October 2006 Archives by author

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Starting: Sun Oct 1 01:09:05 MST 2006
Ending: Tue Oct 31 23:56:30 MST 2006
Messages: 3146

• [asterisk-users] IAX softphones   Francesco Peeters (Asterisk)
• [asterisk-users] IAX softphones   Francesco Peeters (Asterisk)
• [asterisk-users] using asterisk to do remote control   David Cook (Canada)
• [asterisk-users] Ringtones won't work   Mike Haney (Gmail)
• [asterisk-users] New Asterisk StumbleUpon Group   Matt Riddell (IT)
• [asterisk-users] New Asterisk StumbleUpon Group   Matt Riddell (IT)
• [asterisk-users] Go to DIGIUMBOARDS.COM   Matt Riddell (IT)
• [asterisk-users] XO SIP Origination Services   Jason Aarons (US)
• [asterisk-users] RE: Getting Asterisk to work with GoogleTalk   Bromont -
• [asterisk-users] Asterisk architecure   jez .
• [asterisk-users] Architecture for Asterisk   jezzzz .
• [asterisk-users] Re: Architecture for Asterisk   jezzzz .
• [asterisk-users] Architecture for Asterisk   jezzzz .
• [asterisk-users] Architecture for Asterisk   jezzzz .
• [asterisk-users] how to setup call center with media gateway?   吴金中
• [asterisk-users] Core dumps when Releasing clone lock   Øyvind Albrigtsen
• [asterisk-users] Centos kernel 34 vs. 42?   Bryan J. Smith
• [asterisk-users] Get user context from dialplan.   José Luis Gómez
• [asterisk-users] Switchtype,Signalling,rxwink warnings   Josué Conti
• [asterisk-users] Spying a channel in a meetme   Nicolás Gudiño
• [asterisk-users] pop a web page with DID in url   Nicolás Gudiño
• [asterisk-users] Page hangs up after 5 seconds   Torbjörn Abrahamsson
• [asterisk-users] Page hangs up after 5 seconds   Torbjörn Abrahamsson
• [asterisk-users] No Dialtone   Eddie Johnson Jr
• [asterisk-users] No Dialtone   Eddie Johnson Jr
• [asterisk-users] Quintum DX as gateway to PSTN for Asterisk   doki_cti
• Re: Re: [asterisk-users] Quintum DX as gateway to PSTN for Asterisk   doki_cti
• [asterisk-users] 1.4 Beta and oracle   René Enskat [Teamware GmbH]
• [asterisk-users] adding outbound prefix   Ed Nuñez
• [asterisk-users] auto recording extensions   Ed Nuñez
• [asterisk-users] ISDN-BRI issue   Frédéric Blaise
• [asterisk-users] ISDN-BRI issue   Frédéric Blaise
• [asterisk-users] Advice on GUI   Frédéric Blaise
• [asterisk-users] Snom 320, Queues and Transfer not working as expected with * 1.2.12.1   Håkan Källberg
• SV: [asterisk-users] Screen pop based on incoming DID   Jon Schøpzinsky
• [asterisk-users] Choice of soundfile format   Jon Schøpzinsky
• [asterisk-users] Spying a channel in a meetme   Eduard Martínez
• [asterisk-users] Spying a channel in a meetme   Eduard Martínez
• [asterisk-users] Re: IAX2 show peers - description   Frédéric Marti
• [asterisk-users] Re: Voip Buster - CID   Tomislav Parčina
• [asterisk-users] Re: Voip Buster - CID   Tomislav Parčina
• [asterisk-users] Re: RPID   Tomislav Parčina
• [asterisk-users] Re: CDR stats to one mysql database, multiple webstats packages   Tomislav Parčina
• [asterisk-users] Beronet BN4S0 instalation   Tomislav Parčina
• [asterisk-users] Re: Call Asterisk : It calls me backup with a dial tone   Tomislav Parčina
• [asterisk-users] Re: unauthenticated calls   Tomislav Parčina
• [asterisk-users] Re: Cisco 7970 SIP won't update?   Tomislav Parčina
• [asterisk-users] Dial - i parametar   Tomislav Parčina
• [asterisk-users] Digium on Dell PowerEdge 1850   Tomislav Parčina
• [asterisk-users] Cisco 7970 - versionStamp   Tomislav Parčina
• [asterisk-users] Re: Digium on Dell PowerEdge 1850   Tomislav Parčina
• [asterisk-users] Re: echotraining=yes in misdn.conf is invalid or out of range.   Tomislav Parčina
• [asterisk-users] sip.conf - srvlookup   Tomislav Parčina
• [asterisk-users] Re: Cisco 7971G-GE & SEP{MAC}.cnf.xml   Tomislav Parčina
• [asterisk-users] Re: "Fixing the Caller-ID Problem", by John Todd for O'ReillyNet   Tomislav Parčina
• [asterisk-users] Intel S3000AHLX - Digium TE110P   Tomislav Parčina
• [asterisk-users] Re: IAX2 show peers - description   Tomislav Parčina
• [asterisk-users] Re: IAX2 show peers - description   Tomislav Parčina
• [asterisk-users] Re: Forwarding recorded calls to Voicemail   Tomislav Parčina
• [asterisk-users] Advice on GUI   Frédéric Blaise
• [asterisk-users] Asterisk and dialer Running on Thin Clients   Ignacio Ortega A.
• [asterisk-users] Asterisk and dialer Running on Thin Clients   Ignacio Ortega A.
• [asterisk-users] Asterisk and dialer Running on Thin Clients   Ignacio Ortega A.
• [asterisk-users] RE:Asterisk and dialer Running on Thin Clients   Ignacio Ortega A.
• [asterisk-users] cisco 2600   Idris AVCI
• [asterisk-users] Inbound Callcenter with multiple DIDs   Idris AVCI
• [asterisk-users] asterisk-addons-1.2.4 Installation Problem   Abdul
• [asterisk-users] OH323 Fake ring   Abdul
• [asterisk-users] Calls being disconnected across VPN   Jason Adams
• [asterisk-users] "Cheapest" way to determine channels in a group from outside asterisk?   Nick Adams
• [asterisk-users] Re: "Cheapest" way to determine channels in a group from outside asterisk?   Nick Adams
• [asterisk-users] Re: DTMF Tones   Nick Adams
• [asterisk-users] CALEA support within asterisk?   Rich Adamson
• [asterisk-users] CALEA support within asterisk?   Rich Adamson
• [asterisk-users] CALEA support within asterisk?   Rich Adamson
• [asterisk-users] Extremely choppy sound on some of ourPOTSnetwork calls; goes away with mute   Rich Adamson
• [asterisk-users] Outbound FXO call, getting "You must first dial..."   Rich Adamson
• [asterisk-users] No Dialtone   Rich Adamson
• [asterisk-users] No Dialtone   Rich Adamson
• [asterisk-users] FYI - Polycom SoundPoint IP 301 Denial of Service]   Rich Adamson
• [asterisk-users] VOIP with PSTN backup   Rich Adamson
• [asterisk-users] DID failover   Rich Adamson
• [asterisk-users] SPA942 quality for a Bank   Rich Adamson
• [asterisk-users] SPA942 quality for a Bank   Rich Adamson
• [asterisk-users] Stopping putgoing calls after working hours   Rich Adamson
• [asterisk-users] FXO Cards vs. Channel bank with T1   Rich Adamson
• [asterisk-users] Asterisk 1.4.0 compile error on AMD64 Opteron server; recompile with -fPIC?   Gabriel Afana
• [asterisk-users] ECHO Cancellation in SIP Calls   Stefan Agethen
• [asterisk-users] ECHO Cancellation in SIP Calls   Stefan Agethen
• [asterisk-users] Re: asterisk-users Digest, Vol 27, Issue 140   Stefan Agethen
• [asterisk-users] Re: ECHO Cancellation in SIP Calls   Stefan Agethen
• [asterisk-users] RE: ECHO Cancellation in SIP Calls   Stefan Agethen
• [asterisk-users] Re: Linksys PAP2: calling tone stops after 5   Stefan Agethen
• [asterisk-users] CDR problem with call transfer   Kamran Ahmad
• [asterisk-users] Asterisk on 64bit xeon   Akpome Akpoguma
• [asterisk-users] Asterisk on 64bit xeon   Akpome Akpoguma
• [asterisk-users] [OT] wi-fi ip phone scenario   Alban
• [asterisk-users] [OT] wi-fi ip phone scenario   Alban
• [asterisk-users] VoiceOne 0.4.0 released: a new web-based and open source GUI   Alex
• [asterisk-users] VoiceOne 0.4.0 released: a new web-based and open source GUI   Alex
• [asterisk-users] for god's sake somebody help me! ANSWEREDTIME=0 in astcc!!   Ali
• [asterisk-users] astcc help-pleasssssseeee   Ali
• [asterisk-users] astcc help-pleasssssseeee   Ali
• [asterisk-users] asterisk gui sans live cd   Patrick Aljord
• [asterisk-users] show logged clients   Pablo Allietti
• [asterisk-users] Possible to set max voicemail mesasge limit per user   Christopher Aloi
• [asterisk-users] Multiple queue_log files based on queue - is it possible??   Christopher Aloi
• [asterisk-users] Asterisk 1.2.12.1 and snom 360 6.2.3 no audio   Holger von Ameln
• [asterisk-users] Re: Two phones, same number   Benny Amorsen
• [asterisk-users] Re: Bandwidth requirements   Benny Amorsen
• [asterisk-users] Re: Bandwidth requirements   Benny Amorsen
• [asterisk-users] Re: PoE IP Phone   Benny Amorsen
• [asterisk-users] Re: OT: Polycom time sync - sorta   Benny Amorsen
• [asterisk-users] Re: Where is the PlayDTMF command?   Benny Amorsen
• [asterisk-users] Re: Asterisk 1.2.12 - Can NOT make call out / Asterisk terminate   Benny Amorsen
• [asterisk-users] Re: Range Operator   Benny Amorsen
• [asterisk-users] Re: Multiple TE110P cards in one chassis   Benny Amorsen
• [asterisk-users] Re: How big is *your* dialplan??   Benny Amorsen
• [asterisk-users] Re: E164 caller ID   Benny Amorsen
