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-users] Asterisk and dialer Running on Thin Clients   Matt Florell
• [asterisk-users] Unique call ID's across several systems   Matt Florell
• [asterisk-users] dialing external number within meetme   Matt Florell
• [asterisk-users] T1 pricing in Oz   Forum
• [asterisk-users] T1 pricing in Oz   Forum
• [asterisk-users] VoIP GSM Gateways   Forum
• [asterisk-users] VoIP GSM Gateways   Forum
• [asterisk-users] spandsp logs?   Francesco Francesconi
• [asterisk-users] Message count requested for mailbox 9002@from-sip but voicemail not loaded.   Francesco Francesconi
• [asterisk-users] No Dialtone   Francesco Francesconi
• [asterisk-users] Redefinition of transfer   Francois
• [asterisk-users] Bridging of PRI calls   Matthew Fredrickson
• [asterisk-users] Looking for Wireless Heaset for Polycom 501   Jim Freeze
• [asterisk-users] Looking for Wireless Heaset for Polycom 501   Jim Freeze
• [asterisk-users] Looking for Wireless Heaset for Polycom 501   Jim Freeze
• [asterisk-users] Asterisk + E1 with MFC/R2 in Argentina?   Guillermo Freige
• [asterisk-users] attended transfer unreliable (2nd try)   Stefan Friedrich
• [asterisk-users] OT: Polycom time sync - sorta   Dave Fullerton
• [asterisk-users] sequential Dial() commands   Dave Fullerton
• [asterisk-users] Asterisk architecure   G(P)L
• [asterisk-users] Fedora Core 6 (FC6) and Asterisk-1.2.13 and Zaptel-1.2.10 compile problems   Michael J. Tubby G8TIC
• [asterisk-users] Windows and file shares   Paul Gaffney
• [asterisk-users] IAX softphones   Paul Gaffney
• [asterisk-users] Access Denied on a Windows share   Paul Gaffney
• [asterisk-users] G729 Codec Loading Error   David Gagnon
• [asterisk-users] How to get the agent id in the recording filename   David Gagnon
• [asterisk-users] CALEA support within asterisk?   Brandon Galbraith
• [asterisk-users] CALEA support within asterisk?   Brandon Galbraith
• [asterisk-users] Home Hardware SIP Proxy with use of POTS in Emergency   Brandon Galbraith
• [asterisk-users] Home Hardware SIP Proxy with use of POTS in Emergency   Brandon Galbraith
• [asterisk-users] Polycom IP 501 phone randomly resets itself (loses Received call log, Missed calls, placed calls)   Mike Garey
• [asterisk-users] Polycom IP 501 phone randomly resets itself (loses Received call log, Missed calls, placed calls)   Mike Garey
• [asterisk-users] Polycom IP 501 phone randomly resets itself (loses Received call log, Missed calls, placed calls)   Mike Garey
• [asterisk-users] Queue AddQueueMember()   Douglas Garstang
• [asterisk-users] Passing Arguments to FastAGI   Douglas Garstang
• [asterisk-users] Polycom Buddy Watch Broken with 2.0.1 Firmware?   Douglas Garstang
• [asterisk-users] AstmanProxy Not Collating Manager Info   Douglas Garstang
• [asterisk-users] Re: Passing Arguments to FastAGI   Douglas Garstang
• [asterisk-users] Asterisk 1.4 moh - mohsuggest   Douglas Garstang
• [asterisk-users] New tutorial - peering two * servers using IAX   Douglas Garstang
• [asterisk-users] Dialplan Syslog   Douglas Garstang
• [asterisk-users] New tutorial - peering two * servers using IAX   Douglas Garstang
• [asterisk-users] Dialplan Syslog   Douglas Garstang
• [asterisk-users] Calling Functions from AEL2   Douglas Garstang
• [asterisk-users] AEL2 #include madness in Asterisk 1.4 - Murf?   Douglas Garstang
• [asterisk-users] Re: Calling Functions from AEL2   Douglas Garstang
• [asterisk-users] How to forward DID to another Server   Douglas Garstang
• [asterisk-users] Voicemail MWI   Douglas Garstang
• [asterisk-users] AGI() in 1.2 and 1.4   Douglas Garstang
• [asterisk-users] ChanIsAvail() in 1.2.12.1   Douglas Garstang
• [asterisk-users] ChanIsAvail() in 1.2.12.1   Douglas Garstang
• [asterisk-users] Range Operator   Douglas Garstang
• [asterisk-users] Number Range   Douglas Garstang
• [asterisk-users] Asterisk RT on Disk On Module Performance andDurability   Douglas Garstang
• [asterisk-users] Asterisk RT on Disk On Module PerformanceandDurability   Douglas Garstang
• [asterisk-users] Function ENUMLOOKUP   Douglas Garstang
• [asterisk-users] Asterisk RT on Disk On Module PerformanceandDurability   Douglas Garstang
• [asterisk-users] whisper paging   Douglas Garstang
• [asterisk-users] Voicemail Press '0'   Douglas Garstang
• [asterisk-users] whisper paging   Douglas Garstang
• [asterisk-users] Voicemail Press '0'   Douglas Garstang
• [asterisk-users] Voicemail Press '0'   Douglas Garstang
• [asterisk-users] Re: Voicemail Press '0'   Douglas Garstang
• [asterisk-users] Re: Voicemail Press '0'   Douglas Garstang
• [asterisk-users] Re: Re: Voicemail Press '0'   Douglas Garstang
• [asterisk-users] How big is *your* dialplan??   Douglas Garstang
• [asterisk-users] RE: Welcome to the "asterisk-users" mailing list   Douglas Garstang
• [asterisk-users] RE: Welcome to the "asterisk-users" mailing list   Douglas Garstang
• [asterisk-users] How big is *your* dialplan??   Douglas Garstang
• [asterisk-users] How big is *your* dialplan??   Douglas Garstang
• [asterisk-users] How big is *your* dialplan??   Douglas Garstang
• [asterisk-users] How big is *your* dialplan??   Douglas Garstang
• [asterisk-users] How big is *your* dialplan??   Douglas Garstang
• [asterisk-users] How big is *your* dialplan??   Douglas Garstang
• [asterisk-users] Macro 'exited non-zero'   Douglas Garstang
• [asterisk-users] Macro 'exited non-zero'   Douglas Garstang
• [asterisk-users] Meetme... No channel type registered for 'zap'   Douglas Garstang
• [asterisk-users] Meetme... No channel type registered for 'zap'   Douglas Garstang
• [asterisk-users] Re: Meetme... No channel type registered for 'zap'   Douglas Garstang
• [asterisk-users] Re: Meetme... No channel type registered for'zap'   Douglas Garstang
• [asterisk-users] Re: Meetme... No channel type registered for 'zap'   Douglas Garstang
• [asterisk-users] Re: Meetme... No channel type registered for 'zap'   Douglas Garstang
• [asterisk-users] SIP v IAX2   Douglas Garstang
• [asterisk-users] SIP v IAX2   Douglas Garstang
• [asterisk-users] Taking a Polycom IP601 home   Douglas Garstang
• [asterisk-users] Confused about SIP Realtime Updates   Douglas Garstang
• [asterisk-users] Voicemail 'exitcontext'   Douglas Garstang
• [asterisk-users] Realtime in the Real World   Douglas Garstang
• [asterisk-users] asterisk-oh323   Christian Gatti
• [asterisk-users] pbx call setup to asterisk, behavior context dependend   Christian Gatti
• [asterisk-users] uniden uip200 phone hangs any ideas?   Jerry Geis
• [asterisk-users] asterisk 1.2.12 lost phone registrations today... why?   Jerry Geis
• [asterisk-users] centos or rhel and txfax with libtiff   Jerry Geis
• [asterisk-users] txfax only getting 1 page of 3.   Jerry Geis
• [asterisk-users] tx_fax not getting entire fax   Jerry Geis
• [asterisk-users] Cisco CCM - Asterisk   Phil Genera
• [asterisk-users] noise gate for asterisk?   Phil Genera
• [asterisk-users] call takeover?   Csibra Gergo
• [asterisk-users] call takeover?   Csibra Gergo
• [asterisk-users] SPA 3102   Csibra Gergo
• [asterisk-users] Linksys PAP2 dial plan help please   Csibra Gergo
• [asterisk-users] Linksys PAP2: calling tone stops after 5 tones   Csibra Gergo
• [asterisk-users] newbie: intelligent handling of two ISDN lines (4 B-channels)   Ekkard Gerlach
• [asterisk-users] Asterisk manager   Bill Gibbs
• [asterisk-users] OT: Polycom time sync - sorta   Bill Gibbs
• [asterisk-users] question about IF   Bill Gibbs
• [asterisk-users] How to do Automatic Daylight Saving on Grandstream GXP-2000   Drew Gibson
• [asterisk-users] How to do Automatic Daylight Saving on Grandstream GXP-2000   Drew Gibson
• [asterisk-users] T1 pricing in Oz   Dennis Gilmore
• [asterisk-users] RE:Asterisk and dialer Running on Thin Clients   Jean-Denis Girard
• [asterisk-users] t1 voip to failover pri   Steve Glaus
• [asterisk-users] Bandwidth usage   Steve Glaus
• [asterisk-users] asterisk to asterisk DID extentions   Steve Glaus
• [asterisk-users] Strange problem(Munin-node-1.2.4-7)   Steve Glaus
• [asterisk-users] Need USA DID + trunk provider   Steve Glaus
• [asterisk-users] Transfer feature - howto?   Steve Glaus
• [asterisk-users] Transfer feature - howto?   Steve Glaus
• [asterisk-users] IP Phones   Steve Glaus
• [asterisk-users] Re: [asteisk-users]USA DID + trunk   Steve Glaus
• [asterisk-users] Re: [asteisk-users]USA DID + trunk   Steve Glaus
• [asterisk-users] Bandwidth requirements   Steve Glaus
• [asterisk-users] Transfer feature - howto?   Steve Glaus
• [asterisk-users] PoE IP Phone   Steve Glaus
• [asterisk-users] Options for moving to * friendly Business VSP   Gleim, Jason
• [asterisk-users] Blank page when sending faxes   Nick Glencross
• [asterisk-users] Blank page when sending faxes (repost)   Nick Glencross
• [asterisk-users] Re: Blank page when sending faxes (repost)   Nick Glencross
• [asterisk-users] How do you like TrixBox?   Mohamed A. Gombolaty
• [asterisk-users] Stopping putgoing calls after working hours   Mohamed A. Gombolaty
• [asterisk-users] Stopping putgoing calls after working hours   Mohamed A. Gombolaty
• [asterisk-users] Stopping putgoing calls after working hours   Mohamed A. Gombolaty
• [asterisk-users] Stopping putgoing calls after working hours   Mohamed A. Gombolaty
• [asterisk-users] Stopping putgoing calls after working hours   Mohamed A. Gombolaty
• [asterisk-users] Compiling Zaptel 1.2.10 on Ubuntu 6.10   Mohamed A. Gombolaty
• [asterisk-users] Hangup or busy when the peer answer outgoing calls   Eloy Gomez
• [asterisk-users] Hangup or busy when the peer answer outgoing calls   Eloy Gomez
• [asterisk-users] Lumenvox speech recognition   Rodrigo Gonzalez
• [asterisk-users] Something is trashing /var/run   Rodrigo Gonzalez
• [asterisk-users] Asterisk and ARI (Aterisk Recording Interface) integration problem   Rodrigo Gonzalez
• [asterisk-users] Asterisk web interface is not parsing the PHP pages   Rodrigo Gonzalez
• [asterisk-users] Moh stuttering   Boyd Goodin
• [asterisk-users] Moh stuttering   Boyd Goodin
• [asterisk-users] Moh stuttering   Boyd Goodin
• [asterisk-users] HTTP Connection Closed on 7960 SIP   Robert Goodyear
• [asterisk-users] HTTP Connection Closed on 7960 SIP   Robert Goodyear
• [asterisk-users] ftp server   Robert Goodyear
• [asterisk-users] ftp server   Robert Goodyear
• [asterisk-users] voicemail issue   Robert Goodyear
• [asterisk-users] detecting busy on queue transfer   Adam Goryachev
• R: [asterisk-users] Siemens Hipath <-> asterisk, pri problem   Giordano Grandis
• R: [asterisk-users] Siemens Hipath <-> asterisk, pri problem   Giordano Grandis
• [asterisk-users] Zaphfc woth florz patch   Giordano Grandis
• R: [asterisk-users] Zaphfc woth florz patch   Giordano Grandis
• [asterisk-users] Oneway audio   Giordano Grandis
• [asterisk-users] Quescom 400   Giordano Grandis
• R: [asterisk-users] Quescom 400   Giordano Grandis
• [asterisk-users] Asterisk RT on Disk On Module Performance and Durability   Michael Graves
• [asterisk-users] Polycom HDVoice   Michael Graves
• [asterisk-users] Embedded Asterisk   Michael Graves
• [asterisk-users] WiFi SIP handset with Bluetooth required   Andy Green
• [asterisk-users] Zap Channel and VM problem   Andy Green
• [asterisk-users] DSL router with integrated SIP proxy?   Ed Greenberg
• [asterisk-users] Audiocodes MP-20x   Ed Greenberg
• [asterisk-users] dialplan issue - 1& 0 should be false   Esteban Guana-Jarrin
• [asterisk-users] dialplan issue - 1& 0 should be evaluated false   Esteban Guana-Jarrin
• [asterisk-users] dialplan issue - 1& 0 should be evaluated false   Esteban Guana-Jarrin
• [asterisk-users] detecting the receivers voicemail   Nitin Gupta
• [asterisk-users] Incorrect Ring tone. Getting a US tone when itshould be AU tone   Craig Guy
• [asterisk-users] Digium vs. Sangoma   TV Guy
• Fwd: [asterisk-users] How can i store PAP2 or any device config in Asterisk   Yair Hakak
• [asterisk-users] asterisk to asterisk DID extentions   Paul Hales
• [asterisk-users] Optus PRI via DSL   Paul Hales
• [asterisk-users] Meeting   Paul Hales
• [asterisk-users] Multiple TE110P cards in one chassis   Paul Hales
• [asterisk-users] VoIP+RJ11 Phone existed?   Paul Hales
• [asterisk-users] Reception Console   Paul Hales
• [asterisk-users] Speed Dials   Paul Hales
• [asterisk-users] adding outbound prefix   Paul Hales
• [asterisk-users] T1 pricing in Oz   Paul Hales
• [asterisk-users] asterisk to asterisk DID extentions   Paul Hales
• [asterisk-users] T1 pricing in Oz   Paul Hales
• [asterisk-users] How to deploy a PBX in such a condition ?   Paul Hales
• [asterisk-users] Incorrect Ring tone. Getting a US tone when it should be AU tone   Paul Hales
• [asterisk-users] Server Recommendations   Paul Hales
• [asterisk-users] Incorrect Ring tone. Getting a US tone when itshould be AU tone   Paul Hales
• [asterisk-users] Architecture for Asterisk   Paul Hales
• [asterisk-users] Re: Architecture for Asterisk   Paul Hales
• [asterisk-users] whisper paging   Hall, Eric M.
• [asterisk-users] Cisco 7970 SIP won't update?   Evan P. Hall
• [asterisk-users] chan_misdn   Mark Hannessen
• [asterisk-users] asterisk misdn incoming line not working.   Mark Hannessen
• [asterisk-users] set outgoing msn on chan_misdn   Mark Hannessen
• [asterisk-users] can't transcode ilbc   James Harper
• [asterisk-users] DISA and legacy PBX   James Harper
• [asterisk-users] different dialtones for DISA   James Harper
• [asterisk-users] Does a "HST Saphir III ML PCI" work with Asterisk?   James Harper
• [asterisk-users] SMS and 1.2.12   James Harper
• [asterisk-users] attended transfer unreliable (2nd try)   Florian Hars
• [asterisk-users] PRI (TE205P) allways RED/NOP   Florian Hars
• [asterisk-users] How big is *your* dialplan??   Darrick Hartman
• [asterisk-users] How big is *your* dialplan??   Darrick Hartman
• [asterisk-users] Is 1.2.12.1 production ready   Darrick Hartman
• [asterisk-users] how to play pre-recorded file in meetme conference   Barry D. Hassler
• [asterisk-users] how to play pre-recorded file in meetme conference   Barry D. Hassler
• [asterisk-users] how to play pre-recorded file in meetme conference   Barry D. Hassler
• [asterisk-users] considering purchasing a t1 card, any recommendations?   Tom Hayden
• [asterisk-users] Asterisk + E1 with MFC/R2 in Argentina?   Angel Heart
• [asterisk-users] ISDN PRI and R2 Trunks in one Dual Port Card   Angel Heart
• [asterisk-users] Unicall Installation   Angel Heart
• [asterisk-users] Unicall Installation   Angel Heart
• [asterisk-users] Unicall Installation   Angel Heart
• [asterisk-users] FW: Why does One-Touch record mute/disconnect call if not dialed quick enough?   Jamie Heckford
• [asterisk-users] Can anyone help? Why does One-Touch record mute/disconnect callif not dialed quick enough?   Jamie Heckford
• [asterisk-users] porting numbers in UK telewest/bt/adept   Jamie Heckford
• [asterisk-users] Example Polycom function key config   Jamie Heckford
• [asterisk-users] two SIP phones as one line   Marc Heckmann
• [asterisk-users] two SIP phones as one line   Marc Heckmann
• [asterisk-users] two SIP phones as one line   Marc Heckmann
• [asterisk-users] two SIP phones as one line   Marc Heckmann
• [asterisk-users] Asterisk and Panasonic KX Model   Gary G. Hendershot
• [asterisk-users] Dial-9 (was Extension Numbering)   Gordon Henderson
• [asterisk-users] Hardware [Was: How big is *your* dialplan??]   Gordon Henderson
• [asterisk-users] Call bridged, but no sound   Henry.L.Coleman
• [asterisk-users] How big is *your* dialplan??   Henry.L.Coleman
• [asterisk-users] Test Call Script   Henry.L.Coleman
• [asterisk-users] two SIP phones as one line   Henry.L.Coleman
• [asterisk-users] two SIP phones as one line   Henry.L.Coleman
• [asterisk-users] Reception Console   Henry.L.Coleman
• [asterisk-users] how to activate recording (automon)   Henry.L.Coleman
• [asterisk-users] say Asterisk to answer   Henry.L.Coleman
• [asterisk-users] Linksys SPA3102 - PSTN hangup detection in the UK   Henry.L.Coleman
• [asterisk-users] Linksys SPA3102 - PSTN hangup detection in the UK   Henry.L.Coleman
• [asterisk-users] (no subject)   Henry.L.Coleman
• [asterisk-users] Linksys SPA3102 - PSTN hangup detection in the UK   Henry.L.Coleman
• [asterisk-users] Add second account to Xlite 3.0   Henry.L.Coleman
• [asterisk-users] "Fixing the Caller-ID Problem", by John Todd for O'ReillyNet   Henry.L.Coleman
• [asterisk-users] SIP v IAX2   Henry.L.Coleman
• [asterisk-users] SIP v IAX2   Henry.L.Coleman
• [asterisk-users] Asterisk 1.2.10 and SCCP   Scott Higginbotham
• [asterisk-users] Polycom Buddy Watch Broken with 2.0.1 Firmware?   Scott Higginbotham
• [asterisk-users] Polycom Buddy Watch Setup help request   Scott Higginbotham
• [asterisk-users] Cisco 7961 - Presence Example?   Scott Higginbotham
• [asterisk-users] Polycom Buddy Watch Setup help request   Scott Higginbotham
• [asterisk-users] Help with gdb bt full results   Scott Higginbotham
• [asterisk-users] Reception Console   Scott Higginbotham
• [asterisk-users] Zultys Phones w/ Encryption   Scott Higginbotham
• [asterisk-users] AW: asterisk-users Digest, Vol 27, Issue 23   Daniel Hikel
• [asterisk-users] snom 360: how to make record button working ?   Joel Hill
• [asterisk-users] snom 360: how to make record button working ?   Joel Hill
• [asterisk-users] Compatability   Joel Hill
• [asterisk-users] T38Passthrough and 1.4 Beta   David Hindmarsh
• [asterisk-users] TNT Max Password reset   Ejay Hire
• [asterisk-users] PoE IP Phone   Ejay Hire
• [asterisk-users] Lots and lots of log files   Ejay Hire
• [asterisk-users] disabling hardware echo can on tdm2400p   Ejay Hire
• [asterisk-users] isdn cross-over ...   Ejay Hire
• [asterisk-users] help this....   Ejay Hire
• [asterisk-users] How big is *your* dialplan??   Ejay Hire
• [asterisk-users] How do you like TrixBox?   Ejay Hire
• [asterisk-users] How do you like TrixBox?   Ejay Hire
• [asterisk-users] How to send correct Caller ID on PRI   Ejay Hire
• [asterisk-users] Inhouse SIP to ZAP has echo sometimes.   Ejay Hire
• [asterisk-users] Windows and file shares   Ejay Hire
• [asterisk-users] "Fixing the Caller-ID Problem", by John Todd for O'ReillyNet   Ejay Hire
• [asterisk-users] Snom or Cisco Phones?   Ejay Hire
• [asterisk-users] Asterisk CDR   Rizwan Hisham
• [asterisk-users] OOH323 GK Context Help   Kengie Ho
• [asterisk-users] RE: Welcome to the "asterisk-users" mailing list   Jessee J Holmes
• [asterisk-users] RE: Welcome to the "asterisk-users" mailing list   Jessee J Holmes
• [asterisk-users] RE: Welcome to the "asterisk-users" mailing list   Jessee J Holmes
• [asterisk-users] RE: Welcome to the "asterisk-users" mailing list   Jessee J Holmes
