asterisk-Users] Possible for Asterisk to output CLID to invoke 3rd party app?   lists.digium.com at al-najjar.co.uk
• [Asterisk-Users] asterisk call start detection   maka
• [Asterisk-Users] Aasterisk large-scale deployment w/analog phones   maka
• [Asterisk-Users] test of FXO and FXS in TDM400P   makevuy
• [Asterisk-Users] asterisk 1.2.4 doesn't detect the PSTN hang up   makevuy
• [Asterisk-Users] Good VoIP providers that support Asterisk PBX's   andrew matthews
• [Asterisk-Users] Good VoIP providers that support Asterisk PBX's   andrew matthews
• [Asterisk-Users] Re: What business IP phone to use   andrew matthews
• [Asterisk-Users] Asterisk on laptop connected to POTS line   mgraves at mstvp.com
• [Asterisk-Users] Aasterisk large-scale deployment w/analog phones   mgraves at mstvp.com
• [Asterisk-Users] Recommended rack-mountable server anyone?   mitcheloc at gmail.com
• [Asterisk-Users] Recommended rack-mountable server anyone?   mitcheloc at gmail.com
• [Asterisk-Users] Recommended rack-mountable server anyone?   mitcheloc at gmail.com
• [Asterisk-Users] Re:   mkumar at mantragroup.com
• [Asterisk-Users] Configure DID   mkumar at mantragroup.com
• [Asterisk-Users] Problem calling out   mkumar at mantragroup.com
• [Asterisk-Users] Problem with incoming call, Please help   mkumar at mantragroup.com
• [Asterisk-Users] TDM04B/TDM2401E Card   housi mueller
• [Asterisk-Users] TDM04B/TDM2401E Card   housi mueller
• [Asterisk-Users] Newbie question   housi mueller
• [Asterisk-Users] Newbie question   housi mueller
• [Asterisk-Users] Bridged line appearance   mustardman29
• [Asterisk-Users] Grandstream GXP-2000   mustardman29
• [Asterisk-Users] Grandstream GXP-2000   mustardman29
• [Asterisk-Users] Bridged line appearance   mustardman29
• [Asterisk-Users] Bridged line appearance   mustardman29
• [Asterisk-Users] Grandstream GXP-2000   mustardman29
• [Asterisk-Users] Grandstream GXP-2000   mustardman29
• [Asterisk-Users] What business IP phone to use   mustardman29
• [Asterisk-Users] What business IP phone to use   mustardman29
• [Asterisk-Users] What business IP phone to use   mustardman29
• [Asterisk-Users] Re: Voicemail problems   mustardman29
• [Asterisk-Users] What business IP phone to use   mustardman29
• [Asterisk-Users] What business IP phone to use   mustardman29
• [Asterisk-Users] What business IP phone to use   mustardman29
• [Asterisk-Users] Asterisk as a dedicated Analog PSTN gateway   mustardman29
• [Asterisk-Users] BLF not working after reload   mustardman29
• FW: [Asterisk-Users] BLF not working after reload   mustardman29
• [Asterisk-Users] BLF not working after reload   mustardman29
• [Asterisk-Users] BLF not working after reload   mustardman29
• [Asterisk-Users] How hard to create Asterisk for Compact Flash?   mustardman29
• [Asterisk-Users] How to determine the power draw on TDM2400P?   mustardman29
• [Asterisk-Users] What are these error messages in my logs?   myotheremail at netacc.net
• [Asterisk-Users] how to log agents into a queue   nik600
• [Asterisk-Users] inform the agent about the queue he is answering   nik600
• [Asterisk-Users] hardware and network requirements   nik600
• [Asterisk-Users] inform the agent about the queue he is answering   nik600
• [Asterisk-Users] inform the agent about the queue he is answering   nik600
• [Asterisk-Users] inform the agent about the queue he is answering   nik600
• [Asterisk-Users] configure TE205P on asterisk@home   nik600
• [Asterisk-Users] configure TE205P on asterisk@home   nik600
• [Asterisk-Users] configure TE205P on asterisk@home   nik600
• [Asterisk-Users] configure TE205P on asterisk@home   nik600
