July 2006 Archives by author

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Starting: Sat Jul 1 01:33:01 MST 2006
Ending: Mon Jul 31 23:53:26 MST 2006
Messages: 3645

• [asterisk-users] VoipNow 1.2.0 Beta   Francesco Peeters (Asterisk)
• [asterisk-users] Intercom mode on Polycom and/or SPA9xx   Brian Vincent (C)
• [asterisk-users] Extensions not busy showing as busy   Brian Vincent (C)
• [asterisk-users] Polycom - simpler transfers?   Brian Vincent (C)
• [asterisk-users] Testing 911?   Brian Vincent (C)
• [asterisk-users] an ATA with lamp support   Brian Vincent (C)
• [asterisk-users] "Legacy" analog data modems and Asterisk   Brian Vincent (C)
• [asterisk-users] Polycom 601 and Paging   Brian Vincent (C)
• [asterisk-users] Polycom 601 and Paging   Brian Vincent (C)
• [asterisk-users] A very lost newbie.   Brian Vincent (C)
• [asterisk-users] Intercom feature on Polycom phones   Brian Vincent (C)
• [asterisk-users] long distance ethernet & Asterisk   Brian Vincent (C)
• [asterisk-users] long distance ethernet & Asterisk   Brian Vincent (C)
• [asterisk-users] adding a voice conversation recording on a existingPBX system   Brian Vincent (C)
• [Asterisk-Users] Integrate asterisk with Database   Chris Mason (Lists)
• [asterisk-users] OT: Sipura SPA-3000 ATA Directing Calls to Asterisk   Chris Mason (Lists)
• [asterisk-users] Provider UNREACHABLE   Chris Mason (Lists)
• [asterisk-users] Provider UNREACHABLE   Chris Mason (Lists)
• [asterisk-users] Please suggest me Best VoIP Service Provider   Chris Mason (Lists)
• [asterisk-users] Server redundancy   Matt Riddell (NZ)
• [asterisk-users] NuFone, please send the log file   Matt Riddell (NZ)
• Re: [asterisk-users] Inc.com Names Mark Spencer of Digium to its “30 Under 30: America’s Coolest Young Entrepreneurs”   Matt Riddell (NZ)
• [asterisk-users] Re: Wrong account code from iax_buddies   Matt Riddell (NZ)
• [asterisk-users] Asterisk and VAD   Matt Riddell (NZ)
• [asterisk-users] Manager action "hold" missing?   Matt Riddell (NZ)
• [asterisk-users] Asterisk 1.2.10 and Zaptel 1.2.7 released!   Matt Riddell (NZ)
• [asterisk-users] Re: Asterisk 1.2.10 and Zaptel 1.2.7 released!   Matt Riddell (NZ)
• [asterisk-users] Asterisk 1.2.10 and Zaptel 1.2.7 released!   Matt Riddell (NZ)
• [asterisk-users] SRTP enabling   Matt Riddell (NZ)
• [asterisk-users] Injecting prerecorded audio into active call   Matt Riddell (NZ)
• [asterisk-users] Sphinx and Asterisk Integration, How?   Matt Riddell (NZ)
• [asterisk-users] Sphinx and Asterisk Integration, How?   Matt Riddell (NZ)
• [asterisk-users] Re: Asterisk 1.2.10 and Zaptel 1.2.7 released!   Matt Riddell (NZ)
• [asterisk-users] Two security holes fixed in latest versions of Asterisk   Matt Riddell (NZ)
• [asterisk-users] zaptel on dual processor, How?   Matt Riddell (NZ)
• [asterisk-users] Sphinx and Asterisk Integration, How?   Matt Riddell (NZ)
• [asterisk-users] Re: Asterisk 1.2.10 and Zaptel 1.2.7 released!   Matt Riddell (NZ)
• [asterisk-users] Two security holes fixed in latest versions of Asterisk   Matt Riddell (NZ)
• [asterisk-users] question about asterisk DB   Matt Riddell (NZ)
• [asterisk-users] SIP and podcasts   Matt Riddell (NZ)
• [asterisk-users] VoipNow 1.2.0 Beta   Matt Riddell (NZ)
• [asterisk-users] If you prefer to read this mail list as a forum ...   Matt Riddell (NZ)
• [asterisk-users] SIP and podcasts   Matt Riddell (NZ)
• [asterisk-users] VoipNow 1.2.0 Beta   Matt Riddell (NZ)
• [Asterisk-Users] The Asterisk console on a Dell D820 with Intel High Definition Audio.   William F. Acker WB2FLW +1-303-722-7209
• [Asterisk-Users] The Asterisk console on a Dell D820 with Intel High Definition Audio.   William F. Acker WB2FLW +1-303-722-7209
• [asterisk-users] call forwarding to mobile phone   Woodoo People .pGa!
• [asterisk-users] intel vs amd motherboards   Woodoo People .pGa!
• [asterisk-users] intel vs amd motherboards   Woodoo People .pGa!
• [Asterisk-Users] GSM gateway flooded cell - how to detect?   Woodoo People .pGa!
• [asterisk-users] Voismart GSM - no billsecs   Woodoo People .pGa!
• [asterisk-users] Codec Negotiation   Woodoo People .pGa!
• [asterisk-users] Codec Negotiation   Woodoo People .pGa!
• [asterisk-users] Zap DMTF detect error   Woodoo People .pGa!
• [asterisk-users] G729 License to Bridge calls through VOIP provider?   Woodoo People .pGa!
• [asterisk-users] Asterisk VOIP / Mikrotik   Woodoo People .pGa!