• [asterisk-users] RE: Welcome to the "asterisk-users" mailing list   Jessee J Holmes
• [asterisk-users] Polycom 2.01 sip issues   Jessee J Holmes
• [asterisk-users] RE: Welcome to the "asterisk-users" mailing list   Jessee J Holmes
• [asterisk-users] Polycom HDVoice   Jessee J Holmes
• [asterisk-users] Polycom HDVoice   Jessee J Holmes
• [asterisk-users] VoipSupply? [Semi-Urgent]   Jessee J Holmes
• [asterisk-users] Polycom IP650   Jessee J Holmes
• [asterisk-users] Polycom boot error   Jessee J Holmes
• [asterisk-users] Audiocodes MP-20x   Jessee J Holmes
• [asterisk-users] Audiocodes MP-114 noise   Jessee J Holmes
• [asterisk-users] new g.729a codecs for asterisk 1.2/1.4 and glibc   Holger Hornung
• [asterisk-users] Fax detection ...   Lee Howard
• [asterisk-users] Fax detection ...   Lee Howard
• [asterisk-users] Fax detection ...   Lee Howard
• [asterisk-users] Fax detection ...   Lee Howard
• [asterisk-users] Bandwidth usage   Lee Howard
• [Asterisk-Users] rxfax problem   Lee Howard
• [Asterisk-Users] rxfax problem   Lee Howard
• [Asterisk-Users] rxfax problem   Lee Howard
• [asterisk-users] Asterisk and ISDN and Hylafax   Lee Howard
• [asterisk-users] Xorcom Astribank   Stepan Hradsky
• [asterisk-users] 480i phone: Is there a trick to registering with *??   Mark Hulber
• [asterisk-users] Strange FXS disconnection problem.   Paul Ianas
• [asterisk-users] user address format   Paul Ianas
• [asterisk-users] MGCP stuff   Paul Ianas
• [asterisk-users] user address format   Paul Ianas
• [asterisk-users] Audiocodes MP-20x   Paul Ianas
• [asterisk-users] UA - number assignment   Paul Ianas
• [asterisk-users] UA - number assignment   Paul Ianas
• [asterisk-users] A Call centre module on Asterisk   Imed Imed
• [asterisk-users] Asterisk Hangups on PRI Interface   Giorgio Incantalupo
• [asterisk-users] Beronet card strange log messages   Giorgio Incantalupo
• [asterisk-users] zt_chanconfig failed   Giorgio Incantalupo
• [asterisk-users] Forcing Marker bit, because SSRC has changed   Giorgio Incantalupo
• [asterisk-users] Unable to open Asterisk database   Giorgio Incantalupo
• [asterisk-users] Unable to open Asterisk database   Giorgio Incantalupo
• [asterisk-users] tdm2400p question   Giorgio Incantalupo
• [asterisk-users] Remote UNIX connection, Remote UNIX disconnected displayed every second   Giorgio Incantalupo
• [asterisk-users] Unable to open Asterisk database   Giorgio Incantalupo
• [asterisk-users] Remote UNIX connection, Remote UNIX disconnected displayed every second   Giorgio Incantalupo
• [asterisk-users] Remote UNIX connection, Remote UNIX disconnected displayed every second   Giorgio Incantalupo
• [asterisk-users] random one way audio and noise between SIP phones on same LAN   Giorgio Incantalupo
• [asterisk-users] random one way audio and noise between SIP phones on same LAN   Giorgio Incantalupo
• [asterisk-users] random one way audio and noise between SIP phoneson same LAN   Giorgio Incantalupo
• [asterisk-users] random one way audio and noise betweenSIP phoneson same LAN   Giorgio Incantalupo
• [asterisk-users] echotraining=yes in misdn.conf is invalid or out of range.   Giorgio Incantalupo
• [asterisk-users] wrong outgoing caller id with PRI lines: maybe usecallingpres involved?   Giorgio Incantalupo
• [asterisk-users] Re: Digium on Dell PowerEdge 1850   Giorgio Incantalupo
• [asterisk-users] wrong outgoing caller id with PRI lines: maybe usecallingpres involved?   Giorgio Incantalupo
• [asterisk-users] /dev/zap/channel ownership   Giorgio Incantalupo
• [asterisk-users] Re: echotraining=yes in misdn.conf is invalid or out of range.   Giorgio Incantalupo
• [asterisk-users] misdn.conf: how to set music on hold   Giorgio Incantalupo
• [asterisk-users] asterisk misdn incoming line not working.   Giorgio Incantalupo
• [asterisk-users] DTMF detection problem in PABX trunk   Giorgio Incantalupo
• [asterisk-users] Asterisk web interface is not parsing the PHP pages   Giorgio Incantalupo
• [asterisk-users] Number Range   Ira
• [asterisk-users] How big is *your* dialplan??   Ira
• [asterisk-users] hold drops audio   Ira
• [asterisk-users] Codes negotiation problems between Asterisk 1.4beta2 and Aastra 480i   Morten Isaksen
• [asterisk-users] Codes negotiation problems between Asterisk1.4beta2 and Aastra 480i   Morten Isaksen
• [asterisk-users] Xorcom TS-1 and Digium TE110P or TE210P   Morten Isaksen
• [asterisk-users] Problem when both Proxy-Authenticate and WWW-Authenticate is required   Morten Isaksen
• [asterisk-users] Gtalk on Asterisk 1.4   Morten Isaksen
• [asterisk-users] Asterisk Advice   K Y Iyer
• [asterisk-users] RE: Asterisk Advice   K Y Iyer
• [asterisk-users] Asterisk Configuration Complete Newbie question   K Y Iyer
• [asterisk-users] Asterisk Configuration Complete Newbie question   K Y Iyer
• [asterisk-users] IP Phones   K Y Iyer
• [asterisk-users] Digium Single Span Card Installation   K Y Iyer
• [asterisk-users] Digium Single Span Card Installation   K Y Iyer
• [asterisk-users] Digium Single Span Card Installation   K Y Iyer
• [asterisk-users] Findme problem   Eric Jacksch
• [asterisk-users] Unable to open Asterisk database   Benjamin Jacob
• [asterisk-users] Stopping putgoing calls after working hours   Benjamin Jacob
• [asterisk-users] nat auto detect ?   Benjamin Jacob
• [asterisk-users] sending sip style messages in response   Benjamin Jacob
• [asterisk-users] sending sip style messages in response   Benjamin Jacob
• [asterisk-users] Why is this happening?   Benjamin Jacob
• [asterisk-users] Stopping putgoing calls after working hours   Benjamin Jacob
• [asterisk-users] nat auto detect ?   Benjamin Jacob
• [asterisk-users] Locking phones at night...   Benjamin Jacob
• [asterisk-users] Locking phones at night...   Benjamin Jacob
• [asterisk-users] Locking phones at night...   Benjamin Jacob
• [asterisk-users] accountcode and amaflags?   Benjamin Jacob
• [asterisk-users] accountcode and amaflags?   Benjamin Jacob
• [asterisk-users] Asterisk Realtime... Help Me!!!   Benjamin Jacob
• [asterisk-users] Asterisk Realtime... Help Me!!!   Benjamin Jacob
• [asterisk-users] Re: checking 'voicemail" externally - doesn't work   Benjamin Jacob
• [asterisk-users] Asterisk Realtime... Help Me!!!   Benjamin Jacob
• [asterisk-users] accountcode and amaflags?   Benjamin Jacob
• [asterisk-users] Voicemail maintenance   Benjamin Jacob
• [asterisk-users] Re: IAX2 goes "one way audio" when lag gets bad   Benjamin Jacob
• [asterisk-users] Asterisk n QoS   Benjamin Jacob
• [asterisk-users] [Fwd: Asterisk n QoS]   Benjamin Jacob
• [asterisk-users] Bandwidth requirements   raphael Jacquot
• [asterisk-users] How to make RTP does not go thru asterisk server   Anuj Jain
• [asterisk-users] How to make RTP does not go thru asterisk server   Anuj Jain
• [asterisk-users] Fax detection ...   James
• [asterisk-users] Fax detection ...   James
• [asterisk-users] TNT Max Password reset   James
• [asterisk-users] TNT Max Password reset   James
• [asterisk-users] odd muting issue   Jason
• [asterisk-users] odd muting issue   Jason
• [asterisk-users] Moh stuttering   Jason
• [asterisk-users] Polycom Buddy Watch Setup help request   Robert Jenkins
• [asterisk-users] Centos kernel 34 vs. 42?   Robert Jenkins
• [asterisk-users] CSU Support on Digium T1 Cards   Christian Jensen
• [asterisk-users] CSU Support on Digium T1 Cards   Christian Jensen
• [asterisk-users] MFC/R2 patch problems   Christian Jensen
• [asterisk-users] IVR   Christian Jensen
• [asterisk-users] cut ip adress from caller id number display (ci$co 7941)   Pavel Jezek
• [asterisk-users] ooh323 dtmf problem   Pavel Jezek
• [asterisk-users] siemens hipath interoperability - PRI/Q.SIG - card recommendation   Pavel Jezek
• [asterisk-users] One way audio half way through call   Pavel Jezek
• [asterisk-users] Callmanager 3.3(5) and Asterisk with ooh323   Pavel Jezek
• [asterisk-users] Re: IAX2 goes "one way audio" when lag gets bad   Pavel Jezek
• [asterisk-users] Callmanager 3.3(5) and Asterisk with ooh323   Pavel Jezek
• [asterisk-users] WiFi Phones (was Looking for Wireless Heaset for Polycom 501)   Pavel Jezek
• [asterisk-users] Callmanager 3.3(5) and Asterisk with ooh323   Pavel Jezek
• [asterisk-users] SIP v IAX2   Pavel Jezek
• [asterisk-users] SIP v IAX2   Pavel Jezek
• [asterisk-users] No ring tone when using IAX   Pavel Jezek
• [asterisk-users] PoE IP Phone   Andrew Joakimsen
• [asterisk-users] MGCP stuff   Andrew Joakimsen
• [asterisk-users] Echo problems on ISDN. (mainly incoming calls)   Andrew Joakimsen
• [asterisk-users] Moh stuttering   Andrew Joakimsen
• [asterisk-users] Psst... Top secret information: Codename Pineapple   Andrew Joakimsen
• [asterisk-users] how can I detect a DTMF tone while on a bridged call ? anyone knows?   Andrew Joakimsen
• [asterisk-users] Re: E164 caller ID   Andrew Joakimsen
• [asterisk-users] VoicePulse Connect 4 Channel Limit?   Andrew Joakimsen
• [asterisk-users] Findme problem   Andrew Joakimsen
• [asterisk-users] wrong outgoing caller id with PRI lines: maybe usecallingpres involved?   Andrew Joakimsen
• [asterisk-users] voicemail usernames can't begin with "j" letter?   Andrew Joakimsen
• [asterisk-users] noise gate for asterisk?   Andrew Joakimsen
• [asterisk-users] Audiocodes MP-20x   Andrew Joakimsen
• [asterisk-users] chan_bluetooth, mobile handset as VoIP terminal?   Andrew Joakimsen
• [asterisk-users] embedded asterisk   Andrew Joakimsen
• [asterisk-users] checking 'voicemail" externally - doesn't work   Andrew Joakimsen
• [asterisk-users] Audiocodes MP-20x   Andrew Joakimsen
• [asterisk-users] checking 'voicemail" externally - doesn't work   Andrew Joakimsen
• [asterisk-users] checking 'voicemail" externally - doesn't work   Andrew Joakimsen
• [asterisk-users] Where to best start looking for voicemail/moh sound quality problem?   Andrew Joakimsen
• [asterisk-users] Polycom provision errors still! Arg!   Andrew Joakimsen
• [Asterisk-Users] rxfax problem   Andrew Joakimsen
• [asterisk-users] Looking for AudioCodes 2 port FXS gateway - Asterisk compatability info   Andrew Joakimsen
• [Asterisk-Users] rxfax problem   Andrew Joakimsen
• [asterisk-users] Zap Channel and VM problem   Andrew Joakimsen
• [asterisk-users] [SOLVED] checking 'voicemail" externally - doesn't work   Andrew Joakimsen
• [Asterisk-Users] rxfax problem   Andrew Joakimsen
• [asterisk-users] Polycom provision errors still! Arg!   Andrew Joakimsen
• [asterisk-users] AstFax Sending a Fax   Andrew Joakimsen
• [asterisk-users] "No Authority Found"   Andrew Joakimsen
• [asterisk-users] Looking for Wireless Heaset for Polycom 501   Andrew Joakimsen
• [asterisk-users] Dynamic Codec Selection   Andrew Joakimsen
• [asterisk-users] Looking for Wireless Heaset for Polycom 501   Andrew Joakimsen
• [asterisk-users] Looking for Wireless Heaset for Polycom 501   Andrew Joakimsen
• [asterisk-users] Simple example for call transfer.   Andrew Joakimsen
• [asterisk-users] Need recommendation for SIP hard phones   Andrew Joakimsen
• [asterisk-users] "No Authority Found"   Andrew Joakimsen
• [asterisk-users] WiFi Phones (was Looking for Wireless Heaset for Polycom 501)   Andrew Joakimsen
• [asterisk-users] [OT] wi-fi ip phone scenario   Andrew Joakimsen
• [asterisk-users] Re: "Fixing the Caller-ID Problem", by John Todd for O'ReillyNet   Andrew Joakimsen
• [asterisk-users] Asterisk Call Statistics   Andrew Joakimsen
• [asterisk-users] DTMF Tones   Andrew Joakimsen
• [asterisk-users] Re: [OT] wi-fi ip phone scenario   Andrew Joakimsen
• [asterisk-users] Server Recommendations   Andrew Joakimsen
• [asterisk-users] No ring tone when using IAX   Andrew Joakimsen
• [asterisk-users] SIP client that runs on Linux or Solaris through X Windows?   Joe
• [asterisk-users] SIP client that runs on Linux or Solaris through X Windows?   Joe
• [asterisk-users] Psst... Top secret information: Codename Pineapple   Olle E Johansson
• [asterisk-users] Psst... Top secret information: Codename Pineapple   Olle E Johansson
• [asterisk-users] Psst... Top secret information: Codename Pineapple   Olle E Johansson
• [asterisk-users] Asterisk-ooh323c Video ?   Olle E Johansson
• [asterisk-users] Psst... Top secret information: Codename Pineapple   Olle E Johansson
• [asterisk-users] Re: SIP stuck channel soft hangup?   Olle E Johansson
• [asterisk-users] sending sip style messages in response   Olle E Johansson
• [asterisk-users] SIP v IAX2   Olle E Johansson
• [asterisk-users] Re: Architecture for Asterisk   Olle E Johansson
• [asterisk-users] Architecture for Asterisk   Olle E Johansson
• [asterisk-users] Polycom HDVoice   Peter Johnson
• [asterisk-users] Polycom IP 501 phone randomly resets itself(loses Received call log, Missed calls, placed calls)   Peter Johnson
• [asterisk-users] Problems with automon   Clif Jones
• [asterisk-users] Problems with automon   Clif Jones
• [asterisk-users] How to make this easier   James Jones
• [asterisk-users] Asterisk Load balancing   James Jones
• [asterisk-users] Polycom 601 & Expansion Module: Light the LEDs???   Jerry Jones
• [asterisk-users] Polycom IP 501 phone randomly resets itself (loses Received call log, Missed calls, placed calls)   Jerry Jones
• [asterisk-users] Polycom HDVoice   Jerry Jones
• [asterisk-users] Occasional one-way audio - Sangoma A101   Jerry Jones
• [asterisk-users] PRI (TE205P) allways RED/NOP   Jerry Jones
• [asterisk-users] RE: ECHO Cancellation in SIP Calls   Jerry Jones
• [asterisk-users] transfer from VM to Cell Phone   Mr. Jones
• [asterisk-users] transfer from VM to Cell Phone   Mr. Jones
• [asterisk-users] 3way calling / codec problem   Mr. Jones
• [asterisk-users] 3way calling / codec problem   Mr. Jones
• [asterisk-users] 3way calling / codec problem   Mr. Jones
• [asterisk-users] Help with fxotune   Jeremy Jongsma
• [asterisk-users] Reception Console   Senad Jordanovic
• [asterisk-users] Digium TDM or SPA-3000?   Joseph
• [asterisk-users] CDR - mysql with asterisk 1.2.12 not working   Joseph
• [asterisk-users] CDR - mysql with asterisk 1.2.12 not working   Joseph
• [asterisk-users] CDR - mysql with asterisk 1.2.12 not working   Joseph
• [asterisk-users] Asterisk 1.2.12 - Can NOT make call out / Asterisk terminate   Joseph
• [asterisk-users] Asterisk 1.2.12 - Can NOT make call out / Asterisk terminate   Joseph
• [asterisk-users] Echo Cancel Cards   Joseph
• [asterisk-users] Asterisk 1.2.12.1 dies after asterisk -rx and never comes up again   Joseph
• [asterisk-users] 1.2.12.1 crashing   Joseph
• [asterisk-users] 1.2.12.1 crashing   Joseph
• [asterisk-users] 1.2.12.1 crashing   Joseph
• [asterisk-users] plainvoip - down ???   Joseph
• [asterisk-users] RIP plainvoip - down ???   Joseph
• [asterisk-users] [SOLVED] 1.2.12.1 crashing   Joseph
• [asterisk-users] [SOLVED] 1.2.12.1 crashing   Joseph
• [asterisk-users] checking 'voicemail" externally - doesn't work   Joseph
• [asterisk-users] Re: checking 'voicemail" externally - doesn't work   Joseph
• [asterisk-users] checking 'voicemail" externally - doesn't work   Joseph
• [asterisk-users] Re: checking 'voicemail" externally - doesn't work   Joseph
• [asterisk-users] Re: checking 'voicemail" externally - doesn't work   Joseph
• [asterisk-users] Re: checking 'voicemail" externally - doesn't work   Joseph
• [asterisk-users] checking 'voicemail" externally - doesn't work   Joseph
• [asterisk-users] [SOLVED] checking 'voicemail" externally - doesn't work   Joseph
• [asterisk-users] Re: can't transcode ilbc   Martin Joseph
• [asterisk-users] Re: can't transcode ilbc   Martin Joseph
• [asterisk-users] Re: WiFi SIP handset with Bluetooth required   Martin Joseph
• [asterisk-users] Re: extensions.conf strangeness   Martin Joseph
• [asterisk-users] Re: Asterisk Server : IDE HDD frequent crash   Martin Joseph
• [asterisk-users] SIP stuck channel soft hangup?   Martin Joseph
• [asterisk-users] Re: Echo Cancel Cards   Martin Joseph
• [asterisk-users] Re: Home Hardware SIP Proxy with use of POTS in Emergency   Martin Joseph
• [asterisk-users] Re: Asterisk 1.2.12 - Can NOT make call out / Asterisk terminate   Martin Joseph
• [asterisk-users] Re: SIP stuck channel soft hangup?   Martin Joseph
• [asterisk-users] Re: SIP stuck channel soft hangup?   Martin Joseph
• [asterisk-users] Re: SIP stuck channel soft hangup?   Martin Joseph
• [asterisk-users] Re: Understanding NAT Traversal   Martin Joseph
• [asterisk-users] Re: SIP stuck channel soft hangup?   Martin Joseph
• [asterisk-users] Re: SIP fails when internet connection lost.   Martin Joseph
• [asterisk-users] Re: Asterisk 'Hosting'   Martin Joseph
• [asterisk-users] Re: VoipSupply? [Semi-Urgent]   Martin Joseph
• [asterisk-users] Re: Codec swap (reinvite)   Martin Joseph
• [asterisk-users] Re: SIP stuck channel soft hangup?   Martin Joseph
• [asterisk-users] Re: 1.4 beta2 on intel mac   Martin Joseph
• [asterisk-users] Re: 1.4 beta2 on intel mac   Martin Joseph
• [asterisk-users] Re: Is 1.2.12.1 production ready   Martin Joseph
• [asterisk-users] Re: Stopping putgoing calls after working hours   Martin Joseph
• [asterisk-users] Re: 1.4 beta2 on intel mac   Martin Joseph
• [asterisk-users] Sipura 901? Any experiences   Martin Joseph
• [asterisk-users] Re: 1.4 on mac OSX 10.4.8   Martin Joseph
• [asterisk-users] Re: IVR problem   Martin Joseph
• [asterisk-users] Re: duplicate "ghost" calls with long duration   Martin Joseph
• [asterisk-users] Re: 1.4 on mac OSX 10.4.8   Martin Joseph
• [asterisk-users] Re: 1.4 on mac OSX 10.4.8   Martin Joseph
• [asterisk-users] Re: 1.4 on mac OSX 10.4.8   Martin Joseph
• [asterisk-users] Re: 1.4 on mac OSX 10.4.8   Martin Joseph
• [asterisk-users] Re: Sip Trunks   Martin Joseph
• [asterisk-users] Re: Sip Trunks   Martin Joseph
• [asterisk-users] 1.4 branch on OSX?   Martin Joseph
• [asterisk-users] Re: 1.4 branch on OSX?   Martin Joseph
• [asterisk-users] Re: 1.4 branch on OSX?   Martin Joseph
• [asterisk-users] Re: 1.4 branch on OSX?   Martin Joseph
• [asterisk-users] Re: 1.4 branch on OSX?   Martin Joseph
• [asterisk-users] Re: G.729 operating on outgoing only   Martin Joseph
• [asterisk-users] Re: checking 'voicemail" externally - doesn't work   Martin Joseph
• [asterisk-users] Disconnect problems and off-hook warning tone   Martin Joseph
• [asterisk-users] Re: 1.4 branch on OSX?   Martin Joseph
• [asterisk-users] Re: rxfax problem   Martin Joseph
• [asterisk-users] Re: IAX2 goes "one way audio" when lag gets bad   Martin Joseph
• [asterisk-users] Re: Dynamic Codec Selection   Martin Joseph
• [asterisk-users] Re: IAX2 goes "one way audio" when lag gets bad   Martin Joseph
• [asterisk-users] Re: Dynamic Codec Selection   Martin Joseph
• [asterisk-users] Re: Choice of soundfile format   Martin Joseph
• [asterisk-users] Re: Looking for Wireless Heaset for Polycom 501   Martin Joseph