• [asterisk-users] Re: SIP stuck channel soft hangup?   Benny Amorsen
• [asterisk-users] Re: Snom 320, Queues and Transfer not working as expected with * 1.2.12.1   Benny Amorsen
• [asterisk-users] Snom, mute and rtptimeout   Benny Amorsen
• [asterisk-users] Re: How to do Automatic Daylight Saving on Grandstream GXP-2000   Benny Amorsen
• [asterisk-users] Re: Realtime in the Real World   Benny Amorsen
• [asterisk-users] Re: SIP RTP flow   Benny Amorsen
• [asterisk-users] Asterisk on virtual machine   Ryan Amos
• [asterisk-users] chan_skype license?   Andreas Anderson
• [asterisk-users] 480i phone: Is there a trick to registering with *?? <--Solved   Colin Anderson
• [asterisk-users] 480i phone: Is there a trick to registering with *?? <--Solved, first impressions   Colin Anderson
• [asterisk-users] t1 voip to failover pri   Colin Anderson
• [asterisk-users] CALEA support within asterisk?   Colin Anderson
• [asterisk-users] DISA and legacy PBX   Colin Anderson
• [asterisk-users] Zaptel problems   Colin Anderson
• [asterisk-users] Dialplan Syslog   Colin Anderson
• [asterisk-users] Transfer feature - howto?   Colin Anderson
• [asterisk-users] Transfer feature - howto?   Colin Anderson
• [asterisk-users] Echo problems on ISDN. (mainly incoming call s)   Colin Anderson
• [asterisk-users] DOA IAXy?   Erik Anderson
• [asterisk-users] Extremely choppy sound on some of our POTS network calls; goes away with mute   Andres
• [asterisk-users] authenticating forwarded calls   Andres
• [asterisk-users] plainvoip - down ???   Andres
• [asterisk-users] Polycom 2.01 sip issues   James Andrewartha
• [asterisk-users] Polycom IP 501 phone randomly resets itself (loses Received call log, Missed calls, placed calls)   James Andrewartha
• [asterisk-users] WiFi SIP handset with Bluetooth required   Cory Andrews
• [asterisk-users] RE: Asterisk Advice   Cory Andrews
• [asterisk-users] Clipcomm CG 410 / FXO Gateway   Cory Andrews
• [asterisk-users] VoipSupply? [Semi-Urgent]   Cory Andrews
• [asterisk-users] Embedded Asterisk   Cory Andrews
• [asterisk-users] Trixbox installation - ZAP channels becoming upresponsive   Cory Andrews
• [SPAM] - [asterisk-users] Looking for Wireless Heaset for Polycom 501 - Email found in subject   Cory Andrews
• [SPAM HEADER] - Re: [asterisk-users] Snom or Cisco Phones? - Email found in subject   Cory Andrews
• [asterisk-users] How to add new codec support?   Rawat Anshuman-cbf738
• [asterisk-users] asterisk guru needed for job in Chicago area   Vitalie Apostu
• [asterisk-users] IVR   Vitalie Apostu
• [asterisk-users] NAT/firewall/Asterisk/Polycom Phones   Michael Araba
• [asterisk-users] Callmanager 3.3(5) and Asterisk with ooh323   Michael Araba
• [asterisk-users] RE: ECHO Cancellation in SIP Calls   Michael Araba
• [asterisk-users] Defining sip users through mysql   Arkaitz
• [asterisk-users] Re: Defining sip users through mysql   Arkaitz
• [asterisk-users] asterisk not detecting hangup   Arkaitz
• [asterisk-users] asterisk not detecting hangup   Arkaitz
• [asterisk-users] Dial-9 (was Extension Numbering)   Jay R. Ashworth
• [asterisk-users] Dial-9 (was Extension Numbering)   Jay R. Ashworth
• [asterisk-users] Fax detection ...   Jay R. Ashworth
• [asterisk-users] Fax detection ...   Jay R. Ashworth
• [asterisk-users] G726 prompts   Jay R. Ashworth
• [asterisk-users] t1 voip to failover pri   Jay R. Ashworth
• [asterisk-users] Fax detection ...   Jay R. Ashworth
• [asterisk-users] Bandwidth usage   Jay R. Ashworth
• [Asterisk-Users] problem configuring a digium quad E1 card   Jay R. Ashworth
• [asterisk-users] CALEA support within asterisk?   Jay R. Ashworth
• [asterisk-users] CALEA support within asterisk?   Jay R. Ashworth
• [asterisk-users] CALEA support within asterisk?   Jay R. Ashworth
• [asterisk-users] CALEA support within asterisk?   Jay R. Ashworth
• [asterisk-users] Building a terrorist-friendly telephone network (Was: CALEA support)   Jay R. Ashworth
• [asterisk-users] Digium TDM or SPA-3000?   Jay R. Ashworth
• [asterisk-users] Digium TDM or SPA-3000?   Jay R. Ashworth
• [asterisk-users] Call Interception   Jay R. Ashworth
• [asterisk-users] TNT Max Password reset   Jay R. Ashworth
• [asterisk-users] Wouldn't Tri-tone detection in Dial() be cool?   Jay R. Ashworth
• [asterisk-users] MODEM (data) througt asterisk ?   Jay R. Ashworth
• [asterisk-users] TNT Max Password reset   Jay R. Ashworth
• [asterisk-users] TNT Max Password reset   Jay R. Ashworth
• [asterisk-users] no callerid from PSTN using TDM2400P   Jay R. Ashworth
• [asterisk-users] snom 360: how to make record button working ?   Jay R. Ashworth
• [asterisk-users] Bandwidth requirements   Jay R. Ashworth
• [asterisk-users] Spelling   Jay R. Ashworth
• [asterisk-users] OT: Polycom time sync - sorta   Jay R. Ashworth
• [asterisk-users] Re: PoE IP Phone   Jay R. Ashworth
• [asterisk-users] Asterisk Server : IDE HDD frequent crash   Jay R. Ashworth
• [asterisk-users] Re: Where is the PlayDTMF command?   Jay R. Ashworth
• [asterisk-users] Re: Asterisk Server : IDE HDD frequent crash   Jay R. Ashworth
• [asterisk-users] snom 360: how to make record button working ?   Jay R. Ashworth
• [asterisk-users] No Dialtone   Jay R. Ashworth
• [asterisk-users] No Dialtone   Jay R. Ashworth
• [asterisk-users] Re: PRI issues   Jay R. Ashworth
• [asterisk-users] isdn cross-over ...   Jay R. Ashworth
• [asterisk-users] ftp server   Jay R. Ashworth
• [asterisk-users] snom 360: how to make record button working ?   Jay R. Ashworth
• [asterisk-users] Re: PRI issues   Jay R. Ashworth
• [asterisk-users] RE: Welcome to the "asterisk-users" mailing list   Jay R. Ashworth
• [asterisk-users] How big is *your* dialplan??   Jay R. Ashworth
• [asterisk-users] How big is *your* dialplan??   Jay R. Ashworth
• [asterisk-users] How big is *your* dialplan??   Jay R. Ashworth
• [asterisk-users] How big is *your* dialplan??   Jay R. Ashworth
• [asterisk-users] call takeover?   Jay R. Ashworth
• [asterisk-users] Psst... Top secret information: Codename Pineapple   Jay R. Ashworth
• [asterisk-users] MGCP stuff - VoiPACK   Jay R. Ashworth
• [asterisk-users] cisco 7960 not registering after * restart   Jay R. Ashworth
• [asterisk-users] asterisk 1.2.12 lost phone registrations today... why?   Jay R. Ashworth
• [asterisk-users] asterisk 1.2.12 lost phone registrations today... why?   Jay R. Ashworth
• [asterisk-users] How big is *your* dialplan??   Jay R. Ashworth
• [asterisk-users] Psst... Top secret information: Codename Pineapple   Jay R. Ashworth
• [asterisk-users] OT: Voipsupply.com phones are down. Was: how big is *your* dialplan   Jay R. Ashworth
• [asterisk-users] two SIP phones as one line   Jay R. Ashworth
• [asterisk-users] DID failover   Jay R. Ashworth
• [asterisk-users] two SIP phones as one line   Jay R. Ashworth
• [asterisk-users] tdm2400p question   Jay R. Ashworth
• [asterisk-users] VoipSupply? [Semi-Urgent]   Jay R. Ashworth
• [asterisk-users] Why is this happening?   Jay R. Ashworth
• [asterisk-users] Stopping putgoing calls after working hours   Jay R. Ashworth
• [asterisk-users] Linksys PAP2 dial plan help please   Jay R. Ashworth
• [asterisk-users] Embedded Asterisk   Jay R. Ashworth
• [asterisk-users] wrong outgoing caller id with PRI lines: maybe usecallingpres involved?   Jay R. Ashworth
• [asterisk-users] "Fixing the Caller-ID Problem", by John Todd for O'ReillyNet   Jay R. Ashworth
• [asterisk-users] "Fixing the Caller-ID Problem", by John Todd for O'ReillyNet   Jay R. Ashworth
• [asterisk-users] "Fixing the Caller-ID Problem", by John Todd for O'ReillyNet   Jay R. Ashworth
• [asterisk-users] FXO Cards vs. Channel bank with T1   Jay R. Ashworth
• [asterisk-users] Asterisk@Home problems   Lacy Moore - Aspendora
• [asterisk-users] Re: PoE IP Phone   Lacy Moore - Aspendora
• [asterisk-users] Asterisk Configuration Complete Newbie question   Lacy Moore - Aspendora
• [asterisk-users] Cisco CCM - Asterisk   Lacy Moore - Aspendora
• [asterisk-users] Cisco CCM - Asterisk   Lacy Moore - Aspendora
• [asterisk-users] How big is *your* dialplan??   Lacy Moore - Aspendora
• [asterisk-users] How big is *your* dialplan??   Lacy Moore - Aspendora
• [asterisk-users] XO SIP Origination Services   Lacy Moore - Aspendora
• [asterisk-users] How big is *your* dialplan??   Lacy Moore - Aspendora
• [asterisk-users] Stopping putgoing calls after working hours   Lacy Moore - Aspendora