• [asterisk-users] Problem handling agents and queues vía RealTime   Álvaro Palma
• [asterisk-users] Which card do you recommend for heavy load application?   Álvaro Palma
• [asterisk-users] Snom 300 headset with static noise   Adrià Vidal
• [asterisk-users] Snom 300 headset with static noise   Adrià Vidal
• [asterisk-users] Snom 300 headset with static noise   Adrià Vidal
• [asterisk-users] Vicidial + Unicall mfcr2   Bruno de Assumpção Loureiro
• [asterisk-users] Vicidial + Unicall mfcr2   Bruno de Assumpção Loureiro
• [asterisk-users] DEBUG[13314]: Didn't get a frame from channel: SIP/   Denis Galvão - iSolve
• [Asterisk-Users] TDM Installation error   Josué Conti
• [Asterisk-Users]MediatrixclientauthenticationfailedEFAILURE_REASON_AUTHENTICATION   Josué Conti
• Re: [asterisk-users] Inc.com Names Mark Spencer of Digium to its “30 Under 30: America’s Coolest Young Entrepreneurs”   Josué Conti
• [asterisk-users] OT: Skype protocol cracked?   Josué Conti
• [asterisk-users] Fwd: Problem with chan_zap.so   Josué Conti
• [Asterisk-Users] trixbox 1.1 download   Kai Fürstenberg
• [Asterisk-Users] Need help with config-files   Kai Fürstenberg
• [Asterisk-Users] Need help with config-files   Kai Fürstenberg
• [Asterisk-Users] Need help with config-files   Kai Fürstenberg
• [Asterisk-Users] Help getting International Dialing setup in extensions.conf   Kai Fürstenberg
• [Asterisk-Users] Need help with config-files   Kai Fürstenberg
• [asterisk-users] intel vs amd motherboards   Kai Fürstenberg
• [asterisk-users] International Dialing setup in extensions.conf   Kai Fürstenberg
• [asterisk-users] intel vs amd motherboards   Kai Fürstenberg
• [asterisk-users] International Dialing setup in extensions.conf   Kai Fürstenberg
• [Asterisk-Users] request @default   Kai Fürstenberg
• [Asterisk-Users] request @default   Kai Fürstenberg
• [asterisk-users] Flash operator panel   Nicolás Gudiño
• [asterisk-users] Manager interface   Nicolás Gudiño
• [Asterisk-Users] IVR menus on different DIDs   Filip Drągowski
• [Asterisk-Users] Need help with config-files   Filip Drągowski
• [Asterisk-Users] Need help with config-files   Filip Drągowski
• [Asterisk-Users] Help getting International Dialing setup in extensions.conf   Filip Drągowski
• [Asterisk-Users] IVR menus on different DIDs   Filip Drągowski
• [Asterisk-Users] time variable   Filip Drągowski
• [asterisk-users] Unable to configure my DID number   Filip Drągowski
• [asterisk-users] Provider UNREACHABLE   Filip Drągowski
• [asterisk-users] Play sound to called party ...   Filip Drągowski
• [asterisk-users] IVR DTMF   Filip Drągowski
• [asterisk-users] IVR DTMF   Filip Drągowski
• [asterisk-users] priority problem   Filip Drągowski
• [asterisk-users] IVR DTMF   Filip Drągowski
• [asterisk-users] realtime oracle dialplan select   Filip Drągowski
• [asterisk-users] Bad luck installing bristuff on multiple Linux'es. Any one got a good-luck story I can repeat?   Filip Drągowski
• [asterisk-users] Bad luck installing bristuff on multiple Linux'es. Any one got a good-luck story I can repeat?   Filip Drągowski
• [asterisk-users] Bad luck installing bristuff on multiple Linux'es. Any one got a good-luck story I can repeat?   Filip Drągowski
• [asterisk-users] Two phone numbers, one SIP provider   Filip Drągowski
• [asterisk-users] Interested in IVR information   Filip Drągowski
• [asterisk-users] Two phone numbers, one SIP provider   Filip Drągowski
• [asterisk-users] TDM01B -1 FXO card not working.   Filip Drągowski
• [asterisk-users] Determining what gets written to the dst field for a CDR   Filip Drągowski
• [asterisk-users] Problem with call receiving (Asterisk+PSTN+Digium TDM04B)   Filip Drągowski
• [asterisk-users] Asterisk 1.2.10 - Continually Restarting Logger   Filip Drągowski
• [asterisk-users] Asterisk 1.2.10 - Continually Restarting Logger   Filip Drągowski
• [asterisk-users] Asterisk 1.2.10 - Continually Restarting Logger   Filip Drągowski
• [asterisk-users] Install asterisk-bristuff for Debian Linux   Filip Drągowski
• [Asterisk-Users] Need help with config-files   Kai Fürstenberg
• [Asterisk-Users] Need help with config-files   Kai Fürstenberg
• [asterisk-users] zaphfc - problem   Kai Fürstenberg
• [asterisk-users] WG: CDR Accounting wrong   René Enskat [Teamware GmbH]
• [asterisk-users] WG: CDR ist getting wrong status   René Enskat [Teamware GmbH]
• [asterisk-users] realtime oracle dialplan select   René Enskat [Teamware GmbH]
• [asterisk-users] CSTA support for asterisk   "Hans-Jürgen Brand"
• SV: [Asterisk-Users] Nokia E61   Jon Schøpzinsky
• SV: [asterisk-users] B2BUA Webbased and Click 2 dial apps   Jon Schøpzinsky
• SV: [asterisk-users] How to collect Call duration, Dialout Call files?   Jon Schøpzinsky
• SV: [asterisk-users] How to collect Call duration, Dialout Call files?   Jon Schøpzinsky
• SV: [asterisk-users] How to collect Call duration, Dialout Call files?   Jon Schøpzinsky
• SV: [asterisk-users] setting up an email to fax with asterisk   Jon Schøpzinsky
• [asterisk-users] IAX2 trunking problems   Jon Schøpzinsky
• [asterisk-users] IAX2 vs TDMoE   Jon Schøpzinsky
• [asterisk-users] IAX2 trunking problems   Jon Schøpzinsky
• SV: [asterisk-users] Sip phone settings set when user registers   Jon Schøpzinsky
• [asterisk-users] Cisco 7941/7961/7971 wont register with asterisk   Per Møller
• SV: [asterisk-users] Cisco 7941/7961/7971 wont register with asterisk   Per Møller
• [asterisk-users] Best method for detecting state of a sip trunk   Per Møller
• [asterisk-users] sendtext or sip message - where in RFC   Fabian Müller
• [asterisk-users] How do you recompile individual source modules?   Fabian Müller
• [asterisk-users] asterisk 1.4 download   Fabian Müller
• [asterisk-users] Asterisk Database   Tomislav Parčina
• [asterisk-users] Re: Channel Redirect   Tomislav Parčina
• [asterisk-users] Re: asterisk dual servers through iax: Accepting UNAUTHENTICATED call   Tomislav Parčina
• [asterisk-users] Re: Asterisk Database   Tomislav Parčina
• [asterisk-users] Re: Channel Redirect   Tomislav Parčina
• [asterisk-users] Re: Asterisk Database   Tomislav Parčina