• [asterisk-users] Re: Choice of soundfile format   Martin Joseph
• [asterisk-users] Re: ECHO Cancellation in SIP Calls   Martin Joseph
• [asterisk-users] Re: SIP v IAX2   Martin Joseph
• [asterisk-users] Re: ECHO Cancellation in SIP Calls   Martin Joseph
• [asterisk-users] Re: Voicemail and OSX 10.4 Intel   Martin Joseph
• [asterisk-users] Re: [OT] wi-fi ip phone scenario   Martin Joseph
• [asterisk-users] Re: [OT] wi-fi ip phone scenario   Martin Joseph
• [asterisk-users] Re: [OT] wi-fi ip phone scenario   Martin Joseph
• [asterisk-users] Re: [OT] wi-fi ip phone scenario   Martin Joseph
• [asterisk-users] Re: [OT] wi-fi ip phone scenario   Martin Joseph
• [asterisk-users] Re: Opinions on the best wholesale origination/term providers   Martin Joseph
• [asterisk-users] No Dialtone   Eddie Johnson Jr
• [asterisk-users] No Dialtone   Eddie Johnson Jr
• [asterisk-users] No Dialtone   Eddie Johnson Jr
• [asterisk-users] No Dialtone   Eddie Johnson Jr
• [asterisk-users] No Dialtone   Eddie Johnson Jr
• [asterisk-users] Becoming a User on IRC   Eddie Johnson Jr
• [asterisk-users] Becoming a User on IRC   Eddie Johnson Jr
• [asterisk-users] Becoming a User on IRC   Eddie Johnson Jr
• [asterisk-users] Understanding NAT Traversal   Samy Kamkar
• [asterisk-users] call takeover?   Samy Kamkar
• [asterisk-users] Asterisk on Embedded platforms   Prasad Kandikonda
• [asterisk-users] Asterisk on Embedded platforms   Prasad Kandikonda
• [asterisk-users] Test Call Script   John Kane
• [asterisk-users] VoicePulse Connect 4 Channel Limit?   Nate Kapi
• [asterisk-users] How big is *your* dialplan??   Alejandro Kauffmann
• [asterisk-users] Multiple TE110P cards in one chassis   Alejandro Kauffmann
• [asterisk-users] asterisk-1.2.13 fails to 'Make' in Fedore Core 6'   Keithy
• [asterisk-users] Digium TDM or SPA-3000?   Shawn Kelley
• [asterisk-users] Increase VoiceMail Messages Recording Gain - AudioCalls are Ok   Shawn Kelley
• [asterisk-users] Out dialing Integration   Shawn Kelley
• [asterisk-users] modprobe Ztdummy is not working   Jean-Etienne Kelly
• [asterisk-users] considering purchasing a t1 card, any recommendations?   Greg Kennedy
• [asterisk-users] disabling hardware echo can on tdm2400p   Sean Kennedy
• [asterisk-users] TNT Max Password reset   Steve Kennedy
• [asterisk-users] porting numbers in UK telewest/bt/adept   Steve Kennedy
• [asterisk-users] Digium TDM or SPA-3000?   Thomas Kenyon
• [asterisk-users] G729 Licence Consumption Problem   Thomas Kenyon
• [asterisk-users] Echo Cancel Cards   Thomas Kenyon
• [asterisk-users] single conference, multiple numbers   Thomas Kenyon
• [asterisk-users] SIP fails when internet connection lost.   Thomas Kenyon
• [asterisk-users] 3way calling / codec problem   Thomas Kenyon
• [asterisk-users] 3way calling / codec problem   Thomas Kenyon
• [asterisk-users] Re: Is 1.2.12.1 production ready (Mauro Zanin)   Thomas Kenyon
• [asterisk-users] Errors in console in every call made when using 1.4b3   Thomas Kenyon
• [asterisk-users] [SOLVED] 1.2.12.1 crashing   Thomas Kenyon
• [asterisk-users] Why does it take at least 4 flipping days before asterisk tries to resolve a provider?   Thomas Kenyon
• [asterisk-users] meet me   Khaled
• [asterisk-users] Need Help in Meetme (Conferencing)   Ehsan Khosrowshahi
• [asterisk-users] Unknown signalling method 'pri_cpe'   Eugeniy Khvastunov
• [asterisk-users] Unknown signalling method 'pri_cpe'   Eugeniy Khvastunov
• [asterisk-users] Rejecting call   Eugeniy Khvastunov
• [asterisk-users] Primary D-Channel on span 2 down   Eugeniy Khvastunov
• [asterisk-users] Primary D-Channel & channal numbers....   Eugeniy Khvastunov
• [asterisk-users] Call from internal num. to VoIP gate   Eugeniy Khvastunov
• [asterisk-users] Dialplan Syslog   Kristian Kielhofner
• [asterisk-users] OT: Polycom time sync - sorta   Kristian Kielhofner
• [asterisk-users] Asterisk RT on Disk On Module Performance and Durability   Kristian Kielhofner
• [asterisk-users] Asterisk RT on Disk On Module Performance and Durability   Kristian Kielhofner
• [asterisk-users] Asterisk RT on Disk On Module Performance and Durability   Kristian Kielhofner
• [asterisk-users] NFAS Not Passing Audio on B-chan 48,72,96   Kristian Kielhofner
• [asterisk-users] NFAS Not Passing Audio on B-chan 48,72,96   Kristian Kielhofner
• [asterisk-users] NFAS Not Passing Audio on B-chan 48,72,96   Kristian Kielhofner
• [asterisk-users] macros reference?   Kristian Kielhofner
• [asterisk-users] Netgear WGT Flash-fest at Astricon   Kristian Kielhofner
• [asterisk-users] Embedded Asterisk   Kristian Kielhofner
• [asterisk-users] Embedded Asterisk   Kristian Kielhofner
• [asterisk-users] Embedded Asterisk   Kristian Kielhofner
• [asterisk-users] Embedded Asterisk   Kristian Kielhofner
• [asterisk-users] Embedded Asterisk   Kristian Kielhofner
• [asterisk-users] Embedded Asterisk   Kristian Kielhofner
• [asterisk-users] Embedded Asterisk   Kristian Kielhofner
• [asterisk-users] embedded asterisk   Kristian Kielhofner
• [asterisk-users] Meetme... No channel type registered for 'zap'   Kristian Kielhofner
• [asterisk-users] Polycom Call Parking   Kevin Kiely
• [asterisk-users] Polycom Call Parking   Kevin Kiely
• [asterisk-users] Broadvoice incoming DTMF problems   Kevin Kiely
• [asterisk-users] Audiocodes MP-114 noise   Jason Kim
• [asterisk-users] Asterisk web interface is not parsing the PHPpages   Jordan Kirby
• [asterisk-users] How do you like TrixBox?   Klaverstyn, David C
• [asterisk-users] Call Asterisk : It calls me backup with a dial tone   Klaverstyn, David C
• [asterisk-users] Phone Rings, Immediate Hangup and then Rings Again.   Klaverstyn, David C
• [asterisk-users] Phone Rings, Immediate Hangup and then Rings Again.   Klaverstyn, David C
• [asterisk-users] Incorrect Ring tone. Getting a US tone when it should be AU tone   Klaverstyn, David C
• [asterisk-users] Incorrect Ring tone. Getting a US tone when it should be AU tone   Klaverstyn, David C
• [asterisk-users] Incorrect Ring tone. Getting a US tone when it should be AU tone   Klaverstyn, David C
• [asterisk-users] wrong password on authentication for INVITE   Klaverstyn, David C
• [asterisk-users] wrong password on authentication for INVITE   Klaverstyn, David C
• [asterisk-users] regexten & regcontext broken for SIP?   Philipp von Klitzing
• [asterisk-users] [Asterisk-Java] SipShowPeerAction   Zack Kneisley
• [asterisk-users] WiFi SIP handset with Bluetooth required   Andrew Kohlsmith
• [asterisk-users] How big is *your* dialplan??   Andrew Kohlsmith
• [asterisk-users] How big is *your* dialplan??   Andrew Kohlsmith
• [asterisk-users] How big is *your* dialplan??   Andrew Kohlsmith
• [asterisk-users] How big is *your* dialplan??   Andrew Kohlsmith
• [asterisk-users] How big is *your* dialplan??   Andrew Kohlsmith
• [asterisk-users] How big is *your* dialplan??   Andrew Kohlsmith
• [asterisk-users] How big is *your* dialplan??   Andrew Kohlsmith
• [asterisk-users] How big is *your* dialplan??   Andrew Kohlsmith
• [asterisk-users] Why is this happening?   Andrew Kohlsmith
• [asterisk-users] Why is this happening?   Andrew Kohlsmith
• [asterisk-users] Why is this happening?   Andrew Kohlsmith
• [asterisk-users] Why is this happening?   Andrew Kohlsmith
• [asterisk-users] PRI boards with g729 capable DSPs   Andrew Kohlsmith
• [asterisk-users] Nerdvittle's Reminders and Zaptel   Andrew Kohlsmith
• [asterisk-users] SIP v IAX2   Andrew Kohlsmith
• [asterisk-users] Transfer app and DTMF via SIP info   Michael Konietzny
• [asterisk-users] Kind of OT : Europeans going to Astricon   Stelios Koroneos
• [asterisk-users] Embedded Asterisk   Stelios Koroneos
• [asterisk-users] Avaya 8300 - Asterisk integration using H.323   Andrey Kovalenko
• [asterisk-users] help   Derek Kruger
• [asterisk-users] Audiocodes MP-20x   Arun Kumar
• [asterisk-users] call file mechanism   K Kuo
• [asterisk-users] Re: Voicemail Press '0'   LJ
• [asterisk-users] Looking for AudioCodes 2 port FXS gateway - Asterisk compatability info   LJ
• [asterisk-users] Re: Re: Voicemail Press '0'   LJ
• [asterisk-users] Re: Re: Re: Voicemail Press '0'   LJ
• [asterisk-users] Re: Audiocodes MP-20x   LJ
• [asterisk-users] New Asterisk-GUI?   LJ
• [asterisk-users] Re: Snom, mute and rtptimeout   LJ
• [asterisk-users] Cisco 7960 Skinny calling SIP phone   Anthony LaMantia
• [asterisk-users] RE: Getting Asterisk to work with GoogleTalk   Robert LaPoint
• [asterisk-users] Gtalk on Asterisk 1.4   Robert LaPoint
• [asterisk-users] Gtalk on Asterisk 1.4   Robert LaPoint
• [asterisk-users] route by caller id   Pezhman Lali
• [asterisk-users] new route by caller id   Pezhman Lali
• [asterisk-users] new route by caller id   Pezhman Lali
• [asterisk-users] anti ex-girlfriend   Pezhman Lali
• [asterisk-users] Polycom SP4000 ftp problem   Edwin Lam
• [asterisk-users] Polycom SP4000 ftp problem   Edwin Lam
• [asterisk-users] Polycom SP4000 ftp problem   Edwin Lam
• [asterisk-users] Polycom SP4000 ftp problem   Edwin Lam
• [asterisk-users] tdm2400p question   Lito Lampitoc
• [asterisk-users] tdm2400p question   Lito Lampitoc
• [asterisk-users] Forcing Marker bit, because SSRC has changed   Steve Langstaff
• [asterisk-users] SIP_HEADER function; what names are available?   Steve Langstaff
• [asterisk-users] SIP_HEADER function; what names are available?   Steve Langstaff
• [asterisk-users] SipAddHeader   Steve Langstaff
• [asterisk-users] G.729 operating on outgoing only   Joel Lansden
• [asterisk-users] Re: G.729 operating on outgoing only   Joel Lansden
• [asterisk-users] prohibit CallerID presentation   Kristian Larsson
• [asterisk-users] asterisk guru needed for job in Chicago area   Andrew Latham
• [asterisk-users] Asterisk on virtual machine   Andrew Latham
• [asterisk-users] Server Recommendations   Andrew Latham
• [asterisk-users] lines usage statistics   Christophorus Laube
• [asterisk-users] detecting busy on queue transfer   Lenz
• [asterisk-users] detecting busy on queue transfer   Lenz
• [asterisk-users] Problem with 2 machines connected with IAX   Lenz
• [asterisk-users] Tutorial - avoiding queue_log file rotation   Lenz
• [asterisk-users] Tutorial - avoiding queue_log file rotation   Lenz
• [asterisk-users] pop a web page with DID in url   Lenz
• [asterisk-users] QueueMetrics 1.3 released today   Lenz
• [asterisk-users] noise gate for asterisk?   Lenz
• [asterisk-users] attempting native bridge on TDM2400   Lenz
• [asterisk-users] Maximum talktime in a queue?   Lenz
• [asterisk-users] Maximum talktime in a queue?   Lenz
• [asterisk-users] "Cheapest" way to determine channels in a group from outside asterisk?   Lenz
• [asterisk-users] OT: (Job) Full-Time Asterisk Opportunity   Ronald Lewis
• [asterisk-users] Do you encounter this REC alarm before?   Xue Liangliang
• [asterisk-users] Why the MusicOnHold sound so soft?   Xue Liangliang
• [asterisk-users] something about Agent Transfer   Xue Liangliang
• [asterisk-users] All calls Hangup after receive these logs.   Xue Liangliang
• [asterisk-users] DID is not working (call is not routing)   R.R Libera
• [asterisk-users] Digium TDM or SPA-3000?   R.R. Libera
• [asterisk-users] Digium Interfaces in Tampa?   R.R. Libera
• [asterisk-users] Intel Chipset 945p compatible?   R.R. Libera
• [asterisk-users] Need USA DID + trunk provider   R.R. Libera
• [asterisk-users] Need USA DID + trunk provider   R.R. Libera
• [asterisk-users] USA Origination recommended service?   R.R. Libera
• [asterisk-users] Echo Cancel Cards   R.R. Libera
• [asterisk-users] Asterisk + E1 with MFC/R2 in Argentina?   R.R. Libera
• [asterisk-users] considering purchasing a t1 card, any recommendations?   R.R. Libera
• [asterisk-users] considering purchasing a t1 card, any recommendations?   R.R. Libera
• [asterisk-users] VoicePulse Connect 4 Channel Limit?   R.R. Libera
• [asterisk-users] VoicePulse Connect 4 Channel Limit?   R.R. Libera
• [asterisk-users] hardware requirements..   R.R. Libera
• [asterisk-users] Billing Solution ?   R.R. Libera
• [asterisk-users] Monitor Current outgoing calls   Ralph Liebessohn
• [asterisk-users] Is 1.2.12.1 production ready   Ralph Liebessohn
• [asterisk-users] how to config chanspy   Ralph Liebessohn
• [asterisk-users] Asterisk@Home problems   Sharon Lim
• [asterisk-users] Linksys PAP2: calling tone stops after 5 tones   Jose Limeres
• [asterisk-users] VOIP with PSTN backup   Peter Lindquist
• [asterisk-users] 12 port FXx PCI card   Peter Lindquist
• [asterisk-users] Reception Console   Peter Lindquist
• [asterisk-users] TDM2400P wiring.   Mailing List
• [asterisk-users] plainvoip - down ???   Mailing List
• [asterisk-users] PRI Card   Duracom Lists
• [asterisk-users] Echo Cancel Cards   Rob Lith
• [asterisk-users] External domain   Josemar Lohn
• [asterisk-users] tools/techniques/metrics for measurement of end-point quality   Nilesh Londhe
• [asterisk-users] Call Quality / Echo / Problems   Alexander Lopez
• [asterisk-users] Polycom Buddy Watch Broken with 2.0.1 Firmware?   Alexander Lopez
• [asterisk-users] (no subject)   Alexander Lopez
• [asterisk-users] Generate Random Numbers in dialplan   Alexander Lopez
• [asterisk-users] Generate Random Numbers in dialplan   Alexander Lopez
• [asterisk-users] Re: Generate Random Numbers in dialplan   Alexander Lopez
• [asterisk-users] How big is *your* ego?   Matt Loretitsch
• [asterisk-users] Clicking Noise on Pure Voip Calls   carl Lougher
• [asterisk-users] sequential Dial() commands   Luki
• [asterisk-users] Is 1.2.12.1 production ready   Luki
• [asterisk-users] Screen pop based on incoming DID   Wolfgang Lumpp
• [asterisk-users] Recording from a script   Nikolai Lusan
• [asterisk-users] Wouldn't Tri-tone detection in Dial() be cool?   Richard Lyman
• [asterisk-users] Asterisk and dialer Running on Thin Clients   Richard Lyman
• [asterisk-users] Where are the kernel sources?   Jim Lynch
• [asterisk-users] Can't get second line of Sangoma A200 to work.   Jim Lynch
• [asterisk-users] Dial and connect to sip provider works, but no audio.   Jim Lynch
• [asterisk-users] sip provider not working   Jim Lynch
• [asterisk-users] No zap* commands?   Jim Lynch
• [asterisk-users] Something is trashing /var/run   Jim Lynch
• [asterisk-users] No zap* commands?   Jim Lynch
• [asterisk-users] Something is trashing /var/run   Jim Lynch
• [asterisk-users] unix sysctl config for asterisk   Mike Lynchfield
• [asterisk-users] Test Call Script   Mike Lynchfield
• [asterisk-users] unauthenticated calls   Mike Lynchfield
• [asterisk-users] Some file aren't loaded its No file in that Directory.   Mike Lynchfield
• [asterisk-users] Call Asterisk : It calls me backup with a dial tone   Mike Lynchfield
• [asterisk-users] Re: unix sysctl config for asterisk   Mike Lynchfield
• [asterisk-users] How big is *your* ego?   Mike Lynchfield
• [asterisk-users] Is 1.2.12.1 production ready   Mike Lynchfield
• [asterisk-users] ChanIsAvail() in 1.2.12.1   Julian Lyndon-Smith
• [asterisk-users] How to enable talking in chanspy while spying?   Julian Lyndon-Smith
• [asterisk-users] Unique call ID's across several systems   Julian Lyndon-Smith
• [asterisk-users] Unique call ID's across several systems   Julian Lyndon-Smith
• [asterisk-users] Unique call ID's across several systems   Julian Lyndon-Smith
• [asterisk-users] Enhancements for the Queue application   Julian Lyndon-Smith
• [asterisk-users] detecting ring   Julian Lyndon-Smith
• [asterisk-users] Need Help in Meetme (Conferencing)   Julian Lyndon-Smith
• [asterisk-users] SPA 3102   Tom Lynn
• [asterisk-users] VoipSupply? [Semi-Urgent] (Big Apology)   Tom Lynn
• [asterisk-users] Avaya 8300 - Asterisk integration using H.323   Tom Lynn
• [asterisk-users] Looking for a Voicemail Lamp device   Tom Lynn
• [asterisk-users] Looking for a Voicemail Lamp device   Tom Lynn
• [asterisk-users] Looking for a Voicemail Lamp device   Tom Lynn
• [asterisk-users] Re: How do you like TrixBox?   Tom Lynn
• [asterisk-users] two SIP phones as one line   Tom Lynn
• [asterisk-users] Asterisk on Embedded platforms   Tom Lynn
• [asterisk-users] Advice on GUI   Tom Lynn
• [asterisk-users] attended transfer unreliable (2nd try)   Doug Lytle
• [asterisk-users] AEL2 Catching on?   Doug Lytle
• [asterisk-users] PRI issues   Doug Lytle
• [asterisk-users] Tellabs and a PRI   Doug Lytle
• [asterisk-users] Tellabs and a PRI   Doug Lytle
• [asterisk-users] Odd echo issue with speaker phone   Doug Lytle
• [asterisk-users] Odd echo issue with speaker phone   Doug Lytle
• [asterisk-users] Re: PRI issues   Doug Lytle
• [asterisk-users] Re: PRI issues   Doug Lytle
• [asterisk-users] Range Operator   Doug Lytle
• [asterisk-users] Number Range   Doug Lytle
• [asterisk-users] GotoIfTime - much slowdown with 90 conditions?   Doug Lytle
• [asterisk-users] Re: PRI issues   Doug Lytle
• [asterisk-users] Voicemail Press '0'   Doug Lytle
• [asterisk-users] Voicemail Press '0'   Doug Lytle
• [asterisk-users] How big is *your* dialplan??   Doug Lytle
• [asterisk-users] How big is *your* dialplan??   Doug Lytle
• [asterisk-users] How big is *your* dialplan??   Doug Lytle
• [asterisk-users] Re: PRI issues   Doug Lytle
• [asterisk-users] E164 caller ID   Doug Lytle
• [asterisk-users] Polycom IP 501 message light   Doug Lytle
• [asterisk-users] Voicemail prompts clipped when retrieving from some SIP phones   Doug Lytle
• [asterisk-users] Switchtype,Signalling,rxwink warnings   Doug Lytle
• [asterisk-users] Tellabs and a PRI   Doug Lytle
• [asterisk-users] Tellabs and PRI   Doug Lytle
• [asterisk-users] Attended transfer hanging PRI channel   Doug Lytle
• [asterisk-users] Attended transfer hanging PRI channel   Doug Lytle
• [asterisk-users] wrong outgoing caller id with PRI lines: maybe usecallingpres involved?   Doug Lytle
• [asterisk-users] Adit 600 resetting   Doug Lytle
• [asterisk-users] All calls Hangup after receive these logs.   Doug Lytle
• [asterisk-users] need help using tftp for polycom 501   Doug Lytle
• [asterisk-users] Nerdvittle's Reminders and Zaptel   Doug Lytle
• [asterisk-users] polycom's don't register with 2.6.18   Doug Lytle
• [asterisk-users] polycom's don't register with 2.6.18   Doug Lytle
• [asterisk-users] Digium TE110P   Doug Lytle
• [asterisk-users] Out bound calls 'you must first dial a 1'   Doug Lytle
• [asterisk-users] How to do Automatic Daylight Saving on Grandstream GXP-2000   Doug Lytle
• [asterisk-users] How to do Automatic Daylight Saving on Grandstream GXP-2000   Doug Lytle
• [asterisk-users] Re: How to do Automatic Daylight Saving on Grandstream GXP-2000   Doug Lytle
• [asterisk-users] dealing with blind transfers to invalid extensions   Doug Lytle
• [asterisk-users] Asterisk Call Statistics   Doug Lytle
• [asterisk-users] MOR and MCC - billing solutions for Asterisk released   Guillermo Salas M.