• [asterisk-users] VoipSupply? [Semi-Urgent]   Lacy Moore - Aspendora
• [asterisk-users] getting DID info..   Lacy Moore - Aspendora
• [asterisk-users] getting DID info..   Lacy Moore - Aspendora
• [asterisk-users] getting DID info..   Lacy Moore - Aspendora
• [asterisk-users] VoicePulse Connect 4 Channel Limit?   Lacy Moore - Aspendora
• [asterisk-users] Digium vs. Sangoma   Lacy Moore - Aspendora
• [asterisk-users] Asterisk Manager   Lacy Moore - Aspendora
• [asterisk-users] Pager Voicemail Message   Lacy Moore - Aspendora
• [asterisk-users] Newbie Questions   Lacy Moore - Aspendora
• [asterisk-users] Got reject for frame XX, retransmitting frame XX now, updating n_r!   Asterisk
• [asterisk-users] Distributing calls among channels in dial group   Asterisk
• [asterisk-users] channel.c: Unable to request channel ZAP   Asterisk
• [asterisk-users] Call Center requirements   Todd Houle Asterisk
• [asterisk-users] DID failover   Todd- Asterisk
• [asterisk-users] Re: 1.4 on mac OSX 10.4.8   Todd- Asterisk
• [asterisk-users] getting DID info..   Todd- Asterisk
• [asterisk-users] getting DID info..   Todd- Asterisk
• [asterisk-users] getting DID info..   Todd- Asterisk
• [asterisk-users] getting DID info..   Todd- Asterisk
• [asterisk-users] Asterisk and Attachment   Giedrius Augys
• [asterisk-users] Spandsp and tif   Giedrius Augys
• [asterisk-users] Spandsp and tif   Giedrius Augys
• [asterisk-users] Sun Cluster and Asterisk   Giedrius Augys
• [asterisk-users] Sipura SPA-481   Giedrius Augys
• [asterisk-users] Spandsp and tif   Giedrius Augys
• [asterisk-users] spandsp and freebsd   Giedrius Augys
• [asterisk-users] spandsp and freebsd   Giedrius Augys
• [asterisk-users] spandsp bug   Giedrius Augys
• [asterisk-users] translate.c:88 powerof: Powerof 0: No power??   Giedrius Augys
• [asterisk-users] translate.c:88 powerof: Powerof 0: No power??   Giedrius Augys
• [asterisk-users] Getting Asterisk to work with GoogleTalk   Alvin Austin
• [asterisk-users] RE: Getting Asterisk to work with GoogleTalk   Alvin Austin
• [asterisk-users] Re: Bandwidth requirements   Dan Austin
• [asterisk-users] modprobe Ztdummy is not working   Dan Austin
• [asterisk-users] Meetme... No channel type registered for 'zap'   Dan Austin
• [asterisk-users] Callmanager 3.3(5) and Asterisk with ooh323   Dan Austin
• [asterisk-users] Callmanager 3.3(5) and Asterisk with ooh323   Dan Austin
• [asterisk-users] Callmanager 3.3(5) and Asterisk with ooh323   Dan Austin
• [asterisk-users] Callmanager 3.3(5) and Asterisk with ooh323   Dan Austin
• [Asterisk-Users] External Custom Extension Timeout   JD Austin
• [asterisk-users] Call Forwarding not working for extension in queue, why?   Dovid B
• [asterisk-users] Newbie h/w Q, and confirming basic concepts   Dovid B
• [asterisk-users] Re: failed registration   Dovid B
• [asterisk-users] ftp server   Dovid B
• [asterisk-users] 2x* and realtime   Dovid B
• [asterisk-users] SIP vs. SIP-B   Dovid B
• [asterisk-users] single conference, multiple numbers   Dovid B
• [asterisk-users] Echo Cancel Cards   Dovid B
• [asterisk-users] Inbound Callcenter with multiple DIDs   Dovid B
• [asterisk-users] Echo Cancel Cards   Dovid B
• [asterisk-users] sequential Dial() commands   Dovid B
• [asterisk-users] transfer from VM to Cell Phone   Dovid B
• [asterisk-users] How big is *your* dialplan??   Dovid B
• [asterisk-users] Connection question...   Dovid B
• [asterisk-users] Change the background of a conversation   Dovid B
• [asterisk-users] Extension and Voice Mail setup   Dovid B
• [asterisk-users] max users   Dovid B
• [asterisk-users] Asterisk users help   Dovid B
• [asterisk-users] How big is *your* dialplan??   Dovid B
• [asterisk-users] Issues with Asterisk 1.4 Beta   Dovid B
• [asterisk-users] How do you like TrixBox?   Dovid B
• [asterisk-users] How do you like TrixBox?   Dovid B
• [asterisk-users] detecting the receivers voicemail   Dovid B
• [asterisk-users] VoicePulse Connect 4 Channel Limit?   Dovid B
• [asterisk-users] Remote UNIX connection, Remote UNIX disconnected displayed every second   Dovid B
• [asterisk-users] considering purchasing a t1 card, any recommendations?   Dovid B
• [asterisk-users] Findme problem   Dovid B
• [asterisk-users] Findme problem   Dovid B
• [asterisk-users] plainvoip - down ???   Dovid B
• [asterisk-users] Using variable as a context extension ?   Dovid B
• [asterisk-users] embedded asterisk   Dovid B
• [asterisk-users] How to deploy a PBX in such a condition ?   Dovid B
• [asterisk-users] CID Issues   Dovid B
• [asterisk-users] Where to best start looking for voicemail/moh sound quality problem?   Dovid B
• [asterisk-users] Becoming a User on IRC   Dovid B
• [asterisk-users] voicemail idea and a question   Dovid B
• [asterisk-users] need help using tftp for polycom 501   Dovid B
• [asterisk-users] Broadvoice incoming DTMF problems   Dovid B
• [asterisk-users] Call is not coming through sipgate.co.uk+Asterisk   Dovid B
• [asterisk-users] simple dialplan trick I can't figure out (smdi, mwi substitute)   Dovid B
• [asterisk-users] Asterisk behind NAT and without portforwarding forrtp   Dovid B
• [asterisk-users] Asterisk behind NAT and without portforwarding forrtp   Dovid B
• [asterisk-users] How to make different ext using different trunks?   Dovid B
• [asterisk-users] Pager Voicemail Message   Dovid B
• [asterisk-users] Re: [OT] wi-fi ip phone scenario   Dovid B
• [asterisk-users] Read cmd   Dovid B
• [asterisk-users] Setting up UTStarcom F300   Dovid B
• [asterisk-users] light web user interface   Dovid B
• [asterisk-users] Multiple dial macros at the same time   Dovid B
• [asterisk-users] auto recording extensions   Dovid B
• [asterisk-users] FXO Cards vs. Channel bank with T1   Dovid B
• [asterisk-users] FXO Card's vs. T1   Dovid B
• [asterisk-users] FXO Cards vs. Channel bank with T1   Dovid B
• [asterisk-users] NAT issue ?   Dovid B
• [asterisk-users] simultaneous ring - call groups or queues orsomething else?   Dovid B
• [asterisk-users] simultaneous ring - call groups or queues orsomething else?   Dovid B
• [asterisk-users] NAT issue ? [More Info]   Dovid B
• [asterisk-users] Newbie Questions   Dovid B
• [asterisk-users] Re: Newbie Questions   Dovid B
• [asterisk-users] T1 Passthrough   Vlad B
• [asterisk-users] MODEM (data) througt asterisk ?   ABC- Florent BARBIER
• [asterisk-users] Asterisk is overwriting proxy Via Header   Fernando BERRETTA
• [asterisk-users] Bristuff vs. vISDN vs. mISDN for hfc card ?   Michiel van Baak
• [asterisk-users] Spying a channel in a meetme   Michiel van Baak
• AW: [asterisk-users] PoE IP Phone   Michiel van Baak
• [asterisk-users] GPL Softphones   Michiel van Baak
• [asterisk-users] How big is *your* dialplan??   Michiel van Baak
• [asterisk-users] Psst... Top secret information: Codename Pineapple   Michiel van Baak
• [asterisk-users] Centos kernel 34 vs. 42?   Michiel van Baak
• [asterisk-users] Newbie question about meetme   Michiel van Baak
• [asterisk-users] bristuff-0.3.0-PRE-1u for Asterisk 1.2.13 on junghanns downloads now   Michiel van Baak
• [asterisk-users] bristuff-0.3.0-PRE-1u for Asterisk 1.2.13 on junghanns downloads now   Michiel van Baak
• [asterisk-users] bristuff-0.3.0-PRE-1u for Asterisk 1.2.13 on junghanns downloads now   Michiel van Baak
• [asterisk-users] bristuff-0.3.0-PRE-1u for Asterisk 1.2.13 on junghanns downloads now   Michiel van Baak
• [asterisk-users] bristuff-0.3.0-PRE-1u for Asterisk 1.2.13 on junghanns downloads now   Michiel van Baak
• [asterisk-users] bristuff-0.3.0-PRE-1u for Asterisk 1.2.13 on junghanns downloads now   Michiel van Baak
• [asterisk-users] VoiceOne 0.4.0 released: a new web-based and open source GUI   Michiel van Baak
• [asterisk-users] Re: Asterisk Manager   Michiel van Baak
• [asterisk-users] Multiple queue_log files based on queue - is it possible??   Michiel van Baak
• [asterisk-users] Re: Meetme... No channel type registered for 'zap'   Michiel van Baak
• [asterisk-users] Re: Meetme... No channel type registered for 'zap'   Michiel van Baak
• [Asterisk-Users] chan_capi and bristuff   Michiel van Baak
• [Asterisk-Users] chan_capi and bristuff   Michiel van Baak
• [Asterisk-Users] chan_capi and bristuff   Michiel van Baak
• [Asterisk-Users] Would you support a Bristuff mailing list ?   Michiel van Baak
• [asterisk-users] VoIP GSM Gateways   Michiel van Baak
• [asterisk-users] No ring tone when using IAX   Michiel van Baak