• [asterisk-users] Re: How do you harden an Asterisk install?   Tomislav Parčina
• [asterisk-users] Re: Asterisk and VAD   Tomislav Parčina
• [asterisk-users] Re: 7970 SIP configs   Tomislav Parčina
• [asterisk-users] Re: Asterisk Database   Tomislav Parčina
• [asterisk-users] Re: Clearing variables in the dialplan?   Tomislav Parčina
• [asterisk-users] Re: Asterisk 1.2.10 and Zaptel 1.2.7 released!   Tomislav Parčina
• [asterisk-users] Re: call forwarding   Tomislav Parčina
• [asterisk-users] Re: Asterisk Database   Tomislav Parčina
• [asterisk-users] Re: asterisk sending connects when it shouldn't   Tomislav Parčina
• [asterisk-users] CentOS 4.3 and Zaptel-1.2.7   Tomislav Parčina
• [asterisk-users] Re: question about asterisk DB   Tomislav Parčina
• [asterisk-users] Re: question about asterisk DB   Tomislav Parčina
• [asterisk-users] Re: Hints to help me debug cdr_odbc not inserting   Tomislav Parčina
• [asterisk-users] Re: Fast busy after one digit dialled? - 7970 SIP 8.0.3   Tomislav Parčina
• [asterisk-users] RE: Redundant Ethernet   Tomislav Parčina
• [asterisk-users] Re: If you prefer to read this mail list as a forum ...   Tomislav Parčina
• [asterisk-users] Re: If you prefer to read this mail list asa forum ...   Tomislav Parčina
• [asterisk-users] Re: question about asterisk DB   Tomislav Parčina
• [Asterisk-Users] GSM gateway flooded cell - how to detect?   Colin Anderson
• [Asterisk-Users] HP Proliant server?   Idris AVCI
• [Asterisk-Users] Intel E7220 chipset?   Idris AVCI
• [Asterisk-Users] Intel E7220 chipset?   Idris AVCI
• [asterisk-users] IVR - Automatic Attendant database query   Idris AVCI
• [asterisk-users] SRTP enabling   Abdul
• [asterisk-users] Sending back a ring signal, SIP 180 i think.   Mark Ackroyd
• [asterisk-users] Server redundancy   Alejandro Acosta
• [asterisk-users] Server redundancy   Alejandro Acosta
• [asterisk-users] Dell PowerEdge 830   Jason Adams
• [asterisk-users] NuFone, please send the log file   Mark Adams
• [Asterisk-Users] SIP qualify time - best practices?   Rich Adamson
• [Asterisk-Users] TDM400 won't answer or dial. Help?   Rich Adamson
• [asterisk-users] International Dialing setup in extensions.conf   Rich Adamson
• [Asterisk-Users] Now that Nufone is dead...   Rich Adamson
• [asterisk-users] spa941 and sip "bye"   Rich Adamson
• [asterisk-users] Cisco SIP Firmware   Rich Adamson
• [asterisk-users] spa941 and sip "bye"   Rich Adamson
• [Asterisk-Users] Now that Nufone is dead...   Rich Adamson
• [asterisk-users] Suggesstion Required   Rich Adamson
• [asterisk-users] CallerID   Rich Adamson
• [asterisk-users] What's the story with X10*P FXO cards?   Rich Adamson
• [Asterisk-Users] Do you need a licence to connect a Ciscohardphone to Asterisk ?   Rich Adamson
• [asterisk-users] Re: What's the story with X10*P FXO cards?   Rich Adamson
• [asterisk-users] Re: What's the story with X10*P FXO cards?   Rich Adamson
• [asterisk-users] spa941 call pickup?   Rich Adamson
• [asterisk-users] spa941 call pickup?   Rich Adamson
• [asterisk-users] SPA-3000 XML Config File   Rich Adamson
• [asterisk-users] Asterisk version: 1.2.9.1 or older?   Rich Adamson
• [asterisk-users] Asterisk version: 1.2.9.1 or older?   Rich Adamson
• [asterisk-users] Asterisk version: 1.2.9.1 or older?   Rich Adamson
• [asterisk-users] Asterisk version: 1.2.9.1 or older?   Rich Adamson
• [asterisk-users] Asterisk version: 1.2.9.1 or older?   Rich Adamson
• [asterisk-users] How do you harden an Asterisk install?   Rich Adamson
• [asterisk-users] How do you harden an Asterisk install?   Rich Adamson
• [asterisk-users] Digium Zaptel volume issues   Rich Adamson
• [asterisk-users] Testing 911?   Rich Adamson
• [asterisk-users] an ATA with lamp support   Rich Adamson
• [asterisk-users] LinkSys SPA 2002 ATA hardphone UNREACHABLE...!!!   Rich Adamson
• [asterisk-users] LinkSys SPA 2002 ATA hardphone UNREACHABLE...!!!   Rich Adamson
• [asterisk-users] BudgeTone BT-102 not registering to Asterisk   Rich Adamson
• [asterisk-users] Help with sip debug?   Rich Adamson
• [asterisk-users] Help with sip debug?   Rich Adamson
• [asterisk-users] Source Clock   Rich Adamson
• [asterisk-users] Sipura ATA's Forwarding PSTN Calls to Asterisk   Rich Adamson
• [asterisk-users] Clocking Multiple T1 Cards   Rich Adamson
• [asterisk-users] Operator in Voicemail   Rich Adamson
• [asterisk-users] ata hook-flash   Rich Adamson
• [asterisk-users] Connecting branch offices through IPsec tunnel -- latency effects?   Rich Adamson
• [asterisk-users] Connecting branch offices through IPsec tunnel --latency effects?   Rich Adamson
• [asterisk-users] New message   Rich Adamson
• [asterisk-users] Clocking Multiple T1 Cards   Rich Adamson
• [asterisk-users] odd sound between SIP & IAX clients   Rich Adamson
• [asterisk-users] odd sound between SIP & IAX clients   Rich Adamson
• [asterisk-users] IAX2 Connection fails over time...   Rich Adamson
• [asterisk-users] IAX2 Connection fails over time...   Rich Adamson
• [asterisk-users] Source Directory of ASterisk   Rich Adamson
• [asterisk-users] Information about Softphone support G729 ?   Adrian
• [asterisk-users] Re: Information about Softphone support G729 ?   Adrian
• [asterisk-users] Ignore This   Diego Aguirre
• [asterisk-users] AGI tutorials   Akpome Akpoguma
• [asterisk-users] Automatic deletion of voicemail messages older than N days?   Sami Knudsen Al-Sayed
• [asterisk-users] H.323 an IAX   Asif Ali
• [asterisk-users] So many configuration files!   Larry Alkoff
• [Asterisk-Users] Recommendations for best Voicemail application manager?   Christopher Aloi
• [asterisk-users] Looking for carrier grade redundant solution   Christopher Aloi
• [asterisk-users] Looking for carrier grade redundant solution   Christopher Aloi
• [asterisk-users] registration process   Christopher Aloi
• [asterisk-users] Top Users in a MeetMe room???   Armando Alvarez
• [asterisk-users] Re: HP Proliant server?   Benny Amorsen
• [asterisk-users] Re: for you guys setting up customer offices...   Benny Amorsen
• [asterisk-users] Re: Asterisk version: 1.2.9.1 or older?   Benny Amorsen
• [asterisk-users] Re: Operator Console(s)/Shared Call Appearances   Benny Amorsen