• [asterisk-users] IAX softphones   Guillermo Salas M.
• [asterisk-users] SIP v IAX2   Guillermo Salas M.
• [asterisk-users] Billing Solution ?   Guillermo Salas M.
• [asterisk-users] Asterisk and ARI (Aterisk Recording Interface) integration problem   Guillermo Salas M.
• [asterisk-users] Codec swap (reinvite)   Julian J. M.
• [asterisk-users] blind transfers with IP Polycom 501   Julian J. M.
• [asterisk-users] Asterisk+Panasonic KX-TDA100+zaphfc NT link problem   MC
• [asterisk-users] we are having trouble detecting the # for making a transfer from an E1, usually under some load, please help   MF
• [asterisk-users] Is there a way to collect dtmf digits during a call? (inband)   MF
• [asterisk-users] how can I detect a DTMF tone while on a bridged call ? anyone knows?   MF
• [asterisk-users] how can I detect a DTMF tone while on a bridged call ? anyone knows?   MF
• [asterisk-users] using asterisk to do remote control functions   Gregory Machin
• [asterisk-users] using asterisk to do remote control functions   Gregory Machin
• [asterisk-users] using asterisk to do remote control functions   Gregory Machin
• [asterisk-users] !! No channel map, no channel, and no ds1? What am I supposed to identify?   Frederico Madeira
• [asterisk-users] Direct call vs Block call   Frederico Madeira
• [asterisk-users] DTMF detection problem in PABX trunk   Frederico Madeira
• [asterisk-users] DTMF detection problem in PABX trunk   Frederico Madeira
• [asterisk-users] Re: channel variable   Leif Madsen
• [asterisk-users] sending sip style messages in response   Mikael Magnusson
• [asterisk-users] Multiple dial macros at the same time   Graham Mainwaring
• [asterisk-users] AGI PHP   Thiago Maluf
• [Asterisk-Users] Help, please help -- IAX2 softphone to server on LAN   Naija Man
• [asterisk-users] no callerid from PSTN using TDM2400P   Naija Man
• [asterisk-users] no callerid from PSTN using TDM2400P   Naija Man
• [asterisk-users] Re: asterisk-users Digest, Vol 27, Issue 49   Naija Man
• [asterisk-users] How big is *your* dialplan??   Naija Man
• [asterisk-users] Re: Generate Random Numbers in dialplan   Naija Man
• [asterisk-users] Asterisk <-> Live Communications Server Integration   William Mandra
• [asterisk-users] Asterisk Manager   Maps
• [asterisk-users] Re: Asterisk Manager   Maps
• [asterisk-users] Re: Asterisk Manager   Maps
• [asterisk-users] Asterisk Manager   MapsAir
• [asterisk-users] Asterisk conferencing features   Rafael Marangoni
• [asterisk-users] Asterisk conferencing features   Rafael Marangoni
• [asterisk-users] Digium TE405 card and Matra PBX   Jan Marek
• [asterisk-users] zaptel 1.2.10 make problem   Jan Marek
• [asterisk-users] INVAL Messages   Adrian Marsh
• [asterisk-users] INVAL Messages   Adrian Marsh
• [asterisk-users] INVAL Messages   Adrian Marsh
• [asterisk-users] Polycom 2.01 sip issues   John Marvin
• [asterisk-users] Re: Choice of soundfile format   John Marvin
• [asterisk-users] "Fixing the Caller-ID Problem", by John Todd for O'ReillyNet   John Marvin
• [asterisk-users] How do I reset a password?   George Masgras
• [asterisk-users] Monitor Current outgoing calls   George Masgras
• [asterisk-users] Monitor Current outgoing calls   George Masgras
• [asterisk-users] asterisk to asterisk DID extentions   Matt
• [asterisk-users] Problem with 2 machines connected with IAX   Matt
• [asterisk-users] Problem with 2 machines connected with IAX   Matt
• [asterisk-users] Problem with 2 machines connected with IAX   Matt
• [asterisk-users] Problem with 2 machines connected with IAX   Matt
• [asterisk-users] Home Hardware SIP Proxy with use of POTS in Emergency   Matt
• [asterisk-users] asterisk to asterisk DID extentions   Matt
• [asterisk-users] 1.2.12.1 crashing   Matt
• [asterisk-users] 1.2.12.1 crashing   Matt
• [asterisk-users] 1.2.12.1 crashing   Matt
• [asterisk-users] 1.2.12.1 crashing   Matt
• [asterisk-users] How do I figure out where this connection is coming from?   Matt
• [asterisk-users] 1.2.12.1 crashing   Matt
• [asterisk-users] VoipSupply? [Semi-Urgent]   Matt
• [asterisk-users] How do I figure out where this connection is coming from?   Matt
• [asterisk-users] VoipSupply? [Semi-Urgent]   Matt
• [asterisk-users] Why is this happening?   Matt
• [asterisk-users] Why is this happening?   Matt
• [asterisk-users] Why is this happening?   Matt
• [asterisk-users] Why is this happening?   Matt
• [asterisk-users] Why is this happening?   Matt
• [asterisk-users] Re: Is 1.2.12.1 production ready (Mauro Zanin)   Matt
• [asterisk-users] One way audio half way through call   Matt
• [asterisk-users] Question on one-way-audio with IAX   Matt
• [asterisk-users] Re: Question on one-way-audio with IAX   Matt
• [asterisk-users] One way audio half way through call   Matt
• [asterisk-users] One way audio half way through call   Matt
• [asterisk-users] One way audio half way through call   Matt
• [asterisk-users] IAX2 goes "one way audio" when lag gets bad   Matt
• [asterisk-users] One way audio half way through call   Matt
• [asterisk-users] Re: IAX2 goes "one way audio" when lag gets bad   Matt
• [asterisk-users] Re: IAX2 goes "one way audio" when lag gets bad   Matt
• [asterisk-users] Re: IAX2 goes "one way audio" when lag gets bad   Matt
• [asterisk-users] Install issues on Freebsd 6.1 with Asterisk 1.4.0-beta2   Artifex Maximus
• [asterisk-users] problem with setting outbound caller id when calling another asterisk   Chris Mazuc
• [asterisk-users] problem with setting outbound caller id when calling another asterisk   Chris Mazuc
• [asterisk-users] Nerdvittle's Reminders and Zaptel   John McCollough
• [asterisk-users] Nerdvittle's Reminders and Zaptel   John McCollough
• [asterisk-users] Echo problems on ISDN. (mainly incoming calls)   John McEntee
• [asterisk-users] Reception Console   Callum McGillivray
• [asterisk-users] T1 pricing in Oz   Callum McGillivray
• [Asterisk-Users] rxfax problem   Jim McIver
• [asterisk-users] Asterisk RT on Disk On Module Performance and Durability   Jeremy McNamara
• [asterisk-users] Asterisk RT on Disk On Module Performance and Durability   Jeremy McNamara
• [asterisk-users] How big is *your* dialplan??   Jeremy McNamara
• [asterisk-users] plainvoip - down ???   Jeremy McNamara
• [asterisk-users] Unique call ID's across several systems   Jeremy McNamara
• [asterisk-users] Asterisk + Huawei   Carlos Medina
• [asterisk-users] Re: Asterisk + Huawei   Carlos Medina
• [asterisk-users] Re: Asterisk + Huawei   Carlos Medina
• [asterisk-users] Asterisk 1.4beta3 and Asterisk Manager API Action: ExtensionState   Jorge Mendoza
• [asterisk-users] 1.4 and slow sound playback   Bill Merriam
• [asterisk-users] duplicate "ghost" calls with long duration   Bjoern Metzdorf
• [asterisk-users] duplicate "ghost" calls with long duration   Bjoern Metzdorf
• [asterisk-users] SIP users with Database   Giovanni Miano
• [asterisk-users] say Asterisk to answer   Giovanni Miano
• [asterisk-users] accountcode and amaflags?   Giovanni Miano
• [asterisk-users] Which is the best ?   Michel
• [asterisk-users] Auto Dial problem!   Michel
• [asterisk-users] Asterisk dial out (in SIP) to another asterisk context !   Michel
• [asterisk-users] Query on Call Parking   Chan Kwang Mien
• [asterisk-users] Query regarding Pulse Dialing at 20 pps   Chan Kwang Mien
• [asterisk-users] Make/Break ratio for Pulse Dialing   Chan Kwang Mien
• [asterisk-users] Make/Break ratio for Pulse Dialing   Chan Kwang Mien
• [asterisk-users] Transfer feature - howto?   Mike
• [asterisk-users] Transfer feature - howto?   Mike
• [asterisk-users] Cisco 7970 Unbootable After FW Upgrade   Jeremiah Millay
• [asterisk-users] asterisk to asterisk DID extentions   Avi Miller
• [asterisk-users] asterisk-addons-1.2.4 Installation Problem   Avi Miller
• [asterisk-users] ftp server   Avi Miller
• [asterisk-users] ftp server   Avi Miller
• [asterisk-users] Optus PRI via DSL   Avi Miller
• [asterisk-users] ftp server   Avi Miller
• [asterisk-users] polycom reboot script   Avi Miller
• [asterisk-users] ftp server   Avi Miller
• [asterisk-users] Reception Console   Avi Miller
• [asterisk-users] checking 'voicemail" externally - doesn't work   Avi Miller
• [asterisk-users] Re: checking 'voicemail" externally - doesn't work   Avi Miller
• [asterisk-users] Re: checking 'voicemail" externally - doesn't work   Avi Miller
• [asterisk-users] Re: checking 'voicemail" externally - doesn't work   Avi Miller
• [asterisk-users] Asterisk and ISDN and Hylafax   Avi Miller
• [asterisk-users] Why is this happening?   Mitch Miller
• [asterisk-users] Getting started with sample dial plans   Mitch Miller
• [asterisk-users] /dev/zap/channel ownership   Mitch Miller
• [asterisk-users] /dev/zap/channel ownership   Mitch Miller
• [asterisk-users] Polycom Call Parking   Noah Miller
• [asterisk-users] Polycom Call Parking   Noah Miller
• [asterisk-users] Which IP Phone is good to use at reception desk?   Noah Miller
• [asterisk-users] digium compatibility notes   Noah Miller
• [asterisk-users] voicemail maintenance questions   Noah Miller
• [asterisk-users] Polycom Call Parking   Noah Miller
• [asterisk-users] Video Conference   Noah Miller
• [asterisk-users] Extremely choppy sound on some of ourPOTSnetwork calls; goes away with mute   Noah Miller
• [asterisk-users] Video Conference   Noah Miller
• [asterisk-users] OT: Polycom time sync - sorta   Noah Miller
• [asterisk-users] How to forward DID to another Server   Noah Miller
• [asterisk-users] No Dialtone   Noah Miller
• [asterisk-users] Voicemail and Forwarding   Noah Miller
• [asterisk-users] No Dialtone   Noah Miller
• [asterisk-users] Extremely choppy sound on some of ourPOTSnetwork calls; goes away with mute   Noah Miller
• [asterisk-users] ftp server   Noah Miller
• [asterisk-users] polycom reboot script   Noah Miller
• [asterisk-users] ftp server   Noah Miller
• [asterisk-users] How do you like TrixBox?   Noah Miller
• [asterisk-users] Polycom IP 501 phone randomly resets itself (loses Received call log, Missed calls, placed calls)   Noah Miller
• [asterisk-users] two SIP phones as one line   Noah Miller
• [asterisk-users] what hardware and is it possible   Noah Miller
• [asterisk-users] Is 1.2.12.1 production ready   Noah Miller
• [asterisk-users] Is 1.2.12.1 production ready   Noah Miller
• [asterisk-users] PRI Card   Noah Miller
• [asterisk-users] Clicking Noise on Pure Voip Calls   Noah Miller
• [asterisk-users] Polycom SP4000 ftp problem   Noah Miller
• [asterisk-users] Voicemail help   Noah Miller
• [asterisk-users] IAX2 goes "one way audio" when lag gets bad   Noah Miller
• [asterisk-users] Without ZapTel inferface or Card install , is Conference working or Not   Noah Miller
• [asterisk-users] Choice of soundfile format   Noah Miller
• [asterisk-users] Example Polycom function key config   Noah Miller
• [asterisk-users] Home Hardware SIP Proxy with use of POTS in Emergency   John Millican
• [asterisk-users] SPA 3102   John Millican
• [asterisk-users] Asterisk Postgres Native support   John Miloo
• [asterisk-users] Asterisk access Postgres for Realtime Configuration   John Miloo
• [asterisk-users] Connection question...   Danko Miocevic
• [asterisk-users] Connection question...   Danko Miocevic
• [asterisk-users] chan_bluetooth - one way audio   Danko Miocevic
• [asterisk-users] Dial without phone   Mir
• [asterisk-users] bristuff problem?   Louis-David Mitterrand
• [asterisk-users] cisco 7960 not registering after * restart   Louis-David Mitterrand
• [asterisk-users] cisco 7960 not registering after * restart   Louis-David Mitterrand
• [asterisk-users] polycom's don't register with 2.6.18   Louis-David Mitterrand
• [asterisk-users] polycom's don't register with 2.6.18   Louis-David Mitterrand
• [asterisk-users] Re: polycom's don't register with 2.6.18   Louis-David Mitterrand
• [asterisk-users] IAX2/SIP Wifi Phones   Ansar Mohammed
• [asterisk-users] Configuring 2 Asterisk servers with a SIP trunk   Alok Mohapatra
• [asterisk-users] Asterisk web interface is not parsing the PHP pages   Alok Mohapatra
• [asterisk-users] Which IP Phone is good to use at reception desk?   Mojo with Horan & Company, LLC
• [asterisk-users] Where is the PlayDTMF command?   Mojo with Horan & Company, LLC
• [asterisk-users] Newbie question about meetme   Mojo with Horan & Company, LLC
• [asterisk-users] How to make RTP does not go thru asterisk server   Mojo with Horan & Company, LLC
• [asterisk-users] How to make RTP does not go thru asterisk server   Mojo with Horan & Company, LLC
• [asterisk-users] CDR problem with call transfer   Mojo with Horan & Company, LLC
• [asterisk-users] Polycom Call Parking   Mojo with Horan & Company, LLC
• [asterisk-users] No voice for when using Playback and background   Mojo with Horan & Company, LLC
• [asterisk-users] IAX or SIP termination provider that reaches6421xxxxxxx?   Mojo with Horan & Company, LLC
• [asterisk-users] IAX or SIP termination provider that reaches6421xxxxxxx?   Mojo with Horan & Company, LLC
• [asterisk-users] snom 360: how to make record button working ?   Mojo with Horan & Company, LLC
• [asterisk-users] snom 360: how to make record button working ?   Mojo with Horan & Company, LLC
• [asterisk-users] snom 360: how to make record button working ?   Mojo with Horan & Company, LLC
• [asterisk-users] Range Operator   Mojo with Horan & Company, LLC
• [asterisk-users] Understanding NAT Traversal   Mojo with Horan & Company, LLC
• [asterisk-users] Hangup or busy when the peer answer outgoing calls   Mojo with Horan & Company, LLC
• [asterisk-users] Understanding NAT Traversal   Mojo with Horan & Company, LLC
• [asterisk-users] Test Call Script   Mojo with Horan & Company, LLC
• [asterisk-users] Polycom IP 501 phone randomly resets itself (loses Received call log, Missed calls, placed calls)   Mojo with Horan & Company, LLC
• [asterisk-users] asterisk upgrade   Mojo with Horan & Company, LLC
• [asterisk-users] Stopping putgoing calls after working hours   Mojo with Horan & Company, LLC
• [asterisk-users] Why is this happening?   Mojo with Horan & Company, LLC
• [asterisk-users] Locking phones at night...   Mojo with Horan & Company, LLC
• [asterisk-users] gotoiftime and Macro question   Mojo with Horan & Company, LLC
• [asterisk-users] Electric usage of a tdm400p   Mojo with Horan & Company, LLC
• [asterisk-users] Locking phones at night...   Mojo with Horan & Company, LLC
• [asterisk-users] Electric usage of a tdm400p   Mojo with Horan & Company, LLC
• [asterisk-users] Why is this happening?   Melcon Moraes
• [asterisk-users] VoiceOne 0.4.0 released: a new web-based and open source GUI   Melcon Moraes
• [asterisk-users] dialplan issue - 1& 0 should be evaluated false   Melcon Moraes
• [asterisk-users] dialplan issue - 1& 0 should be evaluated false   Melcon Moraes
• [asterisk-users] TDM22B   Jack Morgan
• [asterisk-users] IVR problem   Jack Morgan
• [asterisk-users] IVR problem   Jack Morgan
• [asterisk-users] Newbie h/w Q, and confirming basic concepts   Mike Morris
• [asterisk-users] Re: Passing Arguments to FastAGI   Tony Mountifield
• [asterisk-users] Re: AGI() in 1.2 and 1.4   Tony Mountifield
• [asterisk-users] Re: Real-time and priority "n"   Tony Mountifield
• [asterisk-users] Digium TE410P LED problem   Tony Mountifield
• [asterisk-users] Re: Digium TE410P LED problem   Tony Mountifield
• [asterisk-users] Re: Digium TE410P LED problem   Tony Mountifield
• [asterisk-users] How to busy out PRI channels?   Tony Mountifield
• [asterisk-users] Re: Meetme... No channel type registered for 'zap'   Tony Mountifield
• [asterisk-users] Re: Meetme... No channel type registered for 'zap'   Tony Mountifield
• [asterisk-users] Re: Meetme... No channel type registered for 'zap'   Tony Mountifield
• [asterisk-users] Re: Meetme... No channel type registered for 'zap'   Tony Mountifield
• [asterisk-users] Re: Meetme... No channel type registered for 'zap'   Tony Mountifield
• [asterisk-users] extensions.conf strangeness   Marco Mouta
• [asterisk-users] Fax detection ...   Marco Mouta
• [asterisk-users] detecting busy on queue transfer   Marco Mouta
• [asterisk-users] detecting busy on queue transfer   Marco Mouta
• [asterisk-users] Siemens Hipath <-> asterisk, pri problem   Marco Mouta
• [asterisk-users] Siemens Hipath <-> asterisk, pri problem   Marco Mouta
• [asterisk-users] Issues with calling certain phone numbers...   Marco Mouta
• [asterisk-users] Issues with calling certain phone numbers...   Marco Mouta
• [asterisk-users] Configuration / dialplan problem   Marco Mouta
• [asterisk-users] Codes negotiation problems between Asterisk1.4beta2 and Aastra 480i   Marco Mouta
• [asterisk-users] password for vm users   Marco Mouta
• [asterisk-users] Increase VoiceMail Messages Recording Gain - Audio Calls are Ok   Marco Mouta
• [asterisk-users] Increase VoiceMail Messages Recording Gain - AudioCalls are Ok   Marco Mouta
• [asterisk-users] Hicom 150 -- BRI -- Asterisk   Marco Mouta
• [asterisk-users] Increase VoiceMail Messages Recording Gain -AudioCalls are Ok   Marco Mouta
• [asterisk-users] cdr_addon_mysql.c - Asterisk 1.4 - Asterisk Addons   Marco Mouta
• [asterisk-users] Increase VoiceMail Messages Recording Gain - AudioCalls are Ok   Marco Mouta
• [asterisk-users] Increase VoiceMail Messages Recording Gain - AudioCalls are Ok   Marco Mouta
• [asterisk-users] Locking phones at night...   Marco Mouta
• [asterisk-users] Orange Flash Light Mitel 5215 - Asterisk - working !   Marco Mouta
• [asterisk-users] Locking phones at night...   Marco Mouta
• [asterisk-users] #Transfer - Timeout is configurable?   Marco Mouta
• [asterisk-users] CDR_DISPOSITION_FAILED - Call has been answered correctly   Marco Mouta
• [asterisk-users] UA - number assignment   Marco Mouta
• [asterisk-users] Distributing calls among channels in dial group   Marco Mouta
• [asterisk-users] Iax bug ?   Marco Mouta
• [asterisk-users] Iax bug ?   Marco Mouta
• [asterisk-users] Iax bug ?   Marco Mouta
• [asterisk-users] Direct call vs Block call   Marco Mouta
• [asterisk-users] asterisk misdn incoming line not working.   Marco Mouta
• [asterisk-users] asterisk misdn incoming line not working.   Marco Mouta
• [asterisk-users] Iax bug ?   Marco Mouta
• [asterisk-users] Diva server 4bri and Portuguese BRI   Marco Mouta
• [asterisk-users] SIP Server   Marco Mouta
• [asterisk-users] Diva server 4bri and Portuguese BRI   Marco Mouta
• [asterisk-users] sip realtime broken?   Marco Mouta
• [asterisk-users] Configuration / dialplan problem   Mark Muffett