• [Asterisk-Users] configure TE205P on asterisk@home   nik600
• [Asterisk-Users] problem with outgoing calls Unable to create channel of type 'ZAP' (cause 34 - Circuit/channel congestion)   nik600
• [Asterisk-Users] problem with outgoing calls Unable to createchannel of type 'ZAP' (cause 34 - Circuit/channel congestion)   nik600
• [Asterisk-Users] problem with outgoing calls Unable tocreatechannel of type 'ZAP' (cause 34 - Circuit/channel congestion)   nik600
• [Asterisk-Users] problem with outgoing calls Unable tocreatechannel of type 'ZAP' (cause 34 - Circuit/channel congestion)   nik600
• [Asterisk-Users] problem with outgoing calls Unabletocreatechannel of type 'ZAP' (cause 34 - Circuit/channel congestion)   nik600
• [Asterisk-Users] inbound DID trunked   nik600
• [Asterisk-Users] problem with outgoing callsUnabletocreatechannel of type 'ZAP' (cause 34 - Circuit/channelcongestion)   nik600
• [Asterisk-Users] problem with outgoing callsUnabletocreatechannelof type 'ZAP' (cause 34 - Circuit/channelcongestion)   nik600
• [Asterisk-Users] [slackware 10.2 and TE205P] Unknown signalling method 'pri_cpe'   nik600
• [Asterisk-Users] [slackware 10.2 and TE205P] Unknown signalling method 'pri_cpe'   nik600
• [Asterisk-Users] [slackware 10.2 and TE205P] Unknown signalling method 'pri_cpe'   nik600
• [Asterisk-Users] problem with outgoingcallsUnabletocreatechannelof type 'ZAP' (cause 34 -Circuit/channelcongestion)   nik600
• [Asterisk-Users] monitor outgoing calls in queue / campaings   nik600
• [Asterisk-Users] MP3player Problem   office
• AW: [Asterisk-Users] MP3player Problem   office
• AW: AW: [Asterisk-Users] MP3player Problem   office
• [Asterisk-Users] chan_bluetooth jabra 200 / 250   Jason p
• [Asterisk-Users] RE: Teliax - Codec Preference effective?   tim panton
• [Asterisk-Users] (newby) IAX Trunk on low bandwidth connection   tim panton
• [Asterisk-Users] ddi???   tim panton
• [Asterisk-Users] ddi???   tim panton
• [Asterisk-Users] (newby) Asterisk on the open internet & security   tim panton
• [Asterisk-Users] Application Faxing using SIP   tim panton
• [Asterisk-Users] GSM GATEWAY   tim panton
• [Asterisk-Users] Blocked Callerid   pdhales at optusnet.com.au
• [Asterisk-Users] Dumb Dialout Question   pdhales at optusnet.com.au
• [Asterisk-Users] No Audio on Local Machine, Remote works fine   pdhales at optusnet.com.au
• [Asterisk-Users] Blocked Callerid   pdhales at optusnet.com.au
• [Asterisk-Users] fax possibilities   pdhales at optusnet.com.au
• [Asterisk-Users] Anyway to do this?   pdhales at optusnet.com.au
• [Asterisk-Users] Blocked Callerid   pdhales at optusnet.com.au
• [Asterisk-Users] Blocked Callerid   pdhales at optusnet.com.au
• [Asterisk-Users] limit sip sessions   pdhales at optusnet.com.au
• [Asterisk-Users] IP PAX gateway to PSTN   pdhales at optusnet.com.au
• [Asterisk-Users] IP PAX gateway to PSTN   pdhales at optusnet.com.au
• [Asterisk-Users] IP PAX gateway to PSTN   pdhales at optusnet.com.au
• [Asterisk-Users] Do we need a QOS switch ?   pdhales at optusnet.com.au
• [Asterisk-Users] IP PAX gateway to PSTN   pdhales at optusnet.com.au
• [Asterisk-Users] IP PAX gateway to PSTN   pdhales at optusnet.com.au
• [Asterisk-Users] Two Lines, Two Businesses   pdhales at optusnet.com.au
• [Asterisk-Users] Issues in Australia? Ringing, iaxy etc   pdhales at optusnet.com.au
• [Asterisk-Users] configure TE205P on asterisk@home   pdhales at optusnet.com.au
• [Asterisk-Users] Connecting two phones with different codecs   pdhales at optusnet.com.au
• [Asterisk-Users] Wildfire messsaging server   pdhales at optusnet.com.au
• [Asterisk-Users] Re: Call centre - * hang's up   pdhales at optusnet.com.au