• [asterisk-users] Re: If you prefer to read this mail list as a forum ...   Benny Amorsen
• [asterisk-users] Cisco SIP Firmware   Ryan Amos
• [asterisk-users] Asterisk and H.323   Aaron Anderson
• [asterisk-users] Asterisk and H.323   Aaron Anderson
• [asterisk-users] Asterisk and H.323   Aaron Anderson
• [Asterisk-Users] IP Authorization   Aaron Anderson
• [asterisk-users] Multiple Outbound SIP Trunks   Aaron Anderson
• [asterisk-users] Update for trunk?   Andreas Anderson
• [asterisk-users] PRI issues with telco access codes   Colin Anderson
• [asterisk-users] Phones cutting out.....again - PLEASE HELP!! !   Colin Anderson
• [asterisk-users] Phones cutting out.....again - PLEASE HELP!! !   Colin Anderson
• [Asterisk-Users] GSM gateway flooded cell - how to detect?   Colin Anderson
• [asterisk-users] Mitel 3300 + *   Colin Anderson
• [asterisk-users] Mitel 3300 + *   Colin Anderson
• [asterisk-users] Hitting # to Transfer out of a Queue   Andres
• [asterisk-users] Hitting # to Transfer out of a Queue   Andres
• [asterisk-users] Dial exited non-zero, only if PSTN/ZAP/E1 hangs up first. not if voip hangs up.   Andres
• [Asterisk-Users] Polycom Soundpoint IP 301 w/ MGCP   Cory Andrews
• [Asterisk-Users] Polycom Soundpoint IP 301 w/ MGCP   Cory Andrews
• [asterisk-users] Basic Asterisk Setup   Cory Andrews
• [asterisk-users] Re: Feasability of using * for small appartmentbuilding?   Cory Andrews
• [asterisk-users] Re: Feasability of using * for smallappartmentbuilding?   Cory Andrews
• [asterisk-users] Asterisk with Analogue cards   Cory Andrews
• [Asterisk-Users] Do you need a licence to connect a Ciscohardphone to Asterisk ?   Cory Andrews
• [Asterisk-Users] Do you need a licence to connect aCiscohardphone to Asterisk ?   Cory Andrews
• [asterisk-users] DTMF detection and Sangoma cards   Cory Andrews
• RE: [asterisk-users] Inc.com Names Mark Spencer of Digium to its "30 Under 30: America's Coolest Young Entrepreneurs"   Cory Andrews
• [asterisk-users] Cisco 7960 Call Waiting Beep   Cory Andrews
• [asterisk-users] Cisco 7960 Call Waiting Beep   Cory Andrews
• [asterisk-users] Germany VOIP provider   Thameem Ansari
• [asterisk-users] Germany VOIP provider   Thameem Ansari
• [asterisk-users] Germany VOIP provider   Thameem Ansari
• [asterisk-users] Germany VOIP provider   Thameem Ansari
• [asterisk-users] MWI from Asterisk to Meridian   kritikus Araklidas
• [asterisk-users] can no more compile zaptel !!!   Lee Archer
• [asterisk-users] Manager interface   Lee Archer
• [Asterisk-Users] Motorola and Asterisk   Julio Arruda
• [asterisk-users] Tadiran Coral IP PBX to Asterisk   Julio Arruda
• [asterisk-users] SRTP enabling   Julio Arruda
• [asterisk-users] Re: Re: Re: TE420P/TE415P?   Julio Arruda
• [asterisk-users] Invalid module format (ztdummy)   Pablo L. Arturi
• [asterisk-users] Invalid module format (ztdummy)   Pablo L. Arturi
• [asterisk-users] newbbie question   Pablo L. Arturi
• [asterisk-users] newbbie question   Pablo L. Arturi
• [asterisk-users] How to connect XLite with another public IP?   Pablo L. Arturi
• [asterisk-users] Strange behaviour Panasonic KX-TD1232   Pablo L. Arturi
• [asterisk-users] VoipNow 1.2.0 Beta   Pablo L. Arturi
• [asterisk-users] Cisco 7960 Softkey templates   Lacy Moore - Aspendora
• [asterisk-users] 2 Handsets, Same extension   Lacy Moore - Aspendora
• [Asterisk-Users] Asterisk 1.2.7.1 with Polycom 501 on SIP -> Conf Calling   Lacy Moore - Aspendora
• [asterisk-users] Re: Asterisk 1.2.10 and Zaptel 1.2.7 released!   Lacy Moore - Aspendora
• [asterisk-users] Polycom 501 provisioning : how to secure values located in plein text files   Lacy Moore - Aspendora
• [asterisk-users] Merlin Legend PRI not quite there yet   Lacy Moore - Aspendora
• [asterisk-users] Asterisk/GPL and G.729 licensing   Lacy Moore - Aspendora
• [asterisk-users] Polycom compatible phone for Asterisk   AstATN
• [asterisk-users] meetme application doubt   AstATN
• [asterisk-users] help for SPA-2100   AstATN
• [asterisk-users] RE: alcatel ip touch 4068 ... sip?   AstATN
• [asterisk-users] Manager interface   Asterisk
• [asterisk-users] Sipura ATA's Forwarding PSTN Calls to Asterisk   Jonathan Attwood
• [asterisk-users] zap and fax   Giedrius Augys
• [asterisk-users] mgcp trouble   Giedrius Augys
• [asterisk-users] FreePBX Inbound Route   Giedrius Augys
• [asterisk-users] FreePBX Inbound Route   Giedrius Augys
• [asterisk-users] freepbx and a2billing   Giedrius Augys
• [asterisk-users] Phones cutting out.....again - PLEASE HELP!!!   Dan Austin
• [asterisk-users] Invite someone to Conference   Dan Austin
• [asterisk-users] Connecting branch offices through IPsec tunnel --latency effects?   Dan Austin
• [asterisk-users] OFF-TRACK: Is VOIP -PSTN integration legal in China   Dan Austin
• [asterisk-users] OFF-TRACK: Is VOIP -PSTN integration legal inChina   Dan Austin
• [asterisk-users] playing a sound into a meetme conference   Simon Austin
• [asterisk-users] playing a sound into a meetme conf   Simon Austin
• [asterisk-users] playing a sound into a meetme conf   Simon Austin
• [asterisk-users] accessing dialplan global variables in agi   Simon Austin
• [asterisk-users] playing a sound into a meetme conf   Simon Austin
• [asterisk-users] accessing dialplan global variables in agi   Simon Austin
• [asterisk-users] accessing dialplan global variables in agi   Simon Austin
• [asterisk-users] accessing dialplan global variables in agi   Simon Austin
• [asterisk-users] Message waiting question...   Jean-Yves Avenard
• [asterisk-users] Message waiting question...   Jean-Yves Avenard
• [asterisk-users] Message waiting question...   Jean-Yves Avenard
• [asterisk-users] Message waiting question...   Jean-Yves Avenard
• [asterisk-users] Message waiting question...   Jean-Yves Avenard
• [asterisk-users] Message waiting question...   Jean-Yves Avenard
• [asterisk-users] Message waiting question...   Jean-Yves Avenard
• [Asterisk-Users] How to configure NOKIA N70 with Asterisk?   Jean-Yves Avenard
• [asterisk-users] ata hook-flash   Accursio Avona
• [asterisk-users] ata hook-flash   Accursio Avona
• [asterisk-users] Re: $3,000 server   Nick B.