• [asterisk-users] Configuration / dialplan problem   Mark Muffett
• [asterisk-users] ZAP chanel doesn't reset if external caller hangs up in menu   Devraj Mukherjee
• [asterisk-users] IP Phones   Devraj Mukherjee
• [asterisk-users] Wouldn't Tri-tone detection in Dial() be cool?   Steve Murphy
• [asterisk-users] Re: Calling Functions from AEL2   Steve Murphy
• [asterisk-users] Re: AEL2 #include madness in Asterisk 1.4 - Murf?   Steve Murphy
• [asterisk-users] How big is *your* dialplan??   Steve Murphy
• [asterisk-users] How big is *your* dialplan??   Steve Murphy
• [asterisk-users] Re: AEL Question   Steve Murphy
• [asterisk-users] Re: AEL question   Steve Murphy
• [asterisk-users] Re: Generate Random Numbers in dialplan   Steve Murphy
• [asterisk-users] Re: Generate Random Numbers in dialplan   Steve Murphy
• [asterisk-users] Re: Asterisk hangs up on incoming analog calls after a while   Steve Murphy
• [asterisk-users] Re: AEL2 and the variables   Steve Murphy
• [asterisk-users] Re: anti ex-girlfriend   Steve Murphy
• [asterisk-users] Asterisk users help   Naidu, Vijay
• [asterisk-users] Asterisk 1.2.12.1 dies after asterisk -rx and never comes up again   Dinesh Nair
• [asterisk-users] Anybody using "inphonex" service?   Rajeev Natarajan
• [asterisk-users] A Call centre module on Asterisk   Rajeev Natarajan
• [asterisk-users] Newbie question about meetme   Rajeev Natarajan
• [asterisk-users] How to get the agent id in the recording filename   Rajeev Natarajan
• [asterisk-users] Configuring 2 Asterisk servers with a SIP trunk   Rajeev Natarajan
• [asterisk-users] Extension and Voice Mail setup   Ahmed Ndaula
• [asterisk-users] Polycom boot error   Neider, Clint
• [asterisk-users] polycomphone   clive.chan(Alpha Trilogies Networks)
• [asterisk-users] polycomphone (SOLVED)   clive.chan(Alpha Trilogies Networks)
• [asterisk-users] Problems with automon   Michael Neuhauser
• [asterisk-users] Problems with automon   Michael Neuhauser
• [asterisk-users] Re: extensions.conf strangeness   Michael Neuhauser
• [asterisk-users] voicemail usernames can't begin with "j" letter?   Michael Neuhauser
• [asterisk-users] echotraining=yes in misdn.conf is invalid or out of range.   Jarkko Nevala
• [Asterisk-Users] rxfax problem   M. Shokuie Nia
• [asterisk-users] Default login information for a ArtDio IPF-2600   M. Shokuie Nia
• [asterisk-users] Quintum DX as gateway to PSTN for Asterisk   M. Shokuie Nia
• [asterisk-users] VoiceOne 0.4.0 released: a new web-based and opensource GUI   M. Shokuie Nia
• [asterisk-users] Intrado V9-1-1   Marnus van Niekerk
• [asterisk-users] 2x* and realtime   Marnus van Niekerk
• [asterisk-users] A Call centre module on Asterisk   Marnus van Niekerk
• [asterisk-users] sipXezphone   Marnus van Niekerk
• [asterisk-users] Asterisk Server : IDE HDD frequent crash   Sam Norris
• [asterisk-users] Call Center requirements   John Novack
• [asterisk-users] "set verbose 4" in SVN trunk?   John Novack
• [asterisk-users] Newbie h/w Q, and confirming basic concepts   John Novack
• [asterisk-users] Echo Cancel Cards   John Novack
• [asterisk-users] NFAS Not Passing Audio on B-chan 48,72,96   John Novack
• [asterisk-users] NFAS Not Passing Audio on B-chan 48,72,96   John Novack
• [asterisk-users] considering purchasing a t1 card, any recommendations?   John Novack
• [asterisk-users] Is 1.2.12.1 production ready   John Novack
• [asterisk-users] considering purchasing a t1 card, any recommendations?   John Novack
• [asterisk-users] Make/Break ratio for Pulse Dialing   John Novack
• [asterisk-users] How to do Automatic Daylight Saving on Grandstream GXP-2000   John Novack
• [asterisk-users] FXO Cards vs. Channel bank with T1   John Novack
• [asterisk-users] (no subject)   Jordan Novak
• [asterisk-users] voicemail maintenance questions   Jordan Novak
• [asterisk-users] Voicemail maintenance   Jordan Novak
• [asterisk-users] IVR menu system external database information collection   Jordan Novak
• [asterisk-users] sip agent stuck in queue even after restarts   Jordan Novak
• [asterisk-users] call center status viewer   Jordan Novak
• [asterisk-users] simple dialplan trick I can't figure out (smdi, mwi substitute)   Jordan Novak
• [asterisk-users] Audiocodes MP-20x   Andrew Nowrot
• [asterisk-users] Multiple TE110P cards in one chassis   Massimo Nuvoli
• SV: [asterisk-users] How do you like TrixBox?   Amund Nygaard
• [asterisk-users] [Asterisk-Java] SipShowPeerAction   Richard OSS
• [asterisk-users] TNT Max Password reset   Natambu Obleton
• [asterisk-users] TNT Max Password reset   Natambu Obleton
• [asterisk-users] TNT Max Password reset   Natambu Obleton
• [asterisk-users] RE:Asterisk and dialer Running on Thin Clients   Vitaly Oborsky
• [asterisk-users] RE:Asterisk and dialer Running on Thin Clients   Vitaly Oborsky
• [asterisk-users] Realtime trouble with contex   Nikita Olenets
• [asterisk-users] Cisco 7970 Unbootable After FW Upgrade   Greg Oliver
• [asterisk-users] Cisco CCM - Asterisk   Greg Oliver
• [asterisk-users] Cisco 7970 SIP won't update?   Greg Oliver
• [asterisk-users] Polycom HDVoice   Greg Oliver
• [asterisk-users] tools/techniques/metrics for measurement of end-point quality   Olivier
• [asterisk-users] Fax detection ...   Olivier
• [Asterisk-Users] How to create echo for learning purpose   Olivier
• [Asterisk-Users] How to create echo for learning purpose   Olivier
• [asterisk-users] Psst... Top secret information: Codename Pineapple   Olivier
• [asterisk-users] Call drop and strange CDR records   Olivier
• [asterisk-users] Asterisk 1.2.12.1 and snom 360 6.2.3 no audio   Olivier
• [Asterisk-Users] How to get Linksys-Sipura error codes ?   Olivier
• [asterisk-users] Junghanns quadBRI and mISDN   Olivier
• [Asterisk-Users] chan_capi and bristuff   Olivier
• [Asterisk-Users] Would you support a Bristuff mailing list ?   Olivier
• [asterisk-users] Asterisk and ISDN and Hylafax   Olivier
• [Asterisk-Users] Would you support a Bristuff mailing list ?   Olivier
• [asterisk-users] VoIP GSM Gateways   Olivier
• [asterisk-users] CALEA support within asterisk?   J. Oquendo
• [asterisk-users] CALEA support within asterisk?   J. Oquendo
• [asterisk-users] CALEA support within asterisk?   J. Oquendo
• [asterisk-users] Building a terrorist-friendly telephone network (Was: CALEA support)   J. Oquendo
• [asterisk-users] Re: Bandwidth requirements   J. Oquendo
• [asterisk-users] PoE IP Phone   J. Oquendo
• [asterisk-users] Lots and lots of log files   J. Oquendo
• [asterisk-users] Lots and lots of log files   J. Oquendo
• [asterisk-users] Lots and lots of log files   J. Oquendo
• [asterisk-users] How big is *your* ego?   J. Oquendo
• [asterisk-users] How big is *your* ego?   J. Oquendo
• [asterisk-users] duplicate "ghost" calls with long duration   J. Oquendo
• [asterisk-users] duplicate "ghost" calls with long duration   J. Oquendo
• [asterisk-users] plainvoip - down ???   J. Oquendo
• [asterisk-users] "Fixing the Caller-ID Problem", by John Todd for O'ReillyNet   J. Oquendo
• [asterisk-users] ASterisk Start problem   J. Oquendo
• [asterisk-users] Asteroid SIP Denial of Service Tool   J. Oquendo
• [asterisk-users] Inaccurate CDRs   Philipp Ott
• [asterisk-users] Codes negotiation problems between Asterisk1.4beta2 and Aastra 480i   Gareth Owen
• [asterisk-users] Codes negotiation problems betweenAsterisk1.4beta2 and Aastra 480i   Gareth Owen
• [asterisk-users] Codes negotiation problemsbetweenAsterisk1.4beta2 and Aastra 480i   Gareth Owen
• [asterisk-users] Destar 0.2.0 released   Alejandro Ríos P.
• [asterisk-users] fully featured and robust * client gui?   Andres Paglayan
• [asterisk-users] operator console   Andres Paglayan
• [asterisk-users] How big is *your* dialplan??   George Pajari
• [asterisk-users] tdm2400p question   George Pajari
• [asterisk-users] Conversations Mix   Panitaxx
• [asterisk-users] Multiple 'routes' to extension in different contextes. How to influence search oder?   Benoit Panizzon
• [asterisk-users] Asterisk Advice   Tim Panton
• [asterisk-users] SIP client that runs on Linux or Solaris through X Windows?   Tim Panton
• [asterisk-users] [Asterisk-Java] SipShowPeerAction   Tim Panton
• [asterisk-users] SIP client that runs on Linux or Solaris through X Windows?   Tim Panton
• [asterisk-users] pop a web page with DID in url   Tim Panton
• [asterisk-users] SIP vz IAX...   Tim Panton
• [asterisk-users] Asterisk RT on Disk On Module Performance andDurability   Tim Panton
• [asterisk-users] connecting multiple servers with iax - authentication fails   Tim Panton
• [asterisk-users] 1.4 beta2 on intel mac   Tim Panton
• [asterisk-users] 1.4 beta2 on intel mac   Tim Panton
• [asterisk-users] how to play pre-recorded file in meetme conference   Tim Panton
• [asterisk-users] Re: Understanding NAT Traversal   Tim Panton
• [asterisk-users] How big is *your* dialplan??   Tim Panton
• [asterisk-users] How big is *your* dialplan??   Tim Panton
• [asterisk-users] Re: 1.4 beta2 on intel mac   Tim Panton
• [asterisk-users] Re: 1.4 beta2 on intel mac   Tim Panton
• [asterisk-users] Why is this happening?   Tim Panton
• [asterisk-users] Unique ID   Tim Panton
• [asterisk-users] install MAGI   Tim Panton
• [asterisk-users] Embedded Asterisk   Tim Panton
• [asterisk-users] Embedded Asterisk   Tim Panton
• [asterisk-users] 1.4 branch on OSX?   Tim Panton
• [asterisk-users] Re: 1.4 branch on OSX?   Tim Panton
• [asterisk-users] 1.4 branch on OSX?   Tim Panton
• [asterisk-users] Re: 1.4 branch on OSX?   Tim Panton
• [asterisk-users] Asterisk and dialer Running on Thin Clients   Tim Panton
• [asterisk-users] make menuselect question- Module Embedding   Tim Panton
• [asterisk-users] porting numbers in UK telewest/bt/adept   Tim Panton
• [asterisk-users] SIP v IAX2   Tim Panton
• [asterisk-users] Re: SIP v IAX2   Tim Panton
• [asterisk-users] channel.c: Avoided initial deadlock   Tim Panton
• [asterisk-users] "Cheapest" way to determine channels in a group from outside asterisk?   Tim Panton
• [asterisk-users] IPv6   Tim Panton
• [asterisk-users] Voicemail and OSX 10.4 Intel   David Parcerisa
• [asterisk-users] Re: Voicemail and OSX 10.4 Intel   David Parcerisa
• [asterisk-users] G729 Licence Consumption Problem   Alvaro Parres
• [asterisk-users] G729 Licence Consumption Problem   Alvaro Parres
• [asterisk-users] [OT] Nokia E60/61/70 and SIP   Enrico Pasqualotto
• [asterisk-users] ZapHFC & quadBRI D-Channel going down randomly   Alberto Pastore
• [asterisk-users] Re: ZapHFC & quadBRI D-Channel going down randomly   Alberto Pastore
• [asterisk-users] Re: ZapHFC & quadBRI D-Channel going down randomly   Alberto Pastore
• [asterisk-users] identifying Eicon Diva Server V-4BRI-8M vs 4BRI-8M   Alberto Pastore
• [asterisk-users] vISDN, mISDN, bristuff [was: Re: Bristuff qozap drivers problem]   Alberto Pastore
• [asterisk-users] Problems with chan-capi and Eicon Diva 4BRI   Alberto Pastore
• [asterisk-users] Junghanns quadBRI and mISDN   Alberto Pastore
• [asterisk-users] Junghanns quadBRI and mISDN   Alberto Pastore
• [asterisk-users] Asterisk and ISDN and Hylafax   Alberto Pastore
• [asterisk-users] ISDN-BRI issue   Alberto Pastore
• [asterisk-users] Asterisk and ISDN and Hylafax   Alberto Pastore
• [asterisk-users] [OT] wi-fi ip phone scenario   Alberto Pastore
• [asterisk-users] ISDN-BRI issue   Alberto Pastore
• [asterisk-users] [OT] wi-fi ip phone scenario   Alberto Pastore
• [asterisk-users] Diva server 4bri and Portuguese BRI   Alberto Pastore
• [asterisk-users] Re: [OT] wi-fi ip phone scenario   Alberto Pastore
• [asterisk-users] Diva server 4bri and Portuguese BRI   Alberto Pastore
• [asterisk-users] [OT] wi-fi ip phone scenario   Alberto Pastore
• [asterisk-users] [OT] wi-fi ip phone scenario   Alberto Pastore
• [asterisk-users] Asterisk and Siemens C450IP   Alberto Pastore
• [asterisk-users] Cisco 7960 Skinny calling SIP phone   Alberto Pastore
• [asterisk-users] Problems with Tormenta 2 quad card   Tomasz Paszkowski
• [asterisk-users] CALEA support within asterisk?   Patrick
• [asterisk-users] Student Research - Asterisk H323 Video   Patrick
• [asterisk-users] Asterisk-ooh323c Video ?   Patrick
• [asterisk-users] Asterisk-ooh323c Video ?   Patrick
• [asterisk-users] Compiling H323 channel Asterisk 1.4.Beta3?   Patrick
• [asterisk-users] chan_h323.so Asterisk Beta compilation   Patrick
• [asterisk-users] chan_h323.so Asterisk Beta compilation   Patrick
• [asterisk-users] 1.4 Beta 3 H323 Video?   Patrick
• [asterisk-users] Default login information for a ArtDio IPF-2600   George Patterson
• [asterisk-users] Out bound calls 'you must first dial a 1'   George Patterson
• [asterisk-users] Using the ZOOM 5801 ATA with Asterisk   Paul
• [asterisk-users] Asterisk Realtime with ODBC/MySql   Maurizio Pederneschi
• [asterisk-users] SIP users with Database   Maurizio Pederneschi
• [asterisk-users] SIP users with Database   Maurizio Pederneschi
• [asterisk-users] Asterisk Realtime... Help Me!!!   Maurizio Pederneschi
• [asterisk-users] Asterisk Realtime... Help Me!!!   Maurizio Pederneschi
• [asterisk-users] Asterisk Realtime... Help Me!!!   Maurizio Pederneschi
• [asterisk-users] Asterisk Realtime... Help Me!!!   Maurizio Pederneschi
• [asterisk-users] defining trunks in sip.conf   Joao Pereira
• [asterisk-users] Snom or Cisco Phones?   Joao Pereira
• [asterisk-users] Re: SIP v IAX2   Roberto Pereyra
• [asterisk-users] Asterisk RT on Disk On Module Performance and Durability   Erick Perez
• [asterisk-users] Asterisk RT on Disk On Module Performance and Durability   Erick Perez
• [asterisk-users] Asterisk RT on Disk On Module Performance and Durability   Erick Perez
• [asterisk-users] Asterisk RT on Disk On Module Performance andDurability   Erick Perez
• [asterisk-users] OT: BioFuel to power phone networks   Erick Perez
• [asterisk-users] Issues with Asterisk 1.4 Beta   Erick Perez
• [asterisk-users] 1.2.12.1 crashing   Erick Perez
• [asterisk-users] Electric usage of a tdm400p   Erick Perez
• [asterisk-users] Electric usage of a tdm400p   Erick Perez
• [asterisk-users] res_sqlite problems   Erick Perez
• [asterisk-users] Is SQLite a replacement for Mysql while using ARA in 1.2.x   Erick Perez
• [asterisk-users] Is SQLite a replacement for Mysql while using ARA in 1.2.x   Erick Perez
• [asterisk-users] 0 channels configured with tdm400 (tdm04b rev. G)   Erick Perez
• [asterisk-users] Re: 0 channels configured with tdm400 (tdm04b rev. G)   Erick Perez
• [asterisk-users] Re: 0 channels configured with tdm400 (tdm04b rev. G)   Erick Perez
• [asterisk-users] Grandstream ATA 286 tdm400 and Asterisk 1.2-13   Erick Perez
• [asterisk-users] compilation problem with asterisk-addons   Erick Perez
• [asterisk-users] Grandstream ATA 286 tdm400 and Asterisk 1.2-13   Erick Perez
• [asterisk-users] Problems with Dial In - Dial Out via SIP - no voice   Christian Peter
• [asterisk-users] Problems with Dial In - Dial Out via SIP - no voice   Christian Peter
• [asterisk-users] PRI TON/pridialplan digit prefixing   Jesse Peterson
• [asterisk-users] Mitel 5224/SIP no MWI   Jesse Peterson
• [asterisk-users] Re: Mitel 5224/SIP no MWI [RESOLVED]   Jesse Peterson
• [asterisk-users] Billing Solution ?   Noc Phibee
• [asterisk-users] Billing Solution ?   Noc Phibee
• [asterisk-users] Fxo box for asterisk ?   Noc Phibee
• [asterisk-users] IVR for the called part (IVR inside out)   Mark Phillips
• [asterisk-users] voicemail maintenance questions   Mark Phillips
• [asterisk-users] FS: Sangoma A200 10 port FXO card   Mark Phillips
• [Asterisk-Users] problem configuring a digium quad E1 card   Paul Philpott
• [asterisk-users] (no subject)   Scott Pinhorne
• [asterisk-users] RT Problem: Asterisk & Session Border Controller   Scott Pinhorne
• [asterisk-users] Digium vs. Sangoma   Unmetered Pipe
• [asterisk-users] Asterisk@Home problems   William Piper
• [asterisk-users] DID is not working (call is not routing)   William Piper
• [asterisk-users] DID is not working (call is not routing)   William Piper
• [asterisk-users] Monitor Current outgoing calls   William Piper
• [asterisk-users] Monitor Current outgoing calls   William Piper
• [asterisk-users] Urgent Billing   William Piper
• [asterisk-users] question about CDR command   William Piper
• [asterisk-users] question about CDR command   William Piper
• [asterisk-users] Real-time LDAP config   Richi Plana
• [asterisk-users] Simple example for call transfer.   Jonson Player
• [asterisk-users] T1 Passthrough   Don Pobanz
• [asterisk-users] "Guest" SIP-Invites not accepted   Sascha Pollok
• [asterisk-users] Asterisk manager   Voipers Portugal
• [asterisk-users] Asterisk manager   Voipers Portugal
• [asterisk-users] Asterisk manager   Voipers Portugal
• [asterisk-users] Information on Asterisk 1.4-beta 3 and ARA   Raffaele Porzio
• [asterisk-users] Requirements for Asterisk & SER integratio   Vamsi Pottangi
• [asterisk-users] eagi-sphinx-test how and why   Vamsi Pottangi
• [asterisk-users] asterisk queues with SER, aka "sip show peers"   Mark Price
• [asterisk-users] authenticating forwarded calls   Mark Price
• [asterisk-users] sequential Dial() commands   Mark Price
• [asterisk-users] sequential Dial() commands   Mark Price
• [asterisk-users] external username conflict in dialplan   Mark Price
• [asterisk-users] autocreate peer + sippeers table entry => auth required?   Mark Price
• [asterisk-users] snom 360: how to make record button working ?   Joe Pukepail
• [asterisk-users] Switchtype,Signalling,rxwink warnings   Remi Quezada
• [asterisk-users] Switchtype,Signalling,rxwink warnings   Remi Quezada
• [asterisk-users] ACK when using SIP Proxy   H Quintana
• [asterisk-users] unauthenticated calls   Mark Quitoriano
• [asterisk-users] unauthenticated calls   Mark Quitoriano
• [asterisk-users] unauthenticated calls   Mark Quitoriano
• [asterisk-users] update_header: Unable to find our position   Mark Quitoriano
• [asterisk-users] G726 prompts   RR
• [asterisk-users] G726 prompts   RR
• [asterisk-users] answering machine detection   Raj
• [asterisk-users] New Asterisk StumbleUpon Group   Chris Ramsey
• [asterisk-users] Free copy of "TrixBox Made Easy"   Chris Ramsey
• [asterisk-users] How big is *your* dialplan??   Chris Ramsey
• [asterisk-users] transfer from VM to Cell Phone   Chris Ramsey
• [asterisk-users] How do you like TrixBox?   Chris Ramsey
• [asterisk-users] How do you like TrixBox?   Chris Ramsey
• [asterisk-users] New and Improved   Chris Ramsey