• [Asterisk-Users] good voip   pdhales at optusnet.com.au
• [Asterisk-Users] Re: Call centre - * hang's up   pdhales at optusnet.com.au
• [Asterisk-Users] Multiple TDM400P's in a single machine   pdhales at optusnet.com.au
• [Asterisk-Users] $ for an hr of asterisk support   pdhales at optusnet.com.au
• [Asterisk-Users] immediate pick up in "s"   pdhales at optusnet.com.au
• [Asterisk-Users] Download "Asterisk: The Future Of Telephony"[More Info]   pdhales at optusnet.com.au
• [Asterisk-Users] What business IP phone to use   pdhales at optusnet.com.au
• [Asterisk-Users] Voicemail 0 for operator call routing   pdhales at optusnet.com.au
• [Asterisk-Users] Re: Asterisk Question   pdhales at optusnet.com.au
• [Asterisk-Users] Re: Asterisk Question   pdhales at optusnet.com.au
• [Asterisk-Users] Re: Asterisk Question   pdhales at optusnet.com.au
• [Asterisk-Users] Re: Asterisk Question   pdhales at optusnet.com.au
• [Asterisk-Users] asterisk -rx 'commands'   bill peck
• [Asterisk-Users] Server & Wildcard TE110P   phil.dawson at marnock.com
• [Asterisk-Users] ddi???   phil.dawson at marnock.com
• [Asterisk-Users] Do we need a QOS switch ?   phil.dawson at marnock.com
• [Asterisk-Users] Asterisk errors configuring for PRI   phil.dawson at marnock.com
• [Asterisk-Users] Asterisk errors configuring for PRI   phil.dawson at marnock.com
• [Asterisk-Users] ISDN30E + T1 crossover cable woes   phil.dawson at marnock.com
• [Asterisk-Users] Newbie h323 question   phil at telinco.net
• [Asterisk-Users] newbie debugger needs a little guidance   phoneserver
• [Asterisk-Users] Asterisk with Billing   ram
• [Asterisk-Users] Voice Drop Due to Low RAM   ram
• [Asterisk-Users] TE411P problem-- probably stupid.   rapples at comcast.net
• [Asterisk-Users] Asterisk SIP phones to Cisco Unity via CCM4.0SIP Trunk   sys read
• [Asterisk-Users] Asterisk SIP phones to Cisco Unity via CCM4.0SIPTrunk   sys read
• [Asterisk-Users] Not receiving faxes and other case   yrving rivas
• [Asterisk-Users] Waiting for your help...   yrving rivas
• [Asterisk-Users] Waiting for your help...   yrving rivas
• [Asterisk-Users] How to make asterisk fax and fax to email work?   yrving rivas
• [Asterisk-Users] Grandstream GXP-2000   yrving rivas
• [Asterisk-Users] fax receive using TDM400P   yrving rivas
• [Asterisk-Users] Is it possible ?   amna saleem
• [Asterisk-Users] Linksys WIP300 WiFi Phone   amna saleem
• [Asterisk-Users] lspci don't have Tiger Jet Network Inc   mohamed sammir
• [Asterisk-Users] Codec Selection   sdcharly at gmail.com
• [Asterisk-Users] Aastra phones and common directory?   sdcharly at gmail.com
• [Asterisk-Users] Good VoIP providers that support Asterisk PBX's   sdcharly at gmail.com
• [Asterisk-Users] SIP registration on Sipura 841   sdcharly at gmail.com
• [Asterisk-Users] Re: SIP registration on Sipura 841   sdcharly at gmail.com
• [Asterisk-Users] Re: Anyone using the GSMgateway from CyberTelecom ?   sdcharly at gmail.com
• [Asterisk-Users] Choosing a GSM gateway for personal use.   sdcharly at gmail.com
• [Asterisk-Users] Anyone using the GSM gateway from CyberTelecom ?   sdcharly at gmail.com
• [Asterisk-Users] IAX provider recommendation wanted   sdcharly at gmail.com
• [Asterisk-Users] Re: Asterisk Question   sdcharly at gmail.com
• [Asterisk-Users] intel 536 ep as fxo -> possible?   stevanus
• [Asterisk-Users] Directed Call Pickup   steve
• [Asterisk-Users] Re: Asterisk Question   steve
• [Asterisk-Users] Obtaining billsecs in the dialplan after a call?   steve at daviesfam.org
• [Asterisk-Users] Line Dropouts on E405P   steve at daviesfam.org
• [Asterisk-Users] hardware and network requirements   stoffell