• [asterisk-users] RE: $3,000 server   Nick B.
• [asterisk-users] Zap Problem   Emre BALCI
• [asterisk-users] Cisco Buddies   Michiel van Baak
• [asterisk-users] qozap w/ 1.2.9.1   Michiel van Baak
• [asterisk-users] Cisco Buddies   Michiel van Baak
• [Asterisk-Users] Do you need a licence to connect a Cisco hardphone to Asterisk ?   Michiel van Baak
• [Asterisk-Users] Do you need a licence to connect a Ciscohardphone to Asterisk ?   Michiel van Baak
• Ok "\; " works - let us modify asterisk.vim ; ) Re: [asterisk-users] [tip]semicolon trouble:   Michiel van Baak
• [asterisk-users] PHP AGI   Michiel van Baak
• [asterisk-users] Snom 300 headset with static noise   Michiel van Baak
• [asterisk-users] Cisco 7960 - automated send DTMF digits after dialing?   Michiel van Baak
• [asterisk-users] Queue announcement issues   Michiel van Baak
• [asterisk-users] asterisk 1.4 download   Michiel van Baak
• [asterisk-users] Agent penality for dynamic agents   Wes Baehr
• [Asterisk-Users] Now that Nufone is dead...   Wes Baehr
• [asterisk-users] Serveremail Setting Does Not Work for Text Messages   Wes Baehr
• [asterisk-users] Linksys SPA-3102   Wes Baehr
• [asterisk-users] Grand stream 2000 will not dial *xx   Chris Bagnall
• [asterisk-users] Contacts for Chan_gsm_bt maintainer?   Boris Bakchiev
• [asterisk-users] PHP AGI   Time Bandit
• [asterisk-users] PHP AGI   Time Bandit
• [asterisk-users] PHP AGI   Time Bandit
• [asterisk-users] Can one SIP extension be used for two phones?   Time Bandit
• [asterisk-users] Global variables and AGI   Time Bandit
• [asterisk-users] phpagi problem   Time Bandit
• [asterisk-users] Examples of handeling input from phones with PHP   Time Bandit
• [asterisk-users] NuFone suggests to use Vonage!!!!   Joe Baptista
• Re: [asterisk-users] Inc.com Names Mark Spencer of Digium to its “30 Under 30: America’s Coolest Young Entrepreneurs”   Reynaldo Baquerizo
• Re: [asterisk-users] Inc.com Names Mark Spencer of Digium to its “30 Under 30: America’s Coolest Young Entrepreneurs”   Reynaldo Baquerizo
• [asterisk-users] CHANNEL STATUS of sip and iax devices   Reynaldo Baquerizo
• [asterisk-users] CHANNEL STATUS of sip and iax devices   Reynaldo Baquerizo
• [asterisk-users] Unicall, not HOW but WHY   Barzilai
• [asterisk-users] Encrypting the Conversation   Mike Bates
• [Asterisk-Users] Nufone Tollfree Port   Ariel Batista
• [asterisk-users] zaptel errors   Ariel Batista
• [asterisk-users] Weird Hold Problem   Forrest Beck
• [asterisk-users] Voicemail Forwarding   Forrest Beck
• [Asterisk-Users] RE: Is there a search feature?   David Beckerdite
• [asterisk-users] Is dmtfmode used/valid in iax.conf contexts?   Peter Beckman
• [asterisk-users] Hitting # to Transfer out of a Queue   Nic Bellamy
• [asterisk-users] Hitting # to Transfer out of a Queue   Nic Bellamy
• [Asterisk-Users] Now that Nufone is dead...   Benchev
• [Asterisk-Users] Need help with config-files   Benchev
• [Asterisk-Users] Need help with config-files   Benchev
• [Asterisk-Users] Now that Nufone is dead...   Benchev
• [asterisk-users] Asterisk + fax   Benchev
• [asterisk-users] Asterisk Realtime Macros   Benchev
• [asterisk-users] sip realtime   Benchev
• [asterisk-users] sip realtime   Benchev
• [asterisk-users] 1000s of extensions in one context?   Dovid Bender
• [asterisk-users] Provider UNREACHABLE   Dovid Bender
• [asterisk-users] comcast info -- somewhat offtopic   Dovid Bender
• [asterisk-users] 1000s of extensions in one context?   Dovid Bender
• [asterisk-users] unhappy-about-VoIP-providers@googlegroups.comfounded   Dovid Bender
• [asterisk-users] 1000s of extensions in one context?   Dovid Bender
• [asterisk-users] Polycom 601 and Paging   Dovid Bender
• [asterisk-users] Polycom 601 and Paging   Dovid Bender
• [asterisk-users] Ztdummy   Dovid Bender
• [asterisk-users] Polycom Silent ring   Dovid Bender
• [asterisk-users] Polycom 601 and Paging   Dovid Bender
• [asterisk-users] asterisk extra sounds: what for?   Dovid Bender
• [asterisk-users] Operator in Voicemail   Dovid Bender
• [asterisk-users] G729 Softphone   Dovid Bender
• [asterisk-users] Intercom feature on Polycom phones   Dovid Bender
• [asterisk-users] Polycom_acd_functions SIP trouble   Dovid Bender
• [asterisk-users] Just bought a Polycom 501 - I feel like myGXP-2000 was better...   Dovid Bender
• [asterisk-users] Just bought a Polycom 501 - I feel likemyGXP-2000 was better...   Dovid Bender
• [asterisk-users] Just bought a Polycom 501 - Ifeellike myGXP-2000 was better...   Dovid Bender
• [asterisk-users] Snom 360   Dovid Bender
• [asterisk-users] SNOM 360   Dovid Bender
• [asterisk-users] SNOM 360   Dovid Bender
• [asterisk-users] SNOM 360   Dovid Bender
• [asterisk-users] Manager interface   Dovid Bender
• [asterisk-users] Error On brdging Call   Dovid Bender
• [asterisk-users] Play sounds to the callee and the callersynchronously when call begins   Dovid Bender
• [asterisk-users] Top Users in a MeetMe room???   Dovid Bender
• [asterisk-users] Strange error   Dovid Bender
• [asterisk-users] SIP and podcasts   Dovid Bender