• [asterisk-users] How do you like TrixBox?   Chris Ramsey
• [asterisk-users] Help with Dialplan Rules Please!   Chris Ramsey
• [asterisk-users] Help with Dialplan Rules Please!   Chris Ramsey
• [asterisk-users] Call Routing Time Issue   Chris Ramsey
• [asterisk-users] Polycom 601 & Expansion Module: Light the LEDs???   Faris Raouf
• [asterisk-users] Is 1.2.12.1 production ready   Faris Raouf
• [asterisk-users] Linksys SPA3102 - PSTN hangup detection in the UK   Faris Raouf
• [asterisk-users] Linksys SPA3102 - PSTN hangup detection in the UK   Faris Raouf
• [asterisk-users] Linksys SPA3102 - PSTN hangup detection in the UK   Faris Raouf
• [asterisk-users] Linksys SPA3102 - PSTN hangup detection in the UK   Faris Raouf
• [asterisk-users] Linksys SPA3102 - PSTN hangup detection in the UK   Faris Raouf
• [asterisk-users] Linksys SPA3102 - PSTN hangup detection in the UK   Faris Raouf
• [asterisk-users] Linksys SPA3102 - PSTN hangup detection in the UK   Faris Raouf
• [asterisk-users] Problem with CallerID (UK) TDM400P ( CID timed out waiting for ring )   Faris Raouf
• [asterisk-users] install MAGI   Alvaro A Colunga Rdz
• [asterisk-users] Re: Digium on Dell PowerEdge 1850   Bruce Reeves
• [asterisk-users] voicemail usernames can't begin with "j" letter?   Bruce Reeves
• [asterisk-users] [OT] wi-fi ip phone scenario   Bruce Reeves
• [asterisk-users] Setting the H323 Callerid sent by asterisk (using chan_h323)   Nathan Reeves
• [asterisk-users] PoE IP Phone   Josh Reineke
• [asterisk-users] alive check for HA constellation   Sebastian Reitenbach
• [asterisk-users] alive check for HA constellation   Sebastian Reitenbach
• [asterisk-users] Dial out trhough a FXS channel on a TDM card   Robson Ribeiro
• [asterisk-users] Looking for Wireless Heaset for Polycom 501   Jim Rice
• [asterisk-users] 1.4 beta voicemail warning   Hadley Rich
• [asterisk-users] Locking phones at night...   Hadley Rich
• [asterisk-users] Unicall Installation   Hadley Rich
• [asterisk-users] considering purchasing a t1 card, any recommendations?   Richard
• [asterisk-users] cisco 2600   Gary Richardson
• [asterisk-users] AccountCode set in sip.conf but not showing up in CDR   JR Richardson
• [asterisk-users] RE: AccountCode set in sip.conf but not showing up in CDR   JR Richardson
• [asterisk-users] Re: asterisk not detecting hangup   JR Richardson
• [asterisk-users] How big is *your* dialplan??   Marc Rieber
• [asterisk-users] sip trunk - SIP/2.0 488 Not Acceptable Media   Andrea Riela
• [asterisk-users] VoiceOne 0.4.0 released: a new web-based and open source GUI   Jonathan Rivera
• [asterisk-users] polycomphone   Alex Robar
• [asterisk-users] New tutorial - peering two * servers using IAX   Alex Robar
• [asterisk-users] Asterisk@Home problems   Alex Robar
• [asterisk-users] swap CID with DID   Alex Robar
• [asterisk-users] Asterisk@Home problems   Alex Robar
• [asterisk-users] Asterisk@Home problems   Alex Robar
• [asterisk-users] TDM400P incoming route for DID   Alex Robar
• [asterisk-users] TDM400P incoming route for DID   Alex Robar
• [asterisk-users] unauthenticated calls   Alex Robar
• [asterisk-users] How do you like TrixBox?   Alex Robar
• [asterisk-users] Help with Dialplan Rules Please!   Alex Robar
• [asterisk-users] Stopping putgoing calls after working hours   Alex Robar
• [asterisk-users] Digium vs. Sangoma   Alex Robar
• [asterisk-users] Asterisk web interface is not parsing the PHP pages   Alex Robar
• [asterisk-users] Asterisk on virtual machine   Adam Robins
• [asterisk-users] Polycom Buddy Watch Broken with 2.0.1 Firmware?   Anthony Rodgers
• [asterisk-users] How big is *your* dialplan??   Anthony Rodgers
• [asterisk-users] FOP run control for CentOS/RHEL   Anthony Rodgers
• [asterisk-users] Polycom SP4000 ftp problem   Anthony Rodgers
• [asterisk-users] Becoming a User on IRC   Anthony Rodgers
• [asterisk-users] ASterisk Start problem   Anthony Rodgers
• [asterisk-users] need help using tftp for polycom 501   Anthony Rodgers
• [asterisk-users] Asterisk PBX to a Nortel MICS PBX   Anthony Rodgers
• [asterisk-users] Vancouver Asterisk User Group   Anthony Rodgers
• [asterisk-users] Enterprise Asterisk User Group   Anthony Rodgers
• [asterisk-users] problem in ooh323   Andre Luiz Martins Rodrigues
• [asterisk-users] ooh323 error   Andre Luiz Martins Rodrigues
• [asterisk-users] Help in MySQL + Asterisk.   Brian Rogan
• [asterisk-users] astcc help-pleasssssseeee   Brian Rogan
• [asterisk-users] how to play pre-recorded file in meetme conference   Brian Rogan
• [asterisk-users] CDR - mysql with asterisk 1.2.12 not working   Brian Rogan
• [asterisk-users] single conference, multiple numbers   Brian Rogan
• [asterisk-users] 1.4 and slow sound playback   Brian Rogan
• [asterisk-users] simultaneous ring - call groups or queues or something else?   Brian Rogan
• [asterisk-users] Server Recommendations   Carlos Rojas
• [asterisk-users] blind transfers with IP Polycom 501   Jeronimo Romero
• [asterisk-users] blind transfers with IP Polycom 501   Jeronimo Romero
• [asterisk-users] blind transfers with IP Polycom 501   Jeronimo Romero
• [asterisk-users] blind transfers with IP Polycom 501   Jeronimo Romero
• [asterisk-users] dealing with blind transfers to invalid extensions   Jeronimo Romero
• [asterisk-users] Unique ID   Eric Rousse
• [asterisk-users] Digium vs. Sangoma   Brian Roy
• [asterisk-users] app_meetme not loading   Will Roy
• [asterisk-users] Cisco 7960 Skinny calling SIP phone   Will Roy
• [asterisk-users] Cisco 7960 Skinny calling SIP phone   Will Roy
• [asterisk-users] app_meetme not loading   Will Roy
• [asterisk-users] Bristuff vs. vISDN vs. mISDN for hfc card ?   Robert Rozman
• [asterisk-users] Detecting Busy AGI Extensions?   Matthew Rubenstein
• [asterisk-users] Asterisk Server : IDE HDD frequent crash   Matthew Rubenstein
• [asterisk-users] using asterisk to do remote control functions   Matthew Rubenstein
• [asterisk-users] Choice of soundfile format   Matthew Rubenstein
• [asterisk-users] asterisk upgrade   Simone Ruffilli
• [asterisk-users] Multiple TE110P cards in one chassis   Rur0un1
• [asterisk-users] Calling Functions from AEL2   Rushowr
• [asterisk-users] AEL2 Catching on?   Rushowr
• [asterisk-users] 2x* and realtime   Rushowr
• [asterisk-users] IAX2 thru NAT problem   Marian Rychtecky
• [asterisk-users] INVAL Messages   Marian Rychtecky
• [asterisk-users] INVAL Messages   Marian Rychtecky
• [asterisk-users] IAX2 show peers - description   Marian Rychtecky
• [asterisk-users] Re: IAX2 show peers - description   Marian Rychtecky
• [asterisk-users] Screen pop based on incoming DID   Rajkumar S
• [asterisk-users] India:Reliance - E1configuration using TE110P   Rajkumar S
• [asterisk-users] No voice for when using Playback and background   Rajkumar S
• Fwd: [asterisk-users] No voice for when using Playback and background   Rajkumar S
• [asterisk-users] SIP trunk from an Audiocodes mediant 1000   Rajkumar S
• [asterisk-users] Audiocodes MP-20x   Rajkumar S
• [asterisk-users] Audiocodes MP-20x   Rajkumar S
• [asterisk-users] Maximum talktime in a queue?   Rajkumar S
• [asterisk-users] Maximum talktime in a queue?   Rajkumar S
• [asterisk-users] Maximum talktime in a queue?   Rajkumar S
• [asterisk-users] Maximum talktime in a queue?   Rajkumar S
• [asterisk-users] S(x) - Hang up the call after 'x' seconds - Not working from queue   Rajkumar S
• [asterisk-users] S(x) - Hang up the call after 'x' seconds - Not working from queue   Rajkumar S
• [asterisk-users] S(x) - Hang up the call after 'x' seconds - Not working from queue   Rajkumar S
• [asterisk-users] cdr_addon_mysql.c - Asterisk 1.4 - Asterisk Addons   Nicolas S.
• [asterisk-users] Asterisk MGCP Centrex Experience/success story   Nicolas S.
• [asterisk-users] VoiceOne 0.4.0 released: a new web-based and open source GUI   Nicolas S.
• [asterisk-users] recommended application for salesman using asterisk   Yu Safin
• [asterisk-users] WRT54GP2 provisioning   Alberto Sagredo
• [asterisk-users] Re: 1.4 on mac OSX 10.4.8   Daniel Salama
• [asterisk-users] How to do Automatic Daylight Saving on Grandstream GXP-2000   Daniel Salama
• [asterisk-users] Accessing MySQL DB to set variables in Asterisk   Jean-Marc Salsa
• [asterisk-users] Accessing MySQL DB to set variables in Asterisk   Jean-Marc Salsa
• [asterisk-users] Accessing MySQL DB to set variables in Asterisk   Jean-Marc Salsa
• [asterisk-users] Asterisk Voicemail with ODBC Realtime Access   Jean-Marc Salsa
• [asterisk-users] Asterisk Voicemail with ODBC Realtime Access   Jean-Marc Salsa
• [asterisk-users] How to enable talking in chanspy while spying?   Thirumal Saminathan
• [asterisk-users] Reg. chanspy   Thirumal Saminathan
• [asterisk-users] how to config chanspy   Thirumal Saminathan
• [asterisk-users] pop a web page with DID in url   Michael Sampson
• [asterisk-users] pop a web page with DID in url   Michael Sampson
• [asterisk-users] swap CID with DID   Michael Sampson
• [asterisk-users] Inbound Callcenter with multiple DIDs   Michael Sampson
• [asterisk-users] Change the background of a conversation   Jan Eirik Sandnes
• [asterisk-users] Voicemail Press '0'   Savoy, Kevin - Williston, ND
• [asterisk-users] Voicemail Press '0'   Savoy, Kevin - Williston, ND
• [asterisk-users] random one way audio and noise between SIP phoneson same LAN   Scott Scecina
• [asterisk-users] random one way audio and noise betweenSIP phoneson same LAN   Scott Scecina
• [asterisk-users] random one way audio and noise betweenSIP phoneson same LAN   Scott Scecina
• [asterisk-users] Occasional one-way audio - Sangoma A101   Scott Scecina
• [asterisk-users] Occasional one-way audio - Sangoma A101   Scott Scecina
• [asterisk-users] tdm2400p question   Dave Schardin
• [asterisk-users] Looking for Wireless Heaset for Polycom 501   Dave Schardin
• [asterisk-users] tx_fax not getting entire fax   Bradley Schatz
• [asterisk-users] CAPI channel not available but nobody is using the system   Armin Schindler
• [asterisk-users] identifying Eicon Diva Server V-4BRI-8M vs 4BRI-8M   Armin Schindler
• [asterisk-users] CAPI channel not available but nobody is usingthe system   Armin Schindler
• [Asterisk-Users] chan_capi and bristuff   Armin Schindler
• [asterisk-users] Problems with chan-capi and Eicon Diva 4BRI   Armin Schindler
• [asterisk-users] Asterisk and ISDN and Hylafax   Armin Schindler
• [Asterisk-Users] chan_capi and bristuff   Armin Schindler
• [asterisk-users] Asterisk and ISDN and Hylafax   Armin Schindler
• [asterisk-users] Asterisk and ISDN and Hylafax   Armin Schindler
• [asterisk-users] Diva server 4bri and Portuguese BRI   Armin Schindler
• [asterisk-users] 1.4 beta voicemail warning   Carla Schroder
• [asterisk-users] macros reference?   Carla Schroder
• [asterisk-users] macros reference?   Carla Schroder
• [asterisk-users] 1.4 gsm files changed??   Carla Schroder
• [asterisk-users] 1.4 gsm files changed??   Carla Schroder
• [asterisk-users] identifying Eicon Diva Server V-4BRI-8M vs 4BRI-8M   Carla Schroder
• [asterisk-users] 1.4 downgrade   Carla Schroder
• [asterisk-users] Polycom SP4000 ftp problem   Carla Schroder
• [asterisk-users] make menuselect question- Module Embedding   Carla Schroder
• [asterisk-users] Polycom SP4000 ftp problem   Carla Schroder
• [asterisk-users] WRT54GP2 provisioning   Curt Shaffer
• [asterisk-users] hold drops audio   Curt Shaffer
• [asterisk-users] Polycom provision errors still! Arg!   Curt Shaffer
• [asterisk-users] Polycom provision errors still! Arg!   Curt Shaffer
• [asterisk-users] Polycom provision errors still! Arg!   Curt Shaffer
• [asterisk-users] Polycom provision errors still! Arg!   Curt Shaffer
• [asterisk-users] Polycom provision errors still! Arg!   Curt Shaffer
• [asterisk-users] Open SER or DUNDI   Curt Shaffer
• [asterisk-users] light web user interface   Curt Shaffer
• [asterisk-users] light web user interface   Curt Shaffer
• [asterisk-users] VoipSupply? [Semi-Urgent]   VaibhaV Sharma
• [asterisk-users] asterisk upgrade   Tim Sharp
• [asterisk-users] CAPI channel not available but nobody is using the system   Tim Sharp
• [asterisk-users] CAPI channel not available but nobody is usingthe system   Tim Sharp
• [asterisk-users] Zaptel problems   Shea, Matt
• [asterisk-users] Zaptel problems   Shea, Matt
• [asterisk-users] UPDATE: Zaptel problems   Shea, Matt
• [asterisk-users]FIX FOUND: Zaptel problems   Shea, Matt
• [asterisk-users] Asterisk Server : IDE HDD frequent crash   Stuart Sheldon
• [asterisk-users] CDR Help...   Stuart Sheldon
• [asterisk-users] Extremely choppy sound on some of our POTSnetwork calls; goes away with mute   Andrew Shelton
• [asterisk-users] GXP - 2000 BLF   Andrew Shelton
• [asterisk-users] Extremely choppy sound on some of ourPOTSnetwork calls; goes away with mute   Andrew Shelton
• [asterisk-users] two asterisk and one NBX system   Andrew Shelton
• [asterisk-users] Another Issue with 1.4   Shidan
• [asterisk-users] Asterisk 1.2.12.1 dies after asterisk -rx and never comes up again   Shidan
• [asterisk-users] Fax receive (rx fax) problem   Mohammad Shokuie
• [asterisk-users] rxfax problem ("Trainability test failed")   Mohammad Shokuie
• [asterisk-users] #Transfer - Timeout is configurable?   Mohammad Shokuie
• [Asterisk-Users] rxfax problem   Mohammad Shokuie
• [asterisk-users] E164 caller ID   Andreas Sikkema
• [asterisk-users] Why does it take at least 4 flipping days before asterisk tries to resolve a provider?   Andreas Sikkema
• [asterisk-users] alive check for HA constellation   Leonardo Silva
• [asterisk-users] Registration problem   Leonardo Silva
• [asterisk-users] Where is the PlayDTMF command?   Moises Silva
• [asterisk-users] Newbie question about meetme   Moises Silva
• [asterisk-users] Where is the PlayDTMF command?   Moises Silva
• [asterisk-users] Where is the PlayDTMF command?   Moises Silva
• [asterisk-users] two asterisk and one NBX system   Moises Silva
• [asterisk-users] Problem compiling libmfcr2.0.0.2 on Fedora Core 5   Moises Silva
• [asterisk-users] Error loading Unicall   Moises Silva
• [asterisk-users] Error loading Unicall   Moises Silva
• [asterisk-users] Re: Where is the PlayDTMF command?   Moises Silva
• [asterisk-users] How big is *your* dialplan??   Moises Silva
• [asterisk-users] Asterisk + E1 with MFC/R2 in Argentina?   Moises Silva
• [asterisk-users] compiling libunicall   Moises Silva
• [asterisk-users] Re: Where is the PlayDTMF command?   Moises Silva
• [asterisk-users] OriginateEvent reason codes.   Moises Silva
• [asterisk-users] Asterisk + E1 with MFC/R2 in Argentina?   Moises Silva
• [asterisk-users] compiling libunicall   Moises Silva
• [asterisk-users] Re: Where is the PlayDTMF command?   Moises Silva
• [asterisk-users] install MAGI   Moises Silva
• [asterisk-users] Is SQLite a replacement for Mysql while using ARA in 1.2.x   Moises Silva
• [asterisk-users] Waiting before executing System command   Moises Silva
• [asterisk-users] translate.c:88 powerof: Powerof 0: No power??   Moises Silva
• [asterisk-users] MFC/R2 patch problems   Moises Silva
• [asterisk-users] SIP RTP flow   Moises Silva
• [asterisk-users] Asterisk Call Statistics   Moises Silva
• [asterisk-users] Diva server 4bri and Portuguese BRI   Pedro Silva
• [asterisk-users] Diva server 4bri and Portuguese BRI   Pedro Silva
• [asterisk-users] Diva server 4bri and Portuguese BRI   Pedro Silva
• [asterisk-users] Diva server 4bri and Portuguese BRI   Pedro Silva
• [asterisk-users] Diva server 4bri and Portuguese BRI   Pedro Silva
• [asterisk-users] Diva server 4bri and Portuguese BRI   Pedro Silva
• [asterisk-users] RE: Welcome to the "asterisk-users" mailing list   Issac Simchayof
• [asterisk-users] Polycom 2.01 sip issues   Issac Simchayof
• [asterisk-users] CDR - mysql with asterisk 1.2.12 not working   Garth van Sittert
• [asterisk-users] Cubix / Firefly softphone and Asterisk   Garth van Sittert
• [asterisk-users] Configuration / dialplan problem   Kevin Smith
• [asterisk-users] best gui   Altus Snyman
• [asterisk-users] AstriCon Dallas in Two Weeks   Steven Sokol
• [asterisk-users] AstriCon Hotel Full - Here are some near-by alternates   Steven Sokol
• [asterisk-users] [ast-users] bridging active channels together   Roberto Sottile
• [asterisk-users] Unable to open Asterisk database   Andrea Spadaccini
• [asterisk-users] Accessing MySQL DB to set variables in Asterisk   Andrea Spadaccini
• [asterisk-users] TIMEOUT() function missing   Andrea Spadaccini
• [asterisk-users] Polycom 501 + Voicemail notification   Gerwin van de Steeg
• [asterisk-users] Bridging of PRI calls   Johann Steinwendtner
• [asterisk-users] Bridging of PRI calls   Johann Steinwendtner
• [asterisk-users] Bridging of PRI calls   Johann Steinwendtner
• [asterisk-users] T.38 faxing with spandsp and Grandstream HT.486   Johann Steinwendtner
• [asterisk-users] Options for moving to * friendly Business VSP   Al Stery
• [asterisk-users] Options for moving to * friendly Business VSP   Al Stery
• [asterisk-users] Re: Digium Interfaces in Tampa?   Steven
• [asterisk-users] Re: PRI issues   Steven
• [asterisk-users] Re: Outbound FXO call, getting "You must firstdial..."   Steven
• [asterisk-users] Re: Odd echo issue with speaker phone   Steven
• [asterisk-users] Re: Re: Digium TE410P LED problem   Steven
• [asterisk-users] Re: How do you like TrixBox?   Steven
• [asterisk-users] Re: some transfers dropped.   Steven
• [asterisk-users] Re: some transfers dropped.   Steven
• [asterisk-users] Re: Choice of soundfile format   Steven
• [asterisk-users] Re: Re: ECHO Cancellation in SIP Calls   Steven
• AW: [asterisk-users] PoE IP Phone   Christian Stredicke
• AW: [asterisk-users] Snom or Cisco Phones?   Christian Stredicke
• AW: [asterisk-users] NAT issue ? [More Info]   Christian Stredicke
• [asterisk-users] SPA942 quality for a Bank   VoIP Street
• [asterisk-users] DID failover   VoIP Street
• [asterisk-users] VoIP+RJ11 Phone existed?   Massimiliano Stucchi
• [asterisk-users] wrong outgoing caller id with PRI lines: maybe usecallingpres involved?   Massimiliano Stucchi
• [asterisk-users] echo cancellation on hard phones   Rosli Sukri
• [asterisk-users] Asterisk conferencing features   Rosli Sukri
• [asterisk-users] Asterisk conferencing features   Rosli Sukri
• [asterisk-users] Re: PoE IP Phone   Tech Support
• [asterisk-users] Blacklist to check http://whocalled.us   Noah Swint
• [asterisk-users] OT: Hand free solution recommandation   Administrator TOOTAI
• [asterisk-users] GPL Softphones   Administrator TOOTAI