• [asterisk-users] IPv6   David Bandel
• [asterisk-users] Call Interception   Time Bandit
• [asterisk-users] Dialplan Syslog   Time Bandit
• [asterisk-users] MODEM (data) througt asterisk ?   Time Bandit
• [asterisk-users] pop a web page with DID in url   Time Bandit
• [asterisk-users] OT: Hand free solution recommandation   Time Bandit
• [asterisk-users] Remote UNIX connection, Remote UNIX disconnected displayed every second   Time Bandit
• [asterisk-users] Why is this happening?   Time Bandit
• [asterisk-users] Why is this happening?   Time Bandit
• [asterisk-users] Why is this happening?   Time Bandit
• [asterisk-users] Why is this happening?   Time Bandit
• [asterisk-users] Getting started with sample dial plans   Time Bandit
• [asterisk-users] voicemail usernames can't begin with "j" letter?   Time Bandit
• [asterisk-users] operator console   Time Bandit
• [asterisk-users] Wildcard X100P Suport   Time Bandit
• [asterisk-users] Asterisk web interface is not parsing the PHP pages   Time Bandit
• [asterisk-users] Compatability   Time Bandit
• [asterisk-users] No ring tone when using IAX   Time Bandit
• [asterisk-users] One way audio on chan_gtalk   Gustavo Hernandez Baratta
• [asterisk-users] One way audio on chan_gtalk   Gustavo Hernandez Baratta
• [asterisk-users] snom 360: how to make record button working ?   Remco Barendse
• [asterisk-users] snom 360: how to make record button working ?   Remco Barendse
• [asterisk-users] snom 360: how to make record button working ?   Remco Barendse
• [asterisk-users] Asterisk 1.2.12.1 dies after asterisk -rx and never comes up again   Remco Barendse
• [asterisk-users] Asterisk 1.2.12.1 dies after asterisk -rx and never comes up again   Remco Barendse
• [asterisk-users] Asterisk 1.2.12.1 dies after asterisk -rx and never comes up again   Remco Barendse
• [asterisk-users] 1.2.12.1 crashing   Remco Barendse
• [asterisk-users] Centos kernel 34 vs. 42?   Remco Barendse
• [asterisk-users] Centos kernel 34 vs. 42?   Remco Barendse
• [asterisk-users] Re: Centos kernel 34 vs. 42? [was: asterisk-users Digest, Vol 27, Issue 72]   Remco Barendse
• [asterisk-users] bristuff-0.3.0-PRE-1u for Asterisk 1.2.13 on junghanns downloads now   Remco Barendse
• [asterisk-users] bristuff-0.3.0-PRE-1u for Asterisk 1.2.13 on junghanns downloads now   Remco Barendse
• [asterisk-users] bristuff-0.3.0-PRE-1u for Asterisk 1.2.13 on junghanns downloads now   Remco Barendse
• [asterisk-users] bristuff-0.3.0-PRE-1u for Asterisk 1.2.13 on junghanns downloads now   Remco Barendse
• [asterisk-users] bristuff-0.3.0-PRE-1u for Asterisk 1.2.13 on junghanns downloads now   Remco Barendse
• [asterisk-users] Why does it take at least 4 flipping days before asterisk tries to resolve a provider?   Remco Barendse
• [asterisk-users] Why does it take at least 4 flipping days before asterisk tries to resolve a provider?   Remco Barendse
• [asterisk-users] bristuff-0.3.0-PRE-1u for Asterisk 1.2.13 on junghanns downloads now   Remco Barendse
• [asterisk-users] Strange FXS disconnection problem.   David Bath
• [asterisk-users] Strange FXS disconnection problem.   David Bath
• [asterisk-users] Strange FXS disconnection problem.   David Bath
• [asterisk-users] Strange FXS disconnection problem.   David Bath
• [asterisk-users] Strange FXS disconnection problem.   David Bath
• [asterisk-users] Strange FXS disconnection problem.   David Bath
• [asterisk-users] Strange FXS disconnection problem.   David Bath
• [asterisk-users] Zap disconnect   David Bath
• [asterisk-users] SIP with Qualify and NAT   David Bath
• *****SPAM***** [asterisk-users] asterisk 1.4 problem with call queues   Dean Bath
• [asterisk-users] Unknown signalling method 'pri_cpe'   Forrest Beck
• [asterisk-users] IP Phones   Forrest Beck
• [asterisk-users] Voicemail and Forwarding   Forrest Beck
• [asterisk-users] Re: Voicemail and Forwarding   Forrest Beck
• [asterisk-users] Echo Cancel Cards   Forrest Beck
• [asterisk-users] T1 Passthrough   Forrest Beck
• [asterisk-users] Echo Cancel Cards   Forrest Beck
• [asterisk-users] Asterisk as SIP Client   Forrest Beck
• [asterisk-users] Switchtype,Signalling,rxwink warnings   Forrest Beck
• [asterisk-users] Polycom HDVoice   Forrest Beck
• [asterisk-users] AGI Help   Forrest Beck
• [asterisk-users] Strange Characters in CLI on TTY9   Forrest Beck
• [asterisk-users] Asterisk integration   Brian Becker
• [asterisk-users] quality control   Juraj Bednar
• [asterisk-users] lots of registrations, sip problem   Juraj Bednar
• [asterisk-users] asterisk not detecting hangup   Maxi Belino
• [asterisk-users] Re: Centos kernel 34 vs. 42? [was: asterisk-users Digest, Vol 27, Issue 72]   Les Bell
• [asterisk-users] T1 incoming connects, but no sound   Nathan Bell
• [asterisk-users] SIP stuck channel soft hangup?   Nic Bellamy
• [asterisk-users] Lots and lots of log files   Nic Bellamy
• [asterisk-users] Re: SIP stuck channel soft hangup?   Nic Bellamy
• [asterisk-users] Grandstream ATA 286 tdm400 and Asterisk 1.2-13   Nic Bellamy
• [asterisk-users] Iax bug ?   Jean-Baptiste Bellet
• [asterisk-users] Iax bug ?   Jean-Baptiste Bellet
• [asterisk-users] Iax bug ?   Jean-Baptiste Bellet
• [asterisk-users] Iax bug ?   Jean-Baptiste Bellet
• [asterisk-users] show logged clients   Jean-Baptiste Bellet
• [asterisk-users] verbose logging to file in 1.4   Benko
• [asterisk-users] Re: verbose logging to file in 1.4   Benko
• [asterisk-users] connecting multiple servers with iax - authentication fails   Benko
• [asterisk-users] Re: connecting multiple servers with iax - authentication fails   Benko
• [asterisk-users] Re: Reception Console   Aldo Bergamini
• [asterisk-users] Odd echo issue with speaker phone   BerkHolz, Steven
• [asterisk-users] Odd echo issue with speaker phone   BerkHolz, Steven
• [asterisk-users] Inhouse SIP to ZAP has echo sometimes.   BerkHolz, Steven
• [asterisk-users] some transfers dropped.   BerkHolz, Steven
• [asterisk-users] dial D option with w for wait?   BerkHolz, Steven
• [asterisk-users] SIP Locking Up?   Jeremy Betts
• [asterisk-users] Polycom Buddy Watch Broken with 2.0.1 Firmware?   Eric Bishop
• [asterisk-users] Polycom Buddy Watch Setup help request   Eric Bishop
• [asterisk-users] FOP run control for CentOS/RHEL   Eric Bishop
• [asterisk-users] Fax receive (rx fax) problem   Gareth Blades
• [asterisk-users] Cisco 2621 NM-HDV VWIC-1MFT1   Steve Blair
• [asterisk-users] Speed Dials   Al Bochter
• [asterisk-users] Findme problem   Al Bochter
• [asterisk-users] G.729 operating on outgoing only   Al Bochter
• [asterisk-users] Where to best start looking for voicemail/moh sound quality problem?   Al Bochter
• [asterisk-users] Broadvoice incoming DTMF problems   Al Bochter
• [asterisk-users] Broadvoice incoming DTMF problems   Al Bochter
• [asterisk-users] SIP v IAX2   Al Bochter
• [asterisk-users] DTMF detection problem in PABX trunk   Al Bochter
• [asterisk-users] No ring tone when using IAX   Al Bochter
• [asterisk-users] detecting the receivers voicemail   Leo Ann Boon
• [asterisk-users] wrong outgoing caller id with PRI lines: maybe usecallingpres involved?   Leo Ann Boon
• [asterisk-users] wrong outgoing caller id with PRI lines: maybe usecallingpres involved?   Leo Ann Boon
• [asterisk-users] compiling libunicall   Leo Ann Boon
• [asterisk-users] asterisk and HMP   Leo Ann Boon
• [asterisk-users] Quintum DX as gateway to PSTN for Asterisk   Leo Ann Boon
• [asterisk-users] "No Authority Found"   Leo Ann Boon
• [asterisk-users] Is SQLite a replacement for Mysql while using ARA in 1.2.x   Leo Ann Boon
• [asterisk-users] SIP problem - ACT p160s error   Leo Ann Boon
• [asterisk-users] CID and CDR conflict?   Leo Ann Boon
• [asterisk-users] Architecture for Asterisk   Leo Ann Boon
• [asterisk-users] Asterisk both behind a NAT and outside at the same time   Leo Ann Boon
• [asterisk-users] S(x) - Hang up the call after 'x' seconds - Not working from queue   Leo Ann Boon
• [asterisk-users] Asterisk both behind a NAT and outside at the same time   Leo Ann Boon
• [asterisk-users] S(x) - Hang up the call after 'x' seconds - Not working from queue   Leo Ann Boon
• [asterisk-users] IAX2 outgoing calls delayed before they connect   Stephen Bosch
• [asterisk-users] Polycom IP 501 message light   Stephen Bosch
• [asterisk-users] Polycom IP 501 message light   Stephen Bosch