• [Asterisk-Users] User web portal for Asterisk   stoffell
• [Asterisk-Users] Do we need a QOS switch ?   stoffell
• [Asterisk-Users] thomson speedtouch ST2030   stoffell
• [Asterisk-Users] thomson speedtouch ST2030   stoffell
• [Asterisk-Users] bug in bristuff?   stoffell
• [Asterisk-Users] Performance differences 64-bit vs 32-bit   stoffell
• [Asterisk-Users] aastra v1.3.1 firmware   stoffell
• [Asterisk-Users] RE: aastra v1.3.1 firmware   stoffell
• [Asterisk-Users] Grandstream GXP-2000   stoffell
• [Asterisk-Users] What business IP phone to use   stoffell
• [Asterisk-Users] What business IP phone to use   stoffell
• [Asterisk-Users] BLF not working after reload   stoffell
• [Asterisk-Users] How hard to create Asterisk for Compact Flash?   stoffell
• [Asterisk-Users] How hard to create Asterisk for Compact Flash?   stoffell
• [Asterisk-Users] Polycom 501 issues   rivy strauss
• [Asterisk-Users] hardware suggestion   sukrit
• [Asterisk-Users] NSLU2 Asterisk   sukrit
• [Asterisk-Users] co-location providers in Ottawa, Canada   pcman theMan
• [Asterisk-Users] Mitel 5220 IP phones   tracinet
• [Asterisk-Users] Grandstream GXP-2000   tracinet
• [Asterisk-Users] Dial command to connect two channels and bypass asterisk server   turby
• [Asterisk-Users] asterisk t.38 pass   turby
• [Asterisk-Users] Re: asterisk t.38 pass   turby
• [Asterisk-Users] context being ignored by inbound sip call   turby
• [Asterisk-Users] username as extension   turby
• [Asterisk-Users] callthru and CDR   turby
• [Asterisk-Users] return code from AGI   vivek at staff.ownmail.com
• [Asterisk-Users] Rtp packets being dropped   vivek at staff.ownmail.com
• [Asterisk-Users] new jitter implementation for sip   vivek at staff.ownmail.com
• [Asterisk-Users] TDM400P digium card   vn-elist
• [Asterisk-Users] AVAYA H.323 IP phone account and Asterisk   voip voip
• [Asterisk-Users] Static problems with Asterisk + Polycom phones   volfman at keystreams.com
• [Asterisk-Users] Cisco 2620 as PRI gateway   kurt x
• [Asterisk-Users] Cisco 2620 as PRI gateway   kurt x
• [Asterisk-Users] Zap interface with Atbill   xcel
• [Asterisk-Users] Different Voice Prompts at Different Times   yusuf
• [Asterisk-Users] Developing a call centre app. Communication with asterisk?   yusuf
• [Asterisk-Users] Asterisk errors configuring for PRI   yusuf
• [Asterisk-Users] G723 error   yusuf
• [Asterisk-Users] CDR for Inbound Calls   yusuf
• [Asterisk-Users] asterisk h323   yusuf
• [Asterisk-Users] iax2 trunking known problems?   yusuf
• [Asterisk-Users] asterisk-1.2.4 + asterisk-addons-1.2.1 for mysql realtime   yusuf
• [Asterisk-Users] G723 error   yusuf
• [Asterisk-Users] asterisk h323   yusuf
• [Asterisk-Users] ~1 sec delay from callee answering to call established on dialout   yusuf
• [Asterisk-Users] GSM GATEWAY   yusuf
• [Asterisk-Users] GSM GATEWAY   yusuf
• [Asterisk-Users] Asterisk 1.2.4 IAX2 New Jitterbuffer Tuning   yusuf
• [Asterisk-Users] realtime, iax, trunk   yusuf
• [Asterisk-Users] Cannot see the caller id , When calls made from one server to another   yusuf
• [Asterisk-Users] Asterisk, SIP phone , NAT   yusuf
• [Asterisk-Users] Asterisk Topology   yusuf
• [Asterisk-Users] Problem with T1 installation   yusuf
• [Asterisk-Users] VAD, CNG, for Zap   yusuf
• [Asterisk-Users] Problema calling from elesign h.323 to iax   yusuf at ecntelecoms.com
• [Asterisk-Users] Recommended rack-mountable server anyone?   yusuf at ecntelecoms.com
• [Asterisk-Users] H.323 ( HW PBX to *)   yusuf at ecntelecoms.com
• [Asterisk-Users] Dazed and Confused   vinicius zanc
• [Asterisk-Users] Busy signalling for mobile callers ?   Tristan Graham { Skymarket Limited }