• [asterisk-users] GPL Softphones   Administrator TOOTAI
• [asterisk-users] VoIP+RJ11 Phone existed?   Administrator TOOTAI
• [asterisk-users] IAX softphones   Administrator TOOTAI
• [asterisk-users] Embedded Asterisk   Administrator TOOTAI
• [asterisk-users] IAX Terminal   Administrator TOOTAI
• [asterisk-users] tdm2400p question   Carlo Taguinod
• [asterisk-users] Re: WiFi SIP handset with Bluetooth required   Sam Tam
• [asterisk-users] IAX Terminal   Neil Tancock
• [asterisk-users] Can't change Zaptel driver from FCC mode   Neil Tancock
• [asterisk-users] Can't change Zaptel driver from FCC mode   Neil Tancock
• [asterisk-users] Can't change Zaptel driver from FCC mode   Neil Tancock
• [asterisk-users] Where to best start looking for voicemail/moh sound quality problem?   Frank Tarczynski
• [asterisk-users] Where to best start looking for voicemail/moh sound quality problem?   Frank Tarczynski
• [asterisk-users] Siemens Hipath <-> asterisk, pri problem   Viktor Tatianin
• [asterisk-users] Reception Console   Viktor Tatianin
• [asterisk-users] Asterisk 1.0.12 released - Security Vulnerability Fix   Asterisk Development Team
• [asterisk-users] Asterisk 1.2.13 released - Security Vulnerability Fix   Asterisk Development Team
• [asterisk-users] Asterisk 1.4.0-beta3 released!   Asterisk Development Team
• [asterisk-users] AGI scripts   MBIT Technologies
• [asterisk-users] Asterisk both behind a NAT and outside at the same time   Brad Templeton
• [asterisk-users] Asterisk both behind a NAT and outside at the same time   Brad Templeton
• [asterisk-users] Asterisk both behind a NAT and outside at the same time   Brad Templeton
• [asterisk-users] Opinions on the best wholesale origination/term providers   Brad Templeton
• [asterisk-users] Asterisk both behind a NAT and outside at the same time   Brad Templeton
• [asterisk-users] Re: Opinions on the best wholesale origination/term providers   Brad Templeton
• [asterisk-users] Inbound Callcenter with multiple DIDs   Earl Terwilliger
• [asterisk-users] VoipSupply? [Semi-Urgent]   Shaw Terwilliger
• [asterisk-users] VoipSupply? [Semi-Urgent] (Big Apology)   Shaw Terwilliger
• [asterisk-users] Asterisk Directory listing   James Texter
• [asterisk-users] Polycom IP 501 phone randomly resets itself (loses Received call log, Missed calls, placed calls)   James Texter
• [asterisk-users] Astricon - post show Saturday?   James Texter
• [asterisk-users] Dial-9 (was Extension Numbering)   Matthew Thompson
• [asterisk-users] CALEA support within asterisk?   Matthew Thompson
• [asterisk-users] Echo Cancel Cards   Matthew Thompson
• [asterisk-users] Stripping digits on internal calls   Matthew Thompson
• [asterisk-users] considering purchasing a t1 card, any recommendations?   Matthew Thompson
• [asterisk-users] porting numbers in UK telewest/bt/adept   Matthew Thompson
• [asterisk-users] Re: sending fax with chan-capi   Stefan Tichy
• [asterisk-users] SPA 3102   Tim
• [asterisk-users] SPA 3102   Tim
• [asterisk-users] Where is the PlayDTMF command?   Jan du Toit
• [asterisk-users] Re: Where is the PlayDTMF command?   Jan du Toit
• [asterisk-users] Where is the PlayDTMF command?   Jan du Toit
• [asterisk-users] Re: Where is the PlayDTMF command?   Jan du Toit
• [asterisk-users] Re: Where is the PlayDTMF command?   Jan du Toit
• [asterisk-users] Re: Where is the PlayDTMF command?   Jan du Toit
• [asterisk-users] OriginateEvent reason codes.   Jan du Toit
• [asterisk-users] SPA942 quality for a Bank   Tom
• [asterisk-users] AGI() in 1.2 and 1.4   Steve Totaro
• [asterisk-users] NFAS Not Passing Audio on B-chan 48,72,96   Steve Totaro
• [asterisk-users] NFAS Not Passing Audio on B-chan 48,72,96   Steve Totaro
• [Fwd: Re: [asterisk-users] NFAS Not Passing Audio on B-chan 48, 72, 96]   Steve Totaro
• [asterisk-users] asterisk 1.2.12 lost phone registrations today... why?   Steve Totaro
• [asterisk-users] NFAS Not Passing Audio on B-chan 48,72,96   Steve Totaro
• [asterisk-users] NFAS Not Passing Audio on B-chan 48,72,96   Steve Totaro
• [asterisk-users] average waiting time in a queue   Steve Totaro
• [asterisk-users] Reception Console   Steve Totaro
• [asterisk-users] Unique call ID's across several systems   Steve Totaro
• [asterisk-users] Primary D-Channel & channal numbers....   Tristan
• [asterisk-users] Resampling Audio for use with Asterisk   Tristan
• [asterisk-users] Bridging Video Calls using Zap   Tristan
• [asterisk-users] How big is *your* dialplan??   Justin Tunney
• [asterisk-users] Cisco CCM - Asterisk   Alyed Tzompa
• [asterisk-users] Cisco CCM - Asterisk   Alyed Tzompa
• [asterisk-users] Cisco CCM - Asterisk   Alyed Tzompa
• [asterisk-users] Cisco CCM - Asterisk   Alyed Tzompa
• [asterisk-users] Re: duplicate "ghost" calls with long duration   Alyed Tzompa
• [asterisk-users] Re: ECHO Cancellation in SIP Calls   Alyed Tzompa
• [asterisk-users] Live creation of trunk groups   Alyed Tzompa
• [asterisk-users] Good phones for outside of the office?   Alyed Tzompa
• [asterisk-users] Fax detection ...   Steve Underwood
• [asterisk-users] Fax detection ...   Steve Underwood
• [asterisk-users] Spandsp and tif   Steve Underwood
• [Asterisk-Users] rxfax problem   Steve Underwood
• [Asterisk-Users] rxfax problem   Steve Underwood
• [asterisk-users] asterisk and HMP   Steve Underwood
• [Asterisk-Users] rxfax problem   Steve Underwood
• [asterisk-users] need help using tftp for polycom 501   Marlin Unruh
• [asterisk-users] need help using tftp for polycom 501   Marlin Unruh
• [asterisk-users] Go to DIGIUMBOARDS.COM   VOICEIN at aol.com
• [asterisk-users] sending fax with chan-capi   Jens Vagelpohl
• [asterisk-users] sending fax with chan-capi   Jens Vagelpohl
• [asterisk-users] Re: sending fax with chan-capi   Jens Vagelpohl
• [asterisk-users] Asterisk manager   Michel Vaillancourt
• [asterisk-users] AEL Question   Michel R Vaillancourt
• [asterisk-users] Re: AEL Question   Michel R Vaillancourt
• [asterisk-users] Voicemail MWI   Race Vanderdecken
• [asterisk-users] Digium TE110P   Julian Varanini
• [asterisk-users] TE110P Card   Julian Varanini
• [asterisk-users] New Version of "Tycho" Voicemail Manager released   Arnd Vehling
• [asterisk-users] asterisk gui sans live cd   Arnd Vehling
• [asterisk-users] Voicemail maintenance   Arnd Vehling
• [asterisk-users] bristuff-0.3.0-PRE-1u for Asterisk 1.2.13 on junghanns downloads now   Vidar
• [asterisk-users] Zaptel problems   Bernardo Vieira
• [asterisk-users] Call Interception   Bernardo Vieira
• [asterisk-users] Wouldn't Tri-tone detection in Dial() be cool?   Bernardo Vieira
• [asterisk-users] Echo Cancel Cards   Bernardo Vieira
• [asterisk-users] +Ura +md3200 nao encaminha ligacao   Bernardo Vieira
• [asterisk-users] How to make this easier   Tom Vile
• [asterisk-users] How to make this easier   Tom Vile
• [asterisk-users] How big is *your* dialplan??   Tom Vile
• [asterisk-users] Polycom HDVoice   Tom Vile
• [asterisk-users] Is 1.2.12.1 production ready   Tom Vile
• [asterisk-users] Re: Is 1.2.12.1 production ready (Mauro Zanin)   Tom Vile
• [asterisk-users] considering purchasing a t1 card, any recommendations?   Tom Vile
• [asterisk-users] how to config chanspy   Tom Vile
• [asterisk-users] Access Denied on a Windows share   Tom Vile
• [asterisk-users] CID Issues   Tom Vile
• [asterisk-users] CID Issues   Tom Vile
• [asterisk-users] Mac OS X Desktop / Asterisk integration?   Tom Vile
• [asterisk-users] Forwarding recorded calls to Voicemail   Tom Vile
• [asterisk-users] Asterisk and ARI (Aterisk Recording Interface) integration problem   Tom Vile
• [asterisk-users] Asterisk web interface is not parsing the PHP pages   Tom Vile
• [asterisk-users] Compatability   Tom Vile
• [asterisk-users] channel_find_locked   Alexandru Voinescu
• [asterisk-users] Asterisk does not bridge zap channels on outgoing calls   Alexandru Voinescu
• [asterisk-users] Digium vs. Sangoma   Vulture
• [asterisk-users] snom 360: how to make record button working ?   Terry Wade
• [asterisk-users] T1 Passthrough   Dennis Walker
• [asterisk-users] Issues with Asterisk 1.4 Beta   Jason Walker
• [asterisk-users] 1.4 downgrade   Jason Walker
• [asterisk-users] DTMF / Silence issues   Jason Walker
• [asterisk-users] Escape from Voicemail   Jason Walker
• [asterisk-users] DTMF Tones   Jason Walker
• [asterisk-users] Callmanager 3.3(5) and Asterisk with ooh323 problem   Ward, Bill
• [asterisk-users] Voicemail help   Ward, Bill
• [asterisk-users] Voicemail help   Ward, Bill
• [asterisk-users] Becoming a User on IRC   Ward, Bill
• [asterisk-users] Callmanager 3.3(5) and Asterisk with ooh323   Ward, Bill
• [asterisk-users] Callmanager 3.3(5) and Asterisk with ooh323   Ward, Bill
• [asterisk-users] Voicemail help   Ward, Bill
• [asterisk-users] Callmanager 3.3(5) and Asterisk with ooh323   Ward, Bill
• [asterisk-users] Callmanager 3.3(5) and Asterisk with ooh323   Ward, Bill
• [asterisk-users] Callmanager 3.3(5) and Asterisk with ooh323   Ward, Bill
• [asterisk-users] asterisk-1.2.13 fails to 'Make' in Fedore Core 6'   Ward, Bill
• [asterisk-users] Zaptel problems   Watkins, Bradley
• [asterisk-users] Function ENUMLOOKUP   Watkins, Bradley
• [asterisk-users] voicemail usernames can't begin with "j" letter?   Watkins, Bradley
• [asterisk-users] CDR - mysql with asterisk 1.2.12 not working   Hermann Wecke
• [asterisk-users] CDR - mysql with asterisk 1.2.12 not working   Hermann Wecke
• [asterisk-users] Bridging Video Calls using Zap   Steffen Weinreich
• [asterisk-users] Bridging Video Calls using Zap   Steffen Weinreich
• [asterisk-users] Asterisk 1.4 / install app_bundle problems   Pamela Weis
• [asterisk-users] Generate Random Numbers in dialplan   Jon Weisman
• [asterisk-users] Generate Random Numbers in dialplan   Jon Weisman
• [asterisk-users] Generate Random Numbers in dialplan   Jon Weisman
• [asterisk-users] Generate Random Numbers in dialplan   Jon Weisman
• [asterisk-users] Generate Random Numbers in dialplan   Jon Weisman
• [asterisk-users] Re: Generate Random Numbers in dialplan   Jon Weisman
• [asterisk-users] Re: Generate Random Numbers in dialplan   Jon Weisman
• [asterisk-users] Echo problems on ISDN. (mainly incoming call s)   Ron Wellsted
• [Asterisk-Users] Would you support a Bristuff mailing list ?   Ron Wellsted
• [asterisk-users] Attended transfer hanging PRI channel   Michael Welter
• [asterisk-users] Attended transfer hanging PRI channel   Michael Welter
• [asterisk-users] Lumenvox speech recognition   Michael Welter
• [asterisk-users] Is 1.2.12.1 production ready   Dustin Wenz
• [asterisk-users] Re: 1.4 on mac OSX 10.4.8   Dustin Wenz
• [asterisk-users] Re: 1.4 on mac OSX 10.4.8   Dustin Wenz
• [asterisk-users] unable to call AT&T audio conference bridge   BJ Weschke
• [asterisk-users] Call Center requirements   BJ Weschke
• [asterisk-users] Asterisk 1.2.12.1 dies after asterisk -rx and never comes up again   BJ Weschke
• [asterisk-users] VoipSupply? [Semi-Urgent]   BJ Weschke
• [asterisk-users] Polycom boot error   BJ Weschke
• [asterisk-users] Polycom boot error   BJ Weschke
• [asterisk-users] Multiple queue_log files based on queue - is it possible??   BJ Weschke
• [asterisk-users] Cisco 7971G-GE & SEP{MAC}.cnf.xml   BJ Weschke
• [asterisk-users] Multiple TE110P cards in one chassis   Thermal Wetland
• [asterisk-users] Problem with 3-way calls from a Sipura ATA   Whisker, Peter
• [asterisk-users] random one way audio and noise between SIP phoneson same LAN   Derek Whitten
• [asterisk-users] Why is this happening?   Derek Whitten
• [asterisk-users] +Ura +md3200 nao encaminha ligacao   Tux Wi-FI
• [asterisk-users] Understanding NAT Traversal   Cullin J. Wible
• [asterisk-users] Increase VoiceMail Messages Recording Gain -AudioCalls are Ok   Cullin J. Wible
• [asterisk-users] what hardware and is it possible   Ady Wicaksono
• [asterisk-users] what hardware and is it possible   Ady Wicaksono
• [asterisk-users] Dial-9 (was Extension Numbering)   Eric "ManxPower" Wieling
• [asterisk-users] Dial-9 (was Extension Numbering)   Eric "ManxPower" Wieling
• [asterisk-users] no callerid from PSTN using TDM2400P   Eric "ManxPower" Wieling
• [asterisk-users] Transfer feature - howto?   Eric "ManxPower" Wieling
• [asterisk-users] Unknown signalling method 'pri_cpe'   Eric "ManxPower" Wieling
• [asterisk-users] Transfer feature - howto?   Eric "ManxPower" Wieling
• [asterisk-users] disabling hardware echo can on tdm2400p   Eric "ManxPower" Wieling
• [asterisk-users] Tellabs and a PRI   Eric "ManxPower" Wieling
• [asterisk-users] Range Operator   Eric "ManxPower" Wieling
• [asterisk-users] Voicemail Press '0'   Eric "ManxPower" Wieling
• [asterisk-users] Voicemail Press '0'   Eric "ManxPower" Wieling
• [asterisk-users] RE: Welcome to the "asterisk-users" mailing list   Eric "ManxPower" Wieling
• [asterisk-users] TDM400P incoming route for DID   Eric "ManxPower" Wieling
• [asterisk-users] TDM400P incoming route for DID   Eric "ManxPower" Wieling
• [asterisk-users] TDM400P incoming route for DID   Eric "ManxPower" Wieling
• [asterisk-users] asterisk to asterisk DID extentions   Eric "ManxPower" Wieling
• [asterisk-users] asterisk 1.2.12 lost phone registrations today... why?   Eric "ManxPower" Wieling
• [asterisk-users] asterisk 1.2.12 lost phone registrations today... why?   Eric "ManxPower" Wieling
• [asterisk-users] 1.2.12.1 crashing   Eric "ManxPower" Wieling
• [asterisk-users] How to send correct Caller ID on PRI   Eric "ManxPower" Wieling
• [asterisk-users] unauthenticated calls   Eric "ManxPower" Wieling
• [asterisk-users] Polycom IP 501 message light   Eric "ManxPower" Wieling
• [asterisk-users] Switchtype,Signalling,rxwink warnings   Eric "ManxPower" Wieling
• [asterisk-users] How do I figure out where this connection is coming from?   Eric "ManxPower" Wieling
• [asterisk-users] Psst... Top secret information: Codename Pineapple   Eric "ManxPower" Wieling
• [asterisk-users] Psst... Top secret information: Codename Pineapple   Eric "ManxPower" Wieling
• [asterisk-users] Why is this happening?   Eric "ManxPower" Wieling
• [asterisk-users] Why is this happening?   Eric "ManxPower" Wieling
• [asterisk-users] nat auto detect ?   Eric "ManxPower" Wieling
• [asterisk-users] considering purchasing a t1 card, any recommendations?   Eric "ManxPower" Wieling
• [asterisk-users] Stopping putgoing calls after working hours   Eric "ManxPower" Wieling
• [asterisk-users] Why is this happening?   Eric "ManxPower" Wieling
• [asterisk-users] Asterisk hangs up on incoming analog calls after a while   Eric "ManxPower" Wieling
• [asterisk-users] voicemail usernames can't begin with "j" letter?   Eric "ManxPower" Wieling
• [asterisk-users] getting DID info..   Eric "ManxPower" Wieling
• [asterisk-users] getting DID info..   Eric "ManxPower" Wieling
• [asterisk-users] getting DID info..   Eric "ManxPower" Wieling
• [asterisk-users] getting DID info..   Eric "ManxPower" Wieling
• [asterisk-users] voicemail usernames can't begin with "j" letter?   Eric "ManxPower" Wieling
• [asterisk-users] Re: checking 'voicemail" externally - doesn't work   Eric "ManxPower" Wieling
• [asterisk-users] Re: checking 'voicemail" externally - doesn't work   Eric "ManxPower" Wieling
• [asterisk-users] Re: checking 'voicemail" externally - doesn't work   Eric "ManxPower" Wieling
• [asterisk-users] Re: checking 'voicemail" externally - doesn't work   Eric "ManxPower" Wieling
• [asterisk-users] Why does it take at least 4 flipping days before asterisk tries to resolve a provider?   Eric "ManxPower" Wieling
• [asterisk-users] CID Issues   Eric "ManxPower" Wieling
• [asterisk-users] Cisco 2621 NM-HDV VWIC-1MFT1   Eric "ManxPower" Wieling
• [asterisk-users] Re: checking 'voicemail" externally - doesn't work   Eric "ManxPower" Wieling
• [asterisk-users] Polycom SP4000 ftp problem   Eric "ManxPower" Wieling
• [asterisk-users] CSU Support on Digium T1 Cards   Eric "ManxPower" Wieling
• [asterisk-users] FXO Cards vs. Channel bank with T1   Eric "ManxPower" Wieling
• [asterisk-users] overlap of zap trunk groups   Eric "ManxPower" Wieling
• [asterisk-users] overlap of zap trunk groups   Eric "ManxPower" Wieling
• [asterisk-users] DTMF Tones   Eric "ManxPower" Wieling
• [asterisk-users] Dynamic Codec Selection   Wildheart
• [asterisk-users] Re: Dynamic Codec Selection   Wildheart
• [asterisk-users] Re: Dynamic Codec Selection   Wildheart
• [asterisk-users] Cisco 7971G-GE & SEP{MAC}.cnf.xml   Kelvin Williams
• [asterisk-users] Newbie Questions   Ken Williams
• *****SPAM***** [asterisk-users] Re: Newbie Questions   Ken Williams
• [asterisk-users] single conference, multiple numbers   Mike Williams
• [asterisk-users] single conference, multiple numbers   Mike Williams
• [asterisk-users] SIP RTP flow   Mike Williams
• [asterisk-users] Call Center requirements   Stephen Wingfield
• [asterisk-users] ChanIsAvail() in 1.2.12.1   Thomas Winter
• [asterisk-users] Asterisk and ISDN and Hylafax   Thomas Winter
• [asterisk-users] Asterisk and ISDN and Hylafax   Thomas Winter
• [asterisk-users] Asterisk behind NAT and without portforwarding for rtp   Thomas Winter
• [asterisk-users] Asterisk behind NAT and without portforwarding forrtp   Thomas Winter
• [asterisk-users] Real-time and priority "n"   Ronald Wiplinger
• [asterisk-users] Re: Real-time and priority "n"   Ronald Wiplinger
• [asterisk-users] Adit 600 resetting   Don Wisdom
• [asterisk-users] Polycom IP500 Problems   Don Wisdom
• R: [asterisk-users] Zaphfc woth florz patch   Henrik Woffinden
• [asterisk-users] Reception Console   Henrik Woffinden
• [asterisk-users] Re: ZapHFC & quadBRI D-Channel going down randomly   Henrik Woffinden
• [asterisk-users] o extension for voicemail app   Wing Wong
• [asterisk-users] Building the Perfect Box   Conrad Wood
• [asterisk-users] asterisk crash in res_features.c   Conrad Wood
• [asterisk-users] asterisk upgrade   Conrad Wood
• [asterisk-users] Monitor stops recording midstream?   Conrad Wood
• [asterisk-users] Recording from a script   Conrad Wood
• [asterisk-users] Why the MusicOnHold sound so soft?   Conrad Wood
• [asterisk-users] what hardware and is it possible   Conrad Wood
• [asterisk-users] Locking phones at night...   Conrad Wood
• [asterisk-users] IVR problem   Conrad Wood
• [asterisk-users] Locking phones at night...   Conrad Wood
• [asterisk-users] Locking phones at night...   Conrad Wood
• [asterisk-users] gotoiftime and Macro question   Conrad Wood
• [asterisk-users] considering purchasing a t1 card, any recommendations?   Conrad Wood