• [asterisk-users] Voicemail prompts clipped when retrieving from some SIP phones   Stephen Bosch
• [asterisk-users] Voicemail prompts clipped when retrieving from some SIP phones   Stephen Bosch
• [asterisk-users] simultaneous ring - call groups or queues or something else?   Stephen Bosch
• [asterisk-users] simultaneous ring - call groups or queues or something else?   Stephen Bosch
• [asterisk-users] Asterisk act as a proxy ?   Peter Bowyer
• [asterisk-users] Incoming sip line with INX (internationalnumber.com)   Peter Bowyer
• [asterisk-users] Asterisk 1.2.12 - Can NOT make call out / Asterisk terminate   Peter Bowyer
• [asterisk-users] Asterisk 1.2.12 - Can NOT make call out / Asterisk terminate   Peter Bowyer
• [asterisk-users] Incoming sip line with INX(internationalnumber.com)   Peter Bowyer
• [asterisk-users] Asterisk@Home problems   Peter Bowyer
• [asterisk-users] asterisk upgrade   Peter Bowyer
• [asterisk-users] Strange problem(Munin-node-1.2.4-7)   Crazy Boy
• [asterisk-users] New tutorial - peering two * servers using IAX   Crazy Boy
• [asterisk-users] no callerid from PSTN using TDM2400P   Crazy Boy
• [asterisk-users] Requirements for Asterisk & SER integration   Crazy Boy
• [asterisk-users] DID is not working (call is not routing)   Crazy Boy
• [asterisk-users] DID is not working (call is not routing)   Crazy Boy
• [asterisk-users] DID is not working (call is not routing)   Crazy Boy
• [asterisk-users] Anybody using "inphonex" service?   Crazy Boy
• [asterisk-users] VoIP+RJ11 Phone existed?   Crazy Boy
• [asterisk-users] List of VoIP+RJ11 Phones   Crazy Boy
• [asterisk-users] Reception Console   Crazy Boy
• [asterisk-users] DID is not working (call is not routing)   Crazy Boy
• [asterisk-users] Asterisk+SER help   Crazy Boy
• [asterisk-users] Call is not coming through sipgate.co.uk+Asterisk   Crazy Boy
• [asterisk-users] Is it possible to connect two servers using SIP?   Crazy Boy
• [asterisk-users] VoIP GSM Gateways   Matteo Brancaleoni
• [asterisk-users] VoIP GSM Gateways   Matteo Brancaleoni
• [asterisk-users] How big is *your* dialplan??   Anders Brander
• [asterisk-users] cisco 2600   Tijl Van den Broeck
• [asterisk-users] cisco 2600   Tijl Van den Broeck
• [asterisk-users] Cisco 2621 NM-HDV VWIC-1MFT1   Tijl Van den Broeck
• [asterisk-users] Asterisk Realtime... Help Me!!!   Tijl Van den Broeck
• [asterisk-users] Linksys PAP2 dial plan help please   Tijl Van den Broeck
• [asterisk-users] Asterisk Realtime... Help Me!!!   Tijl Van den Broeck
• [asterisk-users] Waiting before executing System command   Alexander Burke
• [asterisk-users] Waiting before executing System command   Alexander Burke
• [asterisk-users] Trunks and Outbound Routes   Dakota Burns
• [asterisk-users] Wildcard X100P Suport   Michael C. Cambria
• [asterisk-users] extensions.conf strangeness   Brian Candler
• [asterisk-users] newbie: intelligent handling of two ISDN lines (4 B-channels)   Brian Candler
• [asterisk-users] T38Passthrough and 1.4 Beta   Brian Candler
• [asterisk-users] "set verbose 4" in SVN trunk?   Brian Candler
• [asterisk-users] Bandwidth requirements   Brian Candler
• [asterisk-users] "set verbose 4" in SVN trunk?   Brian Candler
• [asterisk-users] Re: extensions.conf strangeness   Brian Candler
• [asterisk-users] Bandwidth requirements   Brian Candler
• [asterisk-users] Re: extensions.conf strangeness   Brian Candler
• [asterisk-users] Newbie h/w Q, and confirming basic concepts   Brian Candler
• [asterisk-users] Re: verbose logging to file in 1.4   Brian Candler
• [asterisk-users] Asterisk RT on Disk On Module Performance and Durability   Brian Candler
• [asterisk-users] VOIP with PSTN backup   Brian Candler
• [asterisk-users] VOIP with PSTN backup   Brian Candler
• [asterisk-users] isdn cross-over ...   Brian Candler
• [asterisk-users] Understanding NAT Traversal   Brian Candler
• [asterisk-users] Understanding NAT Traversal   Brian Candler
• [asterisk-users] Extension and Voice Mail setup   Brian Candler
• [asterisk-users] Understanding NAT Traversal   Brian Candler
• [asterisk-users] Call bridged, but no sound   Brian Candler
• [asterisk-users] Call bridged, but no sound   Brian Candler
• [asterisk-users] Connection question...   Brian Candler
• [asterisk-users] VoIP+RJ11 Phone existed?   Brian Candler
• [asterisk-users] GPL Softphones   Brian Candler
• [asterisk-users] Psst... Top secret information: Codename Pineapple   Brian Candler
• [asterisk-users] VoIP+RJ11 Phone existed?   Brian Candler
• [asterisk-users] Call bridged, but no sound   Brian Candler
• [asterisk-users] Multiple 'routes' to extension in different contextes. How to influence search oder?   Brian Candler
• [asterisk-users] Remote UNIX connection, Remote UNIX disconnected displayed every second   Brian Candler
• [asterisk-users] Why is this happening?   Brian Candler
• [asterisk-users] chan_bluetooth, mobile handset as VoIP terminal?   Brian Candler
• [asterisk-users] Why is this happening?   Brian Candler
• [asterisk-users] Orange Flash Light Mitel 5215 - Asterisk - working !   Brian Candler
• [asterisk-users] random one way audio and noise between SIP phones on same LAN   Brian Candler
• [asterisk-users] Asterisk+SER help   Brian Candler
• [asterisk-users] Why is this happening?   Brian Candler
• [asterisk-users] Linksys PAP2 dial plan help please   Brian Candler
• [asterisk-users] Linksys PAP2 dial plan help please   Brian Candler
• [asterisk-users] spandsp and freebsd   Brian Candler
• [Asterisk-Users] rxfax problem   Brian Candler
• [asterisk-users] UA - number assignment   Brian Candler
• [asterisk-users] UA - number assignment   Brian Candler
• [asterisk-users] Call is not coming through sipgate.co.uk+Asterisk   Brian Candler
• [asterisk-users] ASterisk Start problem   Brian Candler
• [asterisk-users] Simple example for call transfer.   Brian Candler
• [asterisk-users] Add second account to Xlite 3.0   Brian Candler
• [asterisk-users] Re: Real-time and priority "n"   Brian Capouch
• [asterisk-users] RE: Welcome to the "asterisk-users" mailing list   Brian Capouch
• [asterisk-users] How big is *your* dialplan??   Brian Capouch
• [asterisk-users] Reception Console   Brian Capouch
• [asterisk-users] VoipSupply? [Semi-Urgent]   Brian Capouch
• [asterisk-users] Why is this happening?   Brian Capouch
• [asterisk-users] Netgear WGT Flash-fest at Astricon   Brian Capouch
• [asterisk-users] Netgear WGT Flash-fest at Astricon   Brian Capouch
• [asterisk-users] Embedded Asterisk   Brian Capouch
• [asterisk-users] "No Authority Found"   Brian Capouch
• [asterisk-users] Anyone got a dialplan for SPA ATAs for ISN?   Brian Capouch
• [asterisk-users] OT: jobs for asterisk lovers   michel Carette
• [asterisk-users] Compiling Zaptel 1.2.10 on Ubuntu 6.10   Strom Carlson
• [asterisk-users] voicemail usernames can't begin with "j" letter?   Ricardo Carvalho
• [asterisk-users] SIP_HEADER function; what names are available?   Ricardo Carvalho
• [asterisk-users] voicemail usernames can't begin with "j" letter?   Ricardo Carvalho
• [asterisk-users] voicemail usernames can't begin with "j" letter?   Ricardo Carvalho
• [asterisk-users] anti ex-girlfriend   Ricardo Carvalho
• [asterisk-users] Modifying SIP Stack   German Castro
• [asterisk-users] Minexpiry time - how to set this   Cavanna, Richard
• [asterisk-users] tdm2400p question   Cavanna, Richard
• [asterisk-users] Macro 'exited non-zero'   Anthony Cennami
• [asterisk-users] 7960 (8.2) - Call Center - REBOOT   Anthony Cennami
• [asterisk-users] isdn cross-over ...   Cesc
• [asterisk-users] isdn cross-over ...   Cesc
• [asterisk-users] Xorcom Astribank and 64 bit linux   Carlos Chavez
• [asterisk-users] Xorcom Astribank and 64 bit linux   Carlos Chavez
• [asterisk-users] Problem compiling libmfcr2.0.0.2 on Fedora Core 5   Carlos Chavez
• [asterisk-users] Error loading Unicall   Carlos Chavez
• [asterisk-users] Error loading Unicall   Carlos Chavez
• [asterisk-users] Locking phones at night...   Carlos Chavez
• [asterisk-users] DTMF problems with legacy PBX   Carlos Chavez
• [asterisk-users] Without ZapTel inferface or Card install , is Conference working or Not   Carlos Chavez
• [asterisk-users] New Asterisk-GUI?   Carlos Chavez
• [asterisk-users] Billing   Khaled Chehab
• [asterisk-users] Urgent Please help   Khaled Chehab
• [asterisk-users] Urgent Please help   Khaled Chehab
• [asterisk-users] Urgent Billing   Khaled Chehab