• [asterisk-users] Error in ubuntu dapper   don Paolo Benvenuto
• [asterisk-users] Error in ubuntu dapper   don Paolo Benvenuto
• [asterisk-users] Error in ubuntu dapper   don Paolo Benvenuto
• [asterisk-users] IVR with LDAP query for phone number and mobile number??   BerkHolz, Steven
• [asterisk-users] Any Polycom dealers willing help out?   Eric Bishop
• [asterisk-users] Blended?   Eric Bishop
• [asterisk-users] Anyone out there using Junghanns ISDNguard?   Eric Bishop
• [asterisk-users] Anyone out there using Junghanns ISDNguard?   Eric Bishop
• [asterisk-users] Binary/unreadable configuration files?   Eric Bishop
• [asterisk-users] Nokia E61/E70 not always answering voip calls   Gareth Blades
• *****SPAM***** [asterisk-users] Dial plan question   Chris Blunt
• *****SPAM***** [asterisk-users] Load balenced (ADSL) network connections, is it possible?   Chris Blunt
• [asterisk-users] Polycom 1.6.7 Firmware Messages Button   Greg Boehnlein
• [asterisk-users] Polycom 1.6.7 Firmware Messages Button   Greg Boehnlein
• [asterisk-users] link quality is poor   Alexandr Bondar
• [asterisk-users] extensions.conf 4 digit dialing question   Jerry Bonner
• [asterisk-users] buy X100p card in singapore   Leo Ann Boon
• [asterisk-users] Encrypting the Conversation   Leo Ann Boon
• [asterisk-users] Missing close quote in CallerID breaks SIP. . .workaround?   Leo Ann Boon
• [asterisk-users] Force peer source ip   Leo Ann Boon
• [asterisk-users] FW: Conference   Leo Ann Boon
• [asterisk-users] alcatel ip touch 4068 ... sip?   Leo Ann Boon
• [asterisk-users] RE: alcatel ip touch 4068 ... sip?   Leo Ann Boon
• [asterisk-users] RE: alcatel ip touch 4068 ... sip?   Leo Ann Boon
• [asterisk-users] Recommend hard phone which supports IAX2?   Stephen Bosch
• [asterisk-users] Connecting branch offices through IPsec tunnel -- latency effects?   Stephen Bosch
• [asterisk-users] Recommend hard phone which supports IAX2?   Stephen Bosch
• [asterisk-users] Detecting voicemail from CO on FXO port and passing to H.323 phone. Possible?   Bob Bosiljevac
• [asterisk-users] Transfers - No ringback or moh   Nathan Bowyer
• [asterisk-users] International Dialing setup in extensions.conf   Peter Bowyer
• [asterisk-users] context   Peter Bowyer
• [asterisk-users] Problem with making outgoing calls   Peter Bowyer
• [Asterisk-Users] how to decrease answer time !   Peter Bowyer
• [asterisk-users] bugs.digium.com   Peter Bowyer
• [Asterisk-Users] How to configure NOKIA N70 with Asterisk?   Crazy Boy
• [asterisk-users] Unable to find good link to configure Polycom 501 with Asterisk (Plz send good link)   Crazy Boy
• [asterisk-users] Unable to configure my DID number   Crazy Boy
• [asterisk-users] Problem with making outgoing calls   Crazy Boy
• [asterisk-users] Problem with making outgoing calls   Crazy Boy
• [asterisk-users] Please suggest me Best VoIP Service Provider   Crazy Boy
• [asterisk-users] How to connect XLite with public IP?   Crazy Boy
• [asterisk-users] How to connect XLite with public IP?   Crazy Boy
• [asterisk-users] How to connect XLite with another public IP?   Crazy Boy
• [asterisk-users] Please suggest me Best VoIP Service Provider   Crazy Boy
• [asterisk-users] Please suggest me Best VoIP Service Provider   Crazy Boy
• [asterisk-users] Strange error   Crazy Boy
• [asterisk-users] SIP is not working sometimes. IAX is working fine. Why?   Crazy Boy
• [asterisk-users] Problem with call receiving (Asterisk+PSTN+Digium TDM04B)   Crazy Boy
• [Asterisk-Users] Zaptel 1.2.6 / Upgrade Problem   Jerry Brady
• [Asterisk-Users] Zaptel 1.2.6 / Upgrade Problem   Jerry Brady
• [asterisk-users] Again on ISDN - MSN in Italy   Peter Braidwood
• [asterisk-users] Voismart GSM - no billsecs   Matteo Brancaleoni
• [asterisk-users] long distance ethernet & Asterisk   Andy Brezinsky
• [asterisk-users] Install asterisk-bristuff for Debian Linux   Tijl Van den Broeck
• [asterisk-users] Feasability of using * for small appartment building?   Greg Broiles
• [asterisk-users] Encrypting the Conversation   Greg Broiles
• [asterisk-users] CallerID   Ryder Brook
• [asterisk-users] CallerID   Ryder Brook
• [asterisk-users] reboots itlself   Ryder Brook
• [asterisk-users] Queues and qview.pl script   Matt Brown
• [asterisk-users] Queues and qview.pl script [Solved]   Matt Brown
• [asterisk-users] SIP To: header   Tom Browning
• [asterisk-users] QuadBRI + TDM + GSM hangup problems   Paco Brufal
• [asterisk-users] SOLVED: QuadBRI + TDM + GSM hangup problems   Paco Brufal
• [asterisk-users] Asterisk stops abruptly   Dan Brummer
• [asterisk-users] Issues with making Transfers   Dan Brummer
• [asterisk-users] Issues with making Transfers   Dan Brummer
• [asterisk-users] Asterisk stops abruptly   Dan Brummer
• [asterisk-users] Asterisk stops abruptly   Dan Brummer
• [asterisk-users] Issues with making Transfers   Dan Brummer
• [asterisk-users] Issues with making Transfers   Dan Brummer
• [asterisk-users] Asterisk 1.2.7.1 Crashing   Dan Brummer
• [asterisk-users] Asterisk 1.2.7.1 Crashing   Dan Brummer
• [asterisk-users] Warm transfer issues in 1.2.10   Dan Brummer