• [asterisk-users] Unique call ID's across several systems   Conrad Wood
• [asterisk-users] Linksys SPA3102 - PSTN hangup detection in the UK   Conrad Wood
• [asterisk-users] Linksys SPA3102 - PSTN hangup detection in the UK   Conrad Wood
• [asterisk-users] Linksys SPA3102 - PSTN hangup detection in the UK   Conrad Wood
• [asterisk-users] Linksys SPA3102 - PSTN hangup detection in the UK   Conrad Wood
• [asterisk-users] update_header: Unable to find our position   Conrad Wood
• [asterisk-users] Choice of soundfile format   Conrad Wood
• [asterisk-users] Choice of soundfile format   Conrad Wood
• [asterisk-users] porting numbers in UK telewest/bt/adept   Conrad Wood
• [asterisk-users] ECHO Cancellation in SIP Calls   Conrad Wood
• [asterisk-users] SIP v IAX2   Conrad Wood
• [asterisk-users] ECHO Cancellation in SIP Calls   Conrad Wood
• [asterisk-users] porting numbers in UK telewest/bt/adept   Conrad Wood
• [asterisk-users] porting numbers in UK telewest/bt/adept   Conrad Wood
• [asterisk-users] No zap* commands?   Conrad Wood
• [asterisk-users] Server Recommendations   Conrad Wood
• [asterisk-users] Snom or Cisco Phones?   Conrad Wood
• [asterisk-users] T1 pricing in Oz   Isaac Xiao
• [asterisk-users] How do I configure Asterisk if I need to run Mysql server on second Linux   Tielin Xu
• [asterisk-users] Add second account to Xlite 3.0   Tielin Xu
• [asterisk-users] Need recommendation for SIP hard phones   Tielin Xu
• [asterisk-users] Add second account to Xlite 3.0   Tielin Xu
• [asterisk-users] How to deploy a PBX in such a condition ?   Bo Yang
• [asterisk-users] SIP Server   Imran M Yousuf
• [asterisk-users] 12 port FXx PCI card   Yusuf
• [asterisk-users] Student Research - Asterisk H323 Video   Yusuf
• [asterisk-users] unauthenticated calls   Sylvain ZUCCA
• [asterisk-users] Which IP Phone is good to use at reception desk?   Zeeshan Zakaria
• [asterisk-users] Call Forwarding not working for extension in queue, why?   Zeeshan Zakaria
• [asterisk-users] Call Forwarding not working for extension in queue, why?   Zeeshan Zakaria
• [asterisk-users] Which IP Phone is good to use at reception desk?   Zeeshan Zakaria
• [asterisk-users] Asterisk: Can anybody forward anybody's extension?   Zeeshan Zakaria
• [asterisk-users] How to send correct Caller ID on PRI   Zeeshan Zakaria
• [asterisk-users] Zaptel not detecting Tormenta2 PRI Interface card   Zeeshan Zakaria
• [asterisk-users] How to make different ext using different trunks?   Zeeshan Zakaria
• [asterisk-users] meet me   Zeeshan Zakaria
• [asterisk-users] Trixbox installation - ZAP channels becoming upresponsive   Zeeshan Zakaria
• [asterisk-users] How to do Automatic Daylight Saving on Grandstream GXP-2000   Zeeshan Zakaria
• [asterisk-users] Re: How to do Automatic Daylight Saving on Grandstream GXP-2000   Zeeshan Zakaria
• [asterisk-users] Asterisk and ARI (Aterisk Recording Interface) integration problem   Zeeshan Zakaria
• [asterisk-users] Asterisk and ARI (Aterisk Recording Interface) integration problem   Zeeshan Zakaria
• [asterisk-users] Re: DTMF Tones   Zeeshan Zakaria
• [asterisk-users] Re: DTMF Tones   Zeeshan Zakaria
• [asterisk-users] Voicemail app. not working...   Mauro Zanin
• [asterisk-users] Re: Is 1.2.12.1 production ready (Mauro Zanin)   Mauro Zanin
• [asterisk-users] Call bridged, but no sound   Norbert Zawodsky
• [asterisk-users] Call bridged, but no sound   Norbert Zawodsky
• [asterisk-users] Call bridged, but no sound   Norbert Zawodsky
• *****SPAM***** Re: [asterisk-users] Asterisk + Huawei   Ma Zhiyong
• [asterisk-users] Re: Asterisk + Huawei   Ma Zhiyong
• SV: [asterisk-users] lost packets when bridging zap and iax   Dmitry Zhukovski
• [asterisk-users] fonality acquires trixbox (asterisk@home) ?   Zoa
• [asterisk-users] isdn cross-over ...   Zoa
• [asterisk-users] Kind of OT : Europeans going to Astricon   Zoa
• [asterisk-users] Asterisk on 64bit xeon   Zoa
• [asterisk-users] GPL Softphones   Zoa
• [asterisk-users] Multiple TE110P cards in one chassis   Zoa
• [asterisk-users] One way audio on chan_gtalk   Zoa
• [asterisk-users] what hardware and is it possible   Zoa
• [asterisk-users] One way audio on chan_gtalk   Zoa
• [asterisk-users] IAX Terminal   Zoa
• [asterisk-users] say Asterisk to answer   Zoa
• [asterisk-users] considering purchasing a t1 card, any recommendations?   Zoa
• [asterisk-users] GPL Softphones   anban at ecntelecoms.com
• [asterisk-users] Quescom 400   anban at ecntelecoms.com
• I: [asterisk-users] Sip answer one side , ring other side   antonio
• [asterisk-users] TNT Max Password reset   asterisk
• [asterisk-users] ZapHFC & quadBRI D-Channel going down randomly   asterisk
• [asterisk-users] channel.c: Avoided initial deadlock   asterisk
• [asterisk-users] Asterisk Directory listing   asterisk-user
• [asterisk-users] unable to call AT&T audio conference bridge   asterisk-user
• [asterisk-users] how to activate recording (automon)   asterisk at frameweb.it
• [asterisk-users] how to activate recording (automon)   asterisk at frameweb.it
• [asterisk-users] gotoiftime and Macro question   asterisk at frameweb.it
• [asterisk-users] RE gotoiftime and Macro question   asterisk at frameweb.it
• [asterisk-users] MINNESOTA: TwinCities Asterisk Users Group - Saturday October 14th 2006 - 10:30am   asterisk_help at iwishi.nu
• [asterisk-users] DOA IAXy?   bails
• [asterisk-users] Linksys PAP2 dial plan help please   bails
• [asterisk-users] Help: Problems about console color (FC5, XTerm)   zuo bf
• [asterisk-users] Siemens Hipath <-> asterisk, pri problem   bivio
• [asterisk-users] Siemens Hipath <-> asterisk, pri problem   bivio
• [asterisk-users] Siemens Hipath <-> asterisk, pri problem   bivio
• [asterisk-users] vGSM drivers updated (0.17.2)   matteo brancaleoni
• [asterisk-users] vGSM drivers updated (0.17.2)   matteo brancaleoni
• [asterisk-users] Vgsm driver 0.18.0 released today   matteo brancaleoni
• [asterisk-users] VoipSupply? [Semi-Urgent]   brian at txshirts.com
• [asterisk-users] (no subject)   broadbandvoice at comcast.net
• [asterisk-users] Voicemail help   broadbandvoice at comcast.net
• [asterisk-users] Test to list   burke at tailorhosting.com
• [asterisk-users] Why is this happening?   burke at tailorhosting.com
• [asterisk-users] Modifying SIP Stack   burke at tailorhosting.com
• [asterisk-users] digium compatibility notes   marek cervenka
• [asterisk-users] Problem with Digium 400P and asterisk 1.4   John covici
• [asterisk-users] Basic Conf   daniel
• [asterisk-users] Basic Conf   daniel
• [asterisk-users] two asterisk and one NBX system   jose diaz
• [asterisk-users] two asterisk and one NBX system   jose diaz
• [asterisk-users] IP Phones   jose diaz
• [asterisk-users] Asterisk PBX to a Nortel MICS PBX   dthurn
• [asterisk-users] Cisco CCM - Asterisk   end1r at comcast.net
• [asterisk-users] Real Time and Asterisk   f.zamboni
• [asterisk-users] TDM2400P wiring.   f6hqz-m at hamwlan.net
• [asterisk-users] New Asterisk StumbleUpon Group   f6hqz-m at hamwlan.net
• [asterisk-users] asterisk-1.2.13 fails to 'Make' in Fedore Core 6'   barbara figueirido
• [asterisk-users] Segmentation fault asterisk realtime problem   flavio
• [asterisk-users] Segmentation fault issue   flavio
• [asterisk-users] Please help me!!   flavio
• [asterisk-users] suggest a configuration   stan ford
• [asterisk-users] t1 voip to failover pri   stan ford
• [asterisk-users] t1 voip to failover pri   stan ford
• [asterisk-users] failed registration   stan ford
• [asterisk-users] Re: failed registration   stan ford
• [asterisk-users] Re: failed registration   stan ford
• [asterisk-users] commercial asterisk   stan ford
• [asterisk-users] password for vm users   stan ford
• [asterisk-users] password for vm users   stan ford
• [asterisk-users] password for vm users   stan ford
• [asterisk-users] voicemail issue   stan ford
• [asterisk-users] Extremely choppy sound on some of our POTS network calls; goes away with mute   sdgesa gaeharth
• [asterisk-users] Extremely choppy sound on some of our POTS network calls; goes away with mute   sdgesa gaeharth
• [asterisk-users] Extremely choppy sound on some of our POTSnetwork calls; goes away with mute   sdgesa gaeharth
• [asterisk-users] Extremely choppy sound on some of ourPOTSnetwork calls; goes away with mute   sdgesa gaeharth
• [asterisk-users] Extremely choppy sound on some of ourPOTSnetwork calls; goes away with mute   sdgesa gaeharth
• [asterisk-users] Sip Trunks   ggonzalez at telviso.com.ar
• [asterisk-users] Asterisk and Panasonic KX Model   ggonzalez at telviso.com.ar
• [asterisk-users] IP Phones   bilal ghayyad
• [asterisk-users] Video Conference   bilal ghayyad
• [asterisk-users] PoE IP Phone   bilal ghayyad
• [asterisk-users] using asterisk to do remote control functions   marvin horst
• [asterisk-users] Bandwidth requirements   hugolivude
• [asterisk-users] Bandwidth requirements   hugolivude
• [asterisk-users] Bandwidth requirements   hugolivude
• [asterisk-users] Understanding NAT Traversal   hugolivude
• [asterisk-users] Understanding NAT Traversal   hugolivude
• [asterisk-users] SIP problem - ACT p160s error   isamar at isamarmaia.org
• [asterisk-users] Asterisk stopps matching extensions after first digit   jbauer at teec.com
• [asterisk-users] Extension Matching with "Match As You Go" Dialing   jbauer at teec.com
• [asterisk-users] Asterisk Load balancing   jk
• [asterisk-users] Call Forwarding Using Asterisk   jk
• [asterisk-users] Call Forwarding Using Asterisk   jk
• [asterisk-users] Load balance Asterisk server, when it is a SIP client.   jk at bingoconsulting.com
• [asterisk-users] How do you like TrixBox?   joe, at j4computers
• [asterisk-users] Asterisk/VOIP to PSTN (?)   joe, at j4computers
• [asterisk-users] need help using tftp for polycom 501   joe, at j4computers
• [asterisk-users] SIP problem - ACT p160s error   joe, at j4computers
• [asterisk-users] SIP problem - ACT p160s error   joe, at j4computers (joea at j4computers.com)
• [asterisk-users] Redefinition of transfer   john
• [asterisk-users] snom 360: how to make record button working ?   noro kamen
• [asterisk-users] snom 360 - how to make record button working ?   noro kamen
• [asterisk-users] snom 360: how to make record button working ?   noro kamen
• [asterisk-users] snom 360: how to make record button working ?   noro kamen
• [asterisk-users] TDM400P incoming route for DID   kim at anmsinc.com
• [asterisk-users] Increase VoiceMail Messages Recording Gain - AudioCalls are Ok   kjcsb
• [asterisk-users] Asterisk as SIP Client   kmittal at bingoconsulting.com
• [asterisk-users] mgcp registration with asterisk   pottabathini ashok kumar
• [asterisk-users] detecting busy on queue transfer   lenz
• [asterisk-users] New tutorial - peering two * servers using IAX   lenz
• [asterisk-users] New tutorial - peering two * servers using IAX   lenz
• [asterisk-users] New tutorial - peering two * servers using IAX   lenz
• [asterisk-users] Tutorial: Simple queue and agent debug monitoring   lenz
• [asterisk-users] Inbound Callcenter with multiple DIDs   lenz
• [asterisk-users] Taking a Polycom IP601 home   Warren (mailing lists)
• [asterisk-users] Good phones for outside of the office?   Warren (mailing lists)
• [asterisk-users] Queues: roundrobin w/ reset ("circular call distribution")   lists.digium.com at tgice.com
• [asterisk-users] CID Issues   mail-lists
• [asterisk-users] AstFax Sending a Fax   mail-lists
• [asterisk-users] dialing external number within meetme   Bartosz Wegrzyn - maillists
• [asterisk-users] Optus PRI via DSL   mattf at rapidnet.net.au
• [asterisk-users] Optus PRI via DSL   mattf at rapidnet.net.au
• [asterisk-users] Re:UPDATE: Zaptel problems   mavince at optonline.net
• [asterisk-users] Re: PRI issues   mavince at optonline.net
• [asterisk-users] Re: NFAS Not Passing Audio on B-chan 48,72,96   mavince at optonline.net
• [asterisk-users] How to send correct Caller ID on PRI   mavince at optonline.net
• [asterisk-users] eagi-sphinx-test how and why   mawali at news.icns.com
• [asterisk-users] average waiting time in a queue   mbodbg at gmx.net
• [asterisk-users] Cisco 7970 SIP won't update?   mitcheloc
• [asterisk-users] Escape from Voicemail   mitcheloc
• [asterisk-users] Fxo box for asterisk ?   mitcheloc
• [asterisk-users] Snom or Cisco Phones?   mitcheloc
• [asterisk-users] Strange Zaptel Problem   nicu
• [asterisk-users] Cisco 7970 strange Xml , but upgrade success.   nigma nigmus
• [asterisk-users] member queue refresh   nik600
• [asterisk-users] asterisk upgrade   nik600
• [asterisk-users] Caller ID on Zap not always working   nrbwpi at gmail.com
• [asterisk-users] Building the Perfect Box   adebayo omo-dare
• [asterisk-users] t1 voip to failover pri   adebayo omo-dare
• [asterisk-users] t1 voip to failover pri   adebayo omo-dare
• [asterisk-users] sdsl   adebayo omo-dare
• [asterisk-users] Newbie question about meetme   omar parihuana
• [asterisk-users] Bandwidth requirements   omar parihuana
• [asterisk-users] Re: Bandwidth requirements   omar parihuana
• [asterisk-users] Asterisk Call Statistics   omar parihuana
• [asterisk-users] How do I figure out where this connection is coming from?   mike pham
• [asterisk-users] Fax detection ...   phil.dawson at marnock.com
• [asterisk-users] Fax detection ...   phil.dawson at marnock.com
• [asterisk-users] Caller ID on Zap not always working   phil.dawson at marnock.com
• [asterisk-users] Problem with CallerID (UK) TDM400P ( CID timed out waiting for ring )   phil.dawson at marnock.com
• [asterisk-users] Problem with CallerID (UK) TDM400P ( CID timed out waiting for ring )   phil.dawson at marnock.com
• [asterisk-users] Problem with incomming calls   phil.dawson at marnock.com
• [asterisk-users] How to forward DID to another Server   ram
• [asterisk-users] Asterisk act as a proxy ?   ram
• [asterisk-users] How can i store PAP2 or any device config in Asterisk   ram
• [asterisk-users] How can i store PAP2 or any device config in Asterisk   ram
• [asterisk-users] how to play background music   ram
• [asterisk-users] how i can do auto dialing using mysql   ram
• [asterisk-users] Some file aren't loaded its No file in that Directory.   ram
• [asterisk-users] Reception Console   ram
• [asterisk-users] Asterisk 1.2.13 make problem   ram
• [asterisk-users] Increase VoiceMail Messages Recording Gain - AudioCalls are Ok   ram
• [asterisk-users] zaptel 1.2.10 make problem   ram
• [asterisk-users] ASterisk Start problem   ram
• [asterisk-users] voicemail usernames can't begin with "j" letter?   rcarvalho at iric.up.pt
• [asterisk-users] SipAddHeader   ron.asterisk.users at mcleodnet.com
• [asterisk-users] Dropping extra frame of G.729 since we already have a VAD frame at the end   laurent schweizer
• [asterisk-users] g729 Codec for AMD Sempron   sdcharly at gmail.com
• [asterisk-users] g729 Codec for AMD Sempron   sdcharly at gmail.com
• [asterisk-users] sip provider not working   sdcharly at gmail.com
• [asterisk-users] Linksys PAP2 dial plan help please   sdcharly at gmail.com
• [asterisk-users] Linksys PAP2 dial plan help please   sdcharly at gmail.com
• [asterisk-users] Linksys PAP2 dial plan help please   sdcharly at gmail.com
• [asterisk-users] Registration problem   sergio.dippolito at incore-systems.com.ar
• [asterisk-users] fonality acquires trixbox (asterisk@home) ?   shadowym
• [asterisk-users] 1.2.12.1 crashing   shadowym
• [asterisk-users] Is 1.2.12.1 production ready   shadowym
• [asterisk-users] Is 1.2.12.1 production ready   shadowym
• [asterisk-users] Server Recommendations   shadowym
• [asterisk-users] snom 360: how to make record button working ?   sip
• [asterisk-users] E164 caller ID   sip
• [asterisk-users] E164 caller ID   sip
• [asterisk-users] E164 caller ID   sip
• [asterisk-users] E164 caller ID   sip
• [asterisk-users] Re: E164 caller ID   sip
• [asterisk-users] Conference is Not Working.... with OpenSER And Asterisk   sunkara
• [asterisk-users] Without ZapTel inferface or Card install , is Conference working or Not   sunkara
• [asterisk-users] How to disconnect in Conferenceing in between the Confermce .....   sunkara
• [asterisk-users] How to hung up , While in Conference going on.   sunkara
• [asterisk-users] Help in MySQL + Asterisk.   raviprakash sunkara
• [asterisk-users] Message count requested for mailbox 9002@from-sip but voicemail not loaded.   raviprakash sunkara
• [asterisk-users] SIP vz IAX...   raviprakash sunkara
• [asterisk-users] help this....   raviprakash sunkara
• [asterisk-users] Some file aren't loaded its No file in that Directory.   raviprakash sunkara
• [asterisk-users] Problem in Voice Message Storing...............   raviprakash sunkara
• [asterisk-users] Some Warning in Asterisk for Voicemail intgreting,   raviprakash sunkara
• [asterisk-users] H323 <-> SIP   tlott at gmx.net
• [asterisk-users] realtimeupdate error   unplug
• [asterisk-users] question about astdb   unplug
• [asterisk-users] question about CDR command   unplug
• [asterisk-users] question about CDR command   unplug
• [asterisk-users] forward several times   unplug
• [asterisk-users] Problem: Dial command with L option   unplug
• [asterisk-users] astdb error, please help   vivek at staff.ownmail.com
• [asterisk-users] astdb error, please help   vivek at staff.ownmail.com
• [asterisk-users] newbie astdb error, please help   vivek at staff.ownmail.com
• [asterisk-users] Unable to create/find SIP channel for this INVITE & Broadvoice   vol at kokezaru.net
• [asterisk-users] Re: Unable to create/find SIP channel for this INVITE & Broadvoice   vol at kokezaru.net
• [asterisk-users] PoE IP Phone   wendys at tiscali.de
• [asterisk-users] siemens hipath interoperability - PRI/Q.SIG - cardrecommendation   wendys at tiscali.de
• [asterisk-users] IVR for the called part (IVR inside out)   whois wes
• [asterisk-users] Cepstral/Swift TTS app   will at loopfree.net
• [asterisk-users] How big is *your* dialplan??   www.IPKall.com
• [asterisk-users] Asterisk Advice   yusuf
• [asterisk-users] Unknown signalling method 'pri_cpe'   yusuf
• [asterisk-users] asterisk-addons-1.2.4 Installation Problem   yusuf
• [asterisk-users] Inaccurate CDRs   yusuf
• [asterisk-users] Please explain these SIP errors   yusuf
• [asterisk-users] Multiple bridge attempts   yusuf
• [asterisk-users] Please help with these SIP errors   yusuf
• [asterisk-users] Asterisk Call Statistics   yusuf
• [asterisk-users] REQ: Astricon Pictures   zoachien at securax.org