• [asterisk-users] no callerid from PSTN using TDM2400P   Bob Chiodini
• [asterisk-users] asterisk 1.2.12 lost phone registrations today... why?   Bob Chiodini
• [asterisk-users] asterisk 1.2.12 lost phone registrations today... why?   Bob Chiodini
• [asterisk-users] Looking for a Voicemail Lamp device   Bob Chiodini
• [asterisk-users] Looking for a Voicemail Lamp device   Bob Chiodini
• [asterisk-users] Electric usage of a tdm400p   Bob Chiodini
• [asterisk-users] Electric usage of a tdm400p   Bob Chiodini
• [asterisk-users] Where is the PlayDTMF command?   Frank Church
• [asterisk-users] Where is the PlayDTMF command?   Frank Church
• [asterisk-users] Re: Where is the PlayDTMF command?   Frank Church
• [asterisk-users] Re: Where is the PlayDTMF command?   Frank Church
• [asterisk-users] Re: Where is the PlayDTMF command?   Frank Church
• [asterisk-users] Re: Where is the PlayDTMF command?   Frank Church
• [asterisk-users] Re: Where is the PlayDTMF command?   Frank Church
• [asterisk-users] Calls being disconnected across VPN   Frank Church
• [asterisk-users] RE: Welcome to the "asterisk-users" mailing list   Mike Clark
• [asterisk-users] Is 1.2.12.1 production ready   Mike Clark
• [asterisk-users] Occasional one-way audio - Sangoma A101   Mike Clark
• [asterisk-users] Occasional one-way audio - Sangoma A101   Mike Clark
• [asterisk-users] Building a terrorist-friendly telephone network (Was: CALEA support)   Henry J. Cobb
• [asterisk-users] [Asterisk-Java] SipShowPeerAction   richard Coco
• [asterisk-users] [Asterisk-Java] SipShowPeerAction   richard Coco
• [asterisk-users] Unknown signalling method 'pri_cpe'   Tzafrir Cohen
• [asterisk-users] astcc help-pleasssssseeee   Tzafrir Cohen
• [asterisk-users] SIP client that runs on Linux or Solaris through X Windows?   Tzafrir Cohen
• [asterisk-users] Voicemail maintenance   Tzafrir Cohen
• [asterisk-users] Asterisk CDR   Tzafrir Cohen
• [asterisk-users] Xorcom Astribank and 64 bit linux   Tzafrir Cohen
• [asterisk-users] ZAP chanel doesn't reset if external caller hangs up in menu   Tzafrir Cohen
• [asterisk-users] ftp server   Tzafrir Cohen
• [asterisk-users] Asterisk RT on Disk On Module Performance and Durability   Tzafrir Cohen
• [asterisk-users] Lots and lots of log files   Tzafrir Cohen
• [asterisk-users] Lots and lots of log files   Tzafrir Cohen
• [asterisk-users] Asterisk on 64bit xeon   Tzafrir Cohen
• [asterisk-users] single conference, multiple numbers   Tzafrir Cohen
• [asterisk-users] Strange FXS disconnection problem.   Tzafrir Cohen
• [asterisk-users] Asterisk 1.2.12.1 dies after asterisk -rx and never comes up again   Tzafrir Cohen
• [asterisk-users] Asterisk 1.2.12.1 dies after asterisk -rx and never comes up again   Tzafrir Cohen
• [asterisk-users] GPL Softphones   Tzafrir Cohen
• [asterisk-users] zt_chanconfig failed   Tzafrir Cohen
• [asterisk-users] Strange FXS disconnection problem.   Tzafrir Cohen
• [asterisk-users] zt_chanconfig failed   Tzafrir Cohen
• [asterisk-users] Test Call Script   Tzafrir Cohen
• [asterisk-users] Re: Re: Voicemail Press '0'   Tzafrir Cohen
• [asterisk-users] Strange FXS disconnection problem.   Tzafrir Cohen
• [asterisk-users] Re: 1.4 beta2 on intel mac   Tzafrir Cohen
• [asterisk-users] Re: 1.4 beta2 on intel mac   Tzafrir Cohen
• [asterisk-users] Remote UNIX connection, Remote UNIX disconnected displayed every second   Tzafrir Cohen
• Asterisk crashes with ealtime mysql configuration [was: Re: [asterisk-users] Please help me!!]   Tzafrir Cohen
• [asterisk-users] ISDN PRI and R2 Trunks in one Dual Port Card   Tzafrir Cohen
• [asterisk-users] Remote UNIX connection, Remote UNIX disconnected displayed every second   Tzafrir Cohen
• [asterisk-users] Locking phones at night...   Tzafrir Cohen
• [asterisk-users] Digium Single Span Card Installation   Tzafrir Cohen
• [asterisk-users] Locking phones at night...   Tzafrir Cohen
• [asterisk-users] Locking phones at night...   Tzafrir Cohen
• [asterisk-users] Digium Single Span Card Installation   Tzafrir Cohen
• [asterisk-users] Zaptel not detecting Tormenta2 PRI Interface card   Tzafrir Cohen
• [asterisk-users] bristuff-0.3.0-PRE-1u for Asterisk 1.2.13 on junghanns downloads now   Tzafrir Cohen
• [asterisk-users] Bristuff qozap drivers problem   Tzafrir Cohen
• [asterisk-users] Help: Problems about console color (FC5, XTerm)   Tzafrir Cohen
• [asterisk-users] /dev/zap/channel ownership   Tzafrir Cohen
• [asterisk-users] modprobe Ztdummy is not working   Tzafrir Cohen
• [asterisk-users] modprobe Ztdummy is not working   Tzafrir Cohen
• [asterisk-users] bristuff-0.3.0-PRE-1u for Asterisk 1.2.13 on junghanns downloads now   Tzafrir Cohen
• [asterisk-users] bristuff-0.3.0-PRE-1u for Asterisk 1.2.13 on junghanns downloads now   Tzafrir Cohen
• [asterisk-users] bristuff-0.3.0-PRE-1u for Asterisk 1.2.13 on junghanns downloads now   Tzafrir Cohen
• [asterisk-users] bristuff-0.3.0-PRE-1u for Asterisk 1.2.13 on junghanns downloads now   Tzafrir Cohen
• [asterisk-users] new g.729a codecs for asterisk 1.2/1.4 and glibc   Tzafrir Cohen
• [asterisk-users] Can't change Zaptel driver from FCC mode   Tzafrir Cohen
• [asterisk-users] Unicall Installation   Tzafrir Cohen
• [asterisk-users] Zap Channel and VM problem   Tzafrir Cohen
• [asterisk-users] Can't change Zaptel driver from FCC mode   Tzafrir Cohen
• [asterisk-users] chan_h323.so Asterisk Beta compilation   Tzafrir Cohen
• [asterisk-users] Asterisk and dialer Running on Thin Clients   Tzafrir Cohen
• [asterisk-users] Becoming a User on IRC   Tzafrir Cohen
• [asterisk-users] ASterisk Start problem   Tzafrir Cohen
• [asterisk-users] Re: Meetme... No channel type registered for 'zap'   Tzafrir Cohen
• [asterisk-users] SIP v IAX2   Tzafrir Cohen
• [Asterisk-Users] chan_capi and bristuff   Tzafrir Cohen
• [asterisk-users] ECHO Cancellation in SIP Calls   Tzafrir Cohen
• [asterisk-users] Is SQLite a replacement for Mysql while using ARA in 1.2.x   Tzafrir Cohen
• [asterisk-users] Re: ECHO Cancellation in SIP Calls   Tzafrir Cohen
• [asterisk-users] Re: Re: ECHO Cancellation in SIP Calls   Tzafrir Cohen
• [asterisk-users] Is SQLite a replacement for Mysql while using ARA in 1.2.x   Tzafrir Cohen
• [asterisk-users] Asterisk and ISDN and Hylafax   Tzafrir Cohen
• [asterisk-users] ISDN-BRI issue   Tzafrir Cohen
• [asterisk-users] Re: 0 channels configured with tdm400 (tdm04b rev. G)   Tzafrir Cohen
• [asterisk-users] bristuff-0.3.0-PRE-1u for Asterisk 1.2.13 on junghanns downloads now   Tzafrir Cohen
• [asterisk-users] app_meetme not loading   Tzafrir Cohen
• [asterisk-users] VoIP GSM Gateways   Tzafrir Cohen
• [asterisk-users] No zap* commands?   Tzafrir Cohen
• [asterisk-users] Incorrect Ring tone. Getting a US tone when it should be AU tone   Tzafrir Cohen
• [asterisk-users] asterisk-1.2.13 fails to 'Make' in Fedore Core 6'   Tzafrir Cohen
• [asterisk-users] IVR   Tzafrir Cohen
• [asterisk-users] Grandstream ATA 286 tdm400 and Asterisk 1.2-13   Tzafrir Cohen
• [asterisk-users] Live creation of trunk groups   Tzafrir Cohen
• [asterisk-users] Strange Characters in CLI on TTY9   Tzafrir Cohen
• [asterisk-users] Asterisk manager   Dean Collins
• [asterisk-users] Video Conference   Dean Collins
• [asterisk-users] Video Conference   Dean Collins
• [asterisk-users] ftp server   Dean Collins
• [asterisk-users] polycom auto cfg file   Dean Collins
• [asterisk-users] RE: ftp server   Dean Collins
• [asterisk-users] ftp server   Dean Collins
• [asterisk-users] ftp server   Dean Collins
• [asterisk-users] ftp server   Dean Collins
• [asterisk-users] polycom reboot script   Dean Collins
• [asterisk-users] AstriCon Dallas in Two Weeks   Dean Collins
• [asterisk-users] How big is *your* dialplan??   Dean Collins
• [asterisk-users] RE: Welcome to the "asterisk-users" mailing list   Dean Collins
• [asterisk-users] Change the background of a conversation   Dean Collins
• [asterisk-users] RE: Welcome to the "asterisk-users" mailing list   Dean Collins
• [asterisk-users] 1.4 beta2 on intel mac   Dean Collins
• [asterisk-users] What alternatives to Asterisk based virtual PBX?   Dean Collins
• [asterisk-users] AstriCon Hotel Full - Here are some near-byalternates   Dean Collins
• [asterisk-users] How big is *your* dialplan?? MTBF   Dean Collins
• [asterisk-users] unauthenticated calls   Dean Collins
• [asterisk-users] Polycom IP650   Dean Collins
• [asterisk-users] list down?   Dean Collins