• [asterisk-users] Warm transfer issues in 1.2.10   Dan Brummer
• [asterisk-users] Warm transfer issues in 1.2.10   Dan Brummer
• [asterisk-users] Redundant Ethernet   Dan Brummer
• [asterisk-users] header replacement   Russell Bryant
• [asterisk-users] Zaptel Compilation Error   Russell Bryant
• [asterisk-users] header replacement   Russell Bryant
• [asterisk-users] Writing own applications for asterisk - read CALLERIDNUM   Russell Bryant
• [asterisk-users] asterisk database   Russell Bryant
• [asterisk-users] Error in ubuntu dapper   Russell Bryant
• [asterisk-users] question about asterisk DB   Russell Bryant
• [asterisk-users] Asterisk Dial Plan to Play Message   Russell Bryant
• [asterisk-users] termcap support not found   Russell Bryant
• [asterisk-users] Reload of wct4xxp without restarting of Asterisk?   Russell Bryant
• [asterisk-users] Reload of wct4xxp without restarting of Asterisk?   Russell Bryant
• [asterisk-users] bugs.digium.com   Russell Bryant
• [asterisk-users] accessing dialplan global variables in agi   Russell Bryant
• [asterisk-users] Asterisk 1.4 Schedule and Features/Changes   Russell Bryant
• [asterisk-users] french promt   Russell Bryant
• [asterisk-users] How do you recompile individual source modules?   Russell Bryant
• [asterisk-users] accessing dialplan global variables in agi   Russell Bryant
• [asterisk-users] AEL2 Looping   Russell Bryant
• [Asterisk-Users] Muffled Audio   Shaun Bryant
• [asterisk-users] Examples of handeling input from phones with PHP   Chuck Bunn
• [Asterisk-Users] Sample PRI and FXS channel bank zap files for zaptel and asterisk.   Shane Burrell
• [asterisk-users] IAX2 trunking problems   Pierre Burton
• [asterisk-users] Asterisk stops accepting calls   Laurent CARON
• [asterisk-users] VoiceMail Name Variable in Dial Plan   Damien Cahill
• [asterisk-users] Voicemail not sent via email   Damien Cahill
• [Asterisk-Users] Need help with Junghanns Quadbri   Tommaso Calosi
• [asterisk-users] Call transfer asterisk + with SPA-1001   Tommaso Calosi
• [asterisk-users] Cisco SIP Firmware   Francisco Gonzalez Canales
• [asterisk-users] Cisco 7940 dialplan.xml   Francisco Gonzalez Canales
• [asterisk-users] Cisco 7940 dialplan.xml   Francisco Gonzalez Canales
• [asterisk-users] MoH clicks and pops   Francisco Gonzalez Canales
• [Asterisk-Users] Best VoIP provider for Asterisk   Brian Capouch
• [asterisk-users] for you guys setting up customer offices...   Brian Capouch
• [asterisk-users] OT: Sipura SPA-3000 ATA Directing Calls to Asterisk   Brian Capouch
• [asterisk-users] OT: Sipura SPA-3000 ATA Directing Calls to Asterisk   Brian Capouch
• [asterisk-users] Re: for you guys setting up customer offices...   Brian Capouch
• [Asterisk-Users] app_conference DTMFs?   Brian Capouch
• [asterisk-users] Using dproxy to solve "no DNS hangs everything" problem?   Brian Capouch
• [asterisk-users] Using dproxy to solve "no DNS hangs everything"problem?   Brian Capouch
• [asterisk-users] Using dproxy to solve "no DNS hangs everything"problem?   Brian Capouch
• [asterisk-users] Please suggest me Best VoIP Service Provider   Brian Capouch
• [asterisk-users] Re: asterisk-users Digest, Vol 24, Issue 116   Brian Capouch
• [asterisk-users] Sipura ATA's Forwarding PSTN Calls to Asterisk   Brian Capouch
• [asterisk-users] Codec Negotiation   Brian Capouch
• [asterisk-users] Codec Negotiation   Brian Capouch
• [asterisk-users] Sipura ATA's Forwarding PSTN Calls to Asterisk   Brian Capouch
• [asterisk-users] Missing close quote in CallerID breaks SIP. . .workaround?   Brian Capouch
• [asterisk-users] RDNIS and IAX2   Brian Capouch
• [asterisk-users] OH323 registration with gatekeeper problem   Marcus Carlson
• [asterisk-users] OH323 registration with gatekeeper problem   Marcus Carlson
• [asterisk-users] Binary/unreadable configuration files?   Marcus Carlson
• [asterisk-users] Multiple dialing   Marcus Carlson
• [asterisk-users] Queue RoundRobin   Santiago del Castillo
• [asterisk-users] Dell PowerEdge 830   Cavanna, Richard
• [asterisk-users] RE: Just bought a Polycom 501 - I feel like myGXP-2000 was   Cavanna, Richard
• [asterisk-users] gxp-2000 configure line appearances   Cavanna, Richard
• [asterisk-users] Re: gxp-2000 configure line appearances   Cavanna, Richard
• [asterisk-users] Grand stream 2000 will not dial *xx   Cavanna, Richard
• [asterisk-users] Grand stream 2000 will not dial *xx   Cavanna, Richard
• [asterisk-users] Re: Grand stream 2000 will not dial *xx   Cavanna, Richard
• [asterisk-users] Aastra 9133i w/NAT and Asterisk   Frank Cernese
• [asterisk-users] Change current working directory to /tmp   Patrick Cervicek
• [asterisk-users] Asterisk and H.323   Cesc
• [asterisk-users] Asterisk and H.323   Cesc
• [asterisk-users] alcatel ip touch 4068 ... sip?   Cesc
• [asterisk-users] Problems with supervised transfer and agents   Ivan Cetta
• [asterisk-users] How to connect 2 AAH   Gidean Chan
• [asterisk-users] Asteriskguru switchboard   Gidean Chan
• [Asterisk-Users] SIP <--> H323 RTP Questions (1 WAY Audio only)   Ken Chan