• [asterisk-users] Astricon - post show Saturday?   Dean Collins
• [asterisk-users] Polycom boot error   Dean Collins
• [asterisk-users] Embedded Asterisk   Dean Collins
• [asterisk-users] Polycom boot error   Dean Collins
• [asterisk-users] Polycom provision errors still! Arg!   Dean Collins
• [asterisk-users] Polycom provision errors still! Arg!   Dean Collins
• [asterisk-users] Polycom provision errors still! Arg!   Dean Collins
• [asterisk-users] [OT] wi-fi ip phone scenario   Dean Collins
• [asterisk-users] Looking for Wireless Heaset for Polycom 501   Dean Collins
• [asterisk-users] light web user interface   Dean Collins
• [asterisk-users] Astricon followup   Dean Collins
• [asterisk-users] How big is *your* dialplan??   Michael Collins
• [asterisk-users] How big is *your* dialplan??   Michael Collins
• [asterisk-users] How big is *your* dialplan?? MTBF   Michael Collins
• [asterisk-users] How big is *your* ego?   Michael Collins
• [asterisk-users] Re: How do you like TrixBox?   Michael Collins
• [asterisk-users] whisper paging   Joshua Colp
• [asterisk-users] connecting multiple servers with iax - authentication fails   Joshua Colp
• [asterisk-users] Voicemail Press '0'   Joshua Colp
• [asterisk-users] Re: Is 1.2.12.1 production ready (Mauro Zanin)   Joshua Colp
• [asterisk-users] 1.4 branch on OSX?   Joshua Colp
• [asterisk-users] Re: 1.4 branch on OSX?   Joshua Colp
• [asterisk-users] Cisco 7970 SIP won't update?   Tim Connolly
• [asterisk-users] Monitor stops recording midstream?   Tim Connolly
• [asterisk-users] Using variable as a context extension ?   Andre Courchesne - Consultant
• [asterisk-users] Using variable as a context extension ?   Andre Courchesne - Consultant
• [asterisk-users] Live creation of trunk groups   Andre Courchesne - Consultant
• [asterisk-users] Live creation of trunk groups   Andre Courchesne - Consultant
• [asterisk-users] FOP v.27 IAX trunks not "ringing"   David Cook
• [asterisk-users] 7960/SIP MWI Question   David Cook
• [asterisk-users] Issues with calling certain phone numbers...   Luca Corti
• [asterisk-users] PoE IP Phone   Dave Cotton
• [asterisk-users] Asterisk 1.2.12 - Can NOT make call out / Asterisk terminate   Dave Cotton
• [asterisk-users] asterisk 1.2.12 lost phone registrations today... why?   Dave Cotton
• [asterisk-users] asterisk 1.2.12 lost phone registrations today... why?   Dave Cotton
• [asterisk-users] asterisk 1.2.12 lost phone registrations today... why?   Dave Cotton
• [asterisk-users] asterisk 1.2.12 lost phone registrations today... why?   Dave Cotton
• [asterisk-users] SPA 3102   Dave Cotton
• [asterisk-users] SPA 3102   Dave Cotton
• [asterisk-users] Why does it take at least 4 flipping days before asterisk tries to resolve a provider?   Dave Cotton
• [asterisk-users] SIP v IAX2   Dave Cotton
• [asterisk-users] SIP v IAX2   Dave Cotton
• [asterisk-users] Re: SIP v IAX2   Dave Cotton
• [asterisk-users] Resampling Audio for use with Asterisk   Nate Criss
• [asterisk-users] Re: Bandwidth requirements   Matthew Crocker
• [asterisk-users] Problem with ZAPTEL-1.4.0-beta1 and WCT100P card   Matthew Crocker
• [asterisk-users] Problem with ZAPTEL-1.4.0-beta1 and WCT100P card   Matthew Crocker
• [asterisk-users] PRI boards with g729 capable DSPs   Matthew Crocker
• [asterisk-users] PRI boards with g729 capable DSPs   Matthew Crocker
• [asterisk-users] IVR for the called part (IVR inside out)   Daniel Cyt
• [asterisk-users] Incoming sip line with INX (internationalnumber.com)   Daniel Cyt
• [asterisk-users] Incoming sip line with INX(internationalnumber.com)   Daniel Cyt
• [asterisk-users] Incoming sip line withINX(internationalnumber.com)   Daniel Cyt
• [asterisk-users] Warning translate.c:88 powerof   Daniel Cyt
• [asterisk-users] how to config chanspy   Sergio R. D'Ippolito
• [asterisk-users] Registration problem   Sergio R. D'Ippolito
• [asterisk-users] VoicePulse Connect 4 Channel Limit?   DM
• [asterisk-users] Spandsp and tif   DRi at b-w-computer.de
• [asterisk-users] Astricon followup   Dal
• [asterisk-users] Voicemail MWI   Aaron Daniel
• [asterisk-users] HTTP Connection Closed on 7960 SIP   Aaron Daniel
• [asterisk-users] Asterisk RT on Disk On Module Performance andDurability   Aaron Daniel
• [asterisk-users] password for vm users   Aaron Daniel
• [asterisk-users] How big is *your* dialplan??   Aaron Daniel
• [asterisk-users] Increase VoiceMail Messages Recording Gain - AudioCalls are Ok   Aaron Daniel
• [asterisk-users] Increase VoiceMail Messages Recording Gain - AudioCalls are Ok   Aaron Daniel
• [asterisk-users] cisco 7960 not registering after * restart   Aaron Daniel
• [asterisk-users] How big is *your* dialplan??   Aaron Daniel
• [asterisk-users] cisco 7960 not registering after * restart   Aaron Daniel
• [asterisk-users] Digium on Dell PowerEdge 1850   Aaron Daniel
• [asterisk-users] Multiple line phones with different contexts   Aaron Daniel
• [asterisk-users] Digium vs. Sangoma   Aaron Daniel
• AW: [asterisk-users] Snom or Cisco Phones?   Aaron Daniel
• [asterisk-users] sending fax with chan-capi   Klaus Darilion
• [asterisk-users] sending fax with chan-capi   Klaus Darilion
• [asterisk-users] identifying Eicon Diva Server V-4BRI-8M vs 4BRI-8M   Klaus Darilion
• [asterisk-users] random one way audio and noise between SIP phones on same LAN   Klaus Darilion
• [asterisk-users] identifying Eicon Diva Server V-4BRI-8M vs 4BRI-8M   Klaus Darilion
• [asterisk-users] Problems with chan-capi and Eicon Diva 4BRI   Klaus Darilion
• [Asterisk-Users] rxfax problem   Klaus Darilion
• [asterisk-users] question about asterisk txFAX   Darren
• [asterisk-users] AEL2 and the variables   Dominique Dartois
• [asterisk-users] Can't Register Client - Multiple Subnets   Big Wave Dave
• [asterisk-users] Pager Voicemail Message   David
• [asterisk-users] H.263 Video Messages   David
• [asterisk-users] Pager Voicemail Message   David
• [asterisk-users] Pager Voicemail Message   David
• [asterisk-users] Re: Where to best start looking for voicemail/moh sound quality problem?   Paul Davidson
• [asterisk-users] Inaccurate CDRs   Steve Davies
• [asterisk-users] Bristuff qozap drivers problem   Steve Davies
• [asterisk-users] Bristuff qozap drivers problem   Steve Davies
• [Asterisk-Users] rxfax problem   Steve Davies
• [asterisk-users] considering purchasing a t1 card, any recommendations?   Steve Davies
• [Asterisk-Users] rxfax problem   Steve Davies
• [asterisk-users] spandsp and freebsd   Steve Davies
• [asterisk-users] PBAX-Group with QuadBRI card, outgoing call problem   Steve Davies
• [asterisk-users] Mac OS X Desktop / Asterisk integration?   Steve Davies
• [asterisk-users] Mac OS X Desktop / Asterisk integration?   Steve Davies
• [asterisk-users] How big is *your* dialplan??   Peter J Dean
• [asterisk-users] VoIP GSM Gateways   Peter J Dean
• [asterisk-users] Asterisk -> regitration in DB   Thomas Deillon
• [asterisk-users] Call Interception   Delca
• [asterisk-users] Asterisk Billing Plataforms   Delca
• [asterisk-users] Screen pop based on incoming DID   Greg Delgado
• SV: [asterisk-users] Screen pop based on incoming DID   Joe Dennick
• [asterisk-users] PoE IP Phone   Joe Dennick
• [asterisk-users] GPL Softphones   Joe Dennick
• [asterisk-users] GPL Softphones   Joe Dennick
• [asterisk-users] what hardware and is it possible   Joe Dennick
• [asterisk-users] call center status viewer   Joe Dennick
• [asterisk-users] Server Recommendations   Joe Dennick
• [asterisk-users] Snom or Cisco Phones?   Joe Dennick
• [asterisk-users] GotoIfTime - much slowdown with 90 conditions?   Mike Dent
• [asterisk-users] What alternatives to Asterisk based virtual PBX?   Mike Dent
• [asterisk-users] Snom or Cisco Phones?   Mike Dent
• [asterisk-users] PBAX-Group with QuadBRI card, outgoing call problem   Giray Devlet
• [asterisk-users] PBAX-Group with QuadBRI card, outgoing call problem   Giray Devlet
• [asterisk-users] zt_chanconfig failed   DiegoF
• [asterisk-users] compiling libunicall   DiegoF
• [asterisk-users] zt_chanconfig failed   DiegoF
• [asterisk-users] zt_chanconfig failed   DiegoF
• [asterisk-users] compiling libunicall   DiegoF
• [asterisk-users] compiling libunicall   DiegoF
• [asterisk-users] OriginateEvent reason codes.   DiegoF
• [asterisk-users] compiling libunicall   DiegoF
• [asterisk-users] What doe these error messages mean?   Mike Diehl
• [asterisk-users] Embedded Asterisk   Mike Diehl
• [asterisk-users] Embedded Asterisk   Mike Diehl
• [asterisk-users] CID and CDR conflict?   Mike Diehl
• [asterisk-users] CID and CDR conflict?   Mike Diehl
• [asterisk-users] "set verbose 4" in SVN trunk?   Pryakhin Dimitry
• [asterisk-users] "set verbose 4" in SVN trunk?   Pryakhin Dimitry
• [asterisk-users] Silience on random calls   Pryakhin Dimitry
• [asterisk-users] Call Interception   Don
• [asterisk-users] TNT Max Password reset   Don
• [asterisk-users] max users   Don
• [asterisk-users] Multiple TE110P cards in one chassis   Don
• [asterisk-users] Re: Multiple TE110P cards in one chassis   Don
• [asterisk-users] Multiple TE110P cards in one chassis   Don
• [asterisk-users] sip realtime broken?   Don
• [asterisk-users] sip realtime broken?   Don
• [asterisk-users] IP Phones   Doug
• [asterisk-users] Polycom 601 & Expansion Module: Light the LEDs???   Doug
• [asterisk-users] VoipSupply? [Semi-Urgent]   Doug
• [asterisk-users] ZyXEL desktop ethernet switch for QoS   Janto Dreijer
• [asterisk-users] GPL Softphones   Gregory Duchatelet
• [asterisk-users] GPL Softphones   Gregory Duchatelet
• [asterisk-users] GPL Softphones   Gregory Duchatelet
• [asterisk-users] acami   Gregory Duchatelet
• [asterisk-users] say Asterisk to answer   Gregory Duchatelet
• [asterisk-users] using asterisk to do remote control functions   Gregory Duchatelet
• [asterisk-users] asterisk and HMP   Gregory Duchatelet
• [asterisk-users] asterisk and HMP   Gregory Duchatelet
• [asterisk-users] asterisk and HMP   Gregory Duchatelet
• [asterisk-users] Polycom Buddy Watch Broken with 2.0.1 Firmware?   Paul Dugas
• [asterisk-users] Polycom Buddy Watch Broken with 2.0.1 Firmware?   Paul Dugas
• [asterisk-users] Polycom Call Parking   Paul Dugas
• [asterisk-users] Polycom Call Parking   Paul Dugas
• [asterisk-users] CBeyond SIP   Paul Dugas
• [asterisk-users] Fax detection ...   Michelle Dupuis
• [asterisk-users] Blacklist to check http://whocalled.us   Michelle Dupuis
• [asterisk-users] OT: Hand free solution recommandation   Michelle Dupuis
• [asterisk-users] VoipSupply? [Semi-Urgent]   Michelle Dupuis
• [Asterisk-Users] rxfax problem   Michelle Dupuis
• [asterisk-users] anti ex-girlfriend   Michelle Dupuis
• [asterisk-users] Asterisk Server : IDE HDD frequent crash   Dushyanth
• [asterisk-users] Critical - No audio issue with re-invite (wrong media address)   Hoa Thai Duy
• [asterisk-users] Weird problem with beep.wav!   James Dyer
• [asterisk-users] CDR stats to one mysql database, multiple webstats packages   Chris Earle
• [asterisk-users] ftp server   Gary Eck
• [asterisk-users] ftp server   Gary Eck
• [asterisk-users] ftp server   Gary Eck
• [asterisk-users] Asterisk@Home problems   Edward0219 at aol.com
• [asterisk-users] Asterisk@Home problems   Edward0219 at aol.com
• [asterisk-users] Asterisk@Home problems   Edward0219 at aol.com
• [asterisk-users] Asterisk@Home problems   Edward0219 at aol.com
• [asterisk-users] Asterisk@Home problems   Edward0219 at aol.com
• [asterisk-users] Cisco 2621 NM-HDV VWIC-1MFT1   David Edwards
• [asterisk-users] Cisco 2621 NM-HDV VWIC-1MFT1   David Edwards
• [asterisk-users] Cisco 2621 NM-HDV VWIC-1MFT1   David Edwards
• [asterisk-users] Cisco 2621 NM-HDV VWIC-1MFT1   David Edwards
• [asterisk-users] Call Interception   Steve Edwards
• [asterisk-users] Dialplan Syslog   Steve Edwards
• [asterisk-users] How big is *your* dialplan??   Steve Edwards
• [asterisk-users] Asterisk (meetme) and SMP/HT OK?   Steve Edwards
• [asterisk-users] Generate Random Numbers in dialplan   Steve Edwards
• [asterisk-users] Generate Random Numbers in dialplan   Steve Edwards
• [asterisk-users] Re: Asterisk (meetme) and SMP/HT OK?   Steve Edwards
• [asterisk-users] Re: Asterisk (meetme) and SMP/HT OK? Lots 'o crashes   Steve Edwards
• [asterisk-users] How do I configure Asterisk if I need to run Mysql server on second Linux   Steve Edwards
• [asterisk-users] Unique call ID's across several systems   Steve Edwards
• [asterisk-users] Unique call ID's across several systems   Steve Edwards
• [asterisk-users] Caller ID on Zap not always working   Nick Ellson
• [asterisk-users] Outbound FXO call, getting "You must first dial..."   Nick Ellson
• [asterisk-users] Outbound FXO call, getting "You must first dial..."   Nick Ellson
• [asterisk-users] asterisk guru needed for job in Chicago area   Elvar
• [asterisk-users] DTMF logging   Scott England
• [asterisk-users] Bandwidth requirements   Erik
• [asterisk-users] Server power indication   Erik
• [asterisk-users] Server power indication   Erik
• [asterisk-users] Reception Console   Damon Estep
• [asterisk-users] overlap of zap trunk groups   Damon Estep
• [asterisk-users] overlap of zap trunk groups   Damon Estep
• [asterisk-users] GPL Softphones   Dumpolid Exeplish
• [asterisk-users] vGSM drivers updated (0.17.2)   Dumpolid Exeplish
• [asterisk-users] Inaccurate CDRs   Dumpolid Exeplish
• [asterisk-users] Inaccurate CDRs   Dumpolid Exeplish
• [asterisk-users] Inaccurate CDRs   Dumpolid Exeplish
• [asterisk-users] Inaccurate CDRs   Dumpolid Exeplish
• [asterisk-users] Authenticate application   Dumpolid Exeplish
• [asterisk-users] Inaccurate CDRs   Dumpolid Exeplish
• [asterisk-users] AGI for authenticating calls with DTMF   Dumpolid Exeplish
• [asterisk-users] Inaccurate CDRs   Dumpolid Exeplish
• [asterisk-users] TDM2400P wiring.   C F
• [asterisk-users] TDM2400P wiring.   C F
• [asterisk-users] TDM2400P wiring.   C F
• [asterisk-users] whisper paging   C F
• [asterisk-users] whisper paging   C F
• [asterisk-users] whisper paging   C F
• [asterisk-users] whisper paging   C F
• [asterisk-users] How big is *your* dialplan??   C F
• [asterisk-users] RE: Welcome to the "asterisk-users" mailing list   C F
• [asterisk-users] How big is *your* dialplan??   C F
• [asterisk-users] How big is *your* dialplan??   C F
• [asterisk-users] How big is *your* dialplan??   C F
• [asterisk-users] How big is *your* ego?   C F
• [asterisk-users] How big is *your* dialplan??   C F
• [asterisk-users] How big is *your* ego?   C F
• [asterisk-users] OT: Voipsupply.com phones are down. Was: how big is *your* dialplan   C F
• [asterisk-users] tdm2400p question   C F
• [asterisk-users] VoipSupply? [Semi-Urgent]   C F
• [asterisk-users] Locking phones at night...   C F
• [asterisk-users] "Fixing the Caller-ID Problem", by John Todd for O'ReillyNet   C F
• [asterisk-users] blind transfers with IP Polycom 501   C F
• [asterisk-users] Asterisk and Panasonic KX Model   C F
• [asterisk-users] Asterisk both behind a NAT and outside at the same time   C F
• [asterisk-users] Asterisk both behind a NAT and outside at the same time   C F
• [asterisk-users] Asterisk both behind a NAT and outside at the same time   C F
• [asterisk-users] Asterisk and Panasonic KX Model   C F
• [asterisk-users] Call drop and strange CDR records   CAHEN Fabrice
• [asterisk-users] Reception Console   Jon Farmer
• [asterisk-users] Registration problem   Jon Farmer
• [asterisk-users] Call Quality / Echo / Problems   Barry Fawthrop
• [asterisk-users] Clipcomm CG 410 / FXO Gateway   Barry Fawthrop
• [asterisk-users] AstFax Sending a Fax   Barry Fawthrop
• [asterisk-users] AstFax Sending a Fax   Barry Fawthrop
• [asterisk-users] Asterisk hangs up on incoming analog calls after a while   Robert La Ferla
• [asterisk-users] Asterisk hangs up on incoming analog calls after a while   Robert La Ferla
• [asterisk-users] (no subject)   Robert La Ferla
• [asterisk-users] voicemail idea and a question   Robert La Ferla
• [asterisk-users] Polycom provision errors still! Arg!   Ivan Fetch
• [asterisk-users] No zap* commands?   Barbara Figueirido
• [asterisk-users] Match & Chat Author?   Bart Fisher
• [asterisk-users] WiFi SIP handset with Bluetooth required   Zeb Fletcher
• [asterisk-users] Digium Interfaces in Tampa?   Matt Florell
• [asterisk-users] answering machine detection   Matt Florell
• [asterisk-users] 1.2.12.1 crashing   Matt Florell
• [asterisk-users] 1.2.12.1 crashing   Matt Florell
• [asterisk-users] Digium TE410P LED problem   Matt Florell
• [asterisk-users] Re: Digium TE410P LED problem   Matt Florell
• [asterisk-users] Re: Digium TE410P LED problem   Matt Florell
• [asterisk-users] Unique call ID's across several systems   Matt Florell
• [asterisk-users] Unique call ID's across several systems   Matt Florell
• [asterisk-use