• [Asterisk-Users] Now that Nufone is dead...   Carlos Chavez
• [asterisk-users] Asterisk H323 and Alcatel 4400   Carlos Chavez
• [asterisk-users] Problem with MFCR2   Carlos Chavez
• [asterisk-users] Problem with MFCR2   Carlos Chavez
• [asterisk-users] Just bought a Polycom 501 - I feel like my GXP-2000 was better...   Carlos Chavez
• [asterisk-users] create custom cdr's   Carlos Chavez
• [asterisk-users] Binary/unreadable configuration files?   Carlos Chavez
• [asterisk-users] Anyone tried vitelity?   Carlos Chavez
• [Asterisk-Users] Voicemail   Khaled Chehab
• [Asterisk-Users] Voicemail   Khaled Chehab
• [Asterisk-Users] directory   Khaled Chehab
• [Asterisk-Users] SRTP   Khaled Chehab
• [Asterisk-Users] FW: SRTP   Khaled Chehab
• [Asterisk-Users] trixbox 1.1 download   Khaled Chehab
• [asterisk-users] (no subject)   Khaled Chehab
• [asterisk-users] IVR DTMF   Khaled Chehab
• [asterisk-users] Urgent Upgrade   Khaled Chehab
• [asterisk-users] Voicemail   Khaled Chehab
• [asterisk-users] SRTP or zrtp   Khaled Chehab
• [asterisk-users] IVR DTMF   Khaled Chehab
• [asterisk-users] Urgent call forward   Khaled Chehab
• [asterisk-users] Urgent context   Khaled Chehab
• [asterisk-users] Urgent context   Khaled Chehab
• [asterisk-users] context   Khaled Chehab
• [asterisk-users] IVR DTMF   Khaled Chehab
• [asterisk-users] IVR DTMF   Khaled Chehab
• [asterisk-users] IVR DTMF   Khaled Chehab
• [asterisk-users] IVR DTMF   Khaled Chehab
• [asterisk-users] IP CDR   Khaled Chehab
• [asterisk-users] IVR DTMF   Khaled Chehab
• [asterisk-users] IVR DTMF   Khaled Chehab
• [asterisk-users] IP CDR   Khaled Chehab
• [asterisk-users] IP CDR   Khaled Chehab
• [asterisk-users] Conference   Khaled Chehab
• [asterisk-users] Conference   Khaled Chehab
• [asterisk-users] FW: Conference   Khaled Chehab
• [asterisk-users] Conference   Khaled Chehab
• [asterisk-users] Conference help   Khaled Chehab
• [asterisk-users] FW: Conference   Khaled Chehab
• [asterisk-users] FW: IP CDR   Khaled Chehab
• [asterisk-users] FW: Conference   Khaled Chehab
• [asterisk-users] french promt   Khaled Chehab
• [asterisk-users] CDR IP Authorization   Khaled Chehab
• [asterisk-users] CDR IP Authorization   Khaled Chehab
• [asterisk-users] Voice mail limit   Khaled Chehab
• [asterisk-users] SRTP   Khaled Chehab
• [asterisk-users] CDR IP Authorization   Khaled Chehab
• [asterisk-users] how to enable users on other iax server call my iax users   Bobber Cheng
• [asterisk-users] Asterisk and NEC NEAX 2000 IPS   Peter Childs
• [asterisk-users] Help with router setup on new asterisk box   Bob Chiodini
• [asterisk-users] New Asterisk server crashes daily   Bob Chiodini
• [asterisk-users] Cisco 7940 dialplan.xml   Bob Chiodini
• [asterisk-users] Re: Metermaid phone compatibility   Dr. Michael J. Chudobiak
• [asterisk-users] Sangoma Stops Receiving Calls   Dr. Michael J. Chudobiak
• Re: [Asterisk-Users] Running 40 active calls (too much för CPU?)   Simone Cittadini
• [asterisk-users] where the bottleneck lies ? (was: Server redundancy)   Simone Cittadini
• [asterisk-users] where the bottleneck lies ? (was: Serverredundancy)   Simone Cittadini
• [asterisk-users] Connect to 'agi://blablabla' failed: Operation now in progress   Simone Cittadini
• [asterisk-users] billed calls when cellullar phone is unreachable   Simone Cittadini
• [asterisk-users] asterisk sending connects when it shouldn't   Simone Cittadini
• [asterisk-users] Connect to 'agi://blablabla' failed: Operation now in progress   Simone Cittadini
• [asterisk-users] asterisk sending connects when it shouldn't (is there a q931-INFORMATION equivalent in IAX2 ?)   Simone Cittadini
• [asterisk-users] ooh323c - cdr   Simone Cittadini
• [asterisk-users] -- Going to extension s|1 because of immediate=yes, but immediate is 'no'   Simone Cittadini
• [asterisk-users] Asterisk 1.4 Schedule and Features/Changes   Max Clark
• [asterisk-users] IVR - Automatic Attendant database query   Mike Clark
• [asterisk-users] Lots of Asterisk child processes   Mike Clark
• [asterisk-users] Automating the registration process   Kevin Class
• [asterisk-users] Automating the registration process   Kevin Class
• [asterisk-users] Has anybody in here created their own softphones?   Kevin Class
• [asterisk-users] Has anybody in here created their own softphones?   Kevin Class
• [asterisk-users] Encrypting the Conversation   Henry J. Cobb
• [asterisk-users] Server Optimization and Load Balancing   Henry J. Cobb
• [Asterisk-Users] app_conference DTMFs?   Henry J. Cobb
• [Asterisk-Users] app_conference DTMFs?   Henry J. Cobb
• [Asterisk-Users] app_conference DTMFs?   Henry J. Cobb
• [Asterisk-Users] app_conference DTMFs?   Henry J. Cobb
• [asterisk-users] [oh323]FastStart/H245Tunnelling/H245inSetup   richard Coco
• [asterisk-users] SIP client with video???   richard Coco
• [asterisk-users] multiple calls   Gene Cohen
• [solved] sorry... ; ) Re: [Asterisk-Users] can't run "cat $filename" inside scripts with system()   Tzafrir Cohen
• [Asterisk-Users] WebPhone   Tzafrir Cohen
• [Asterisk-Users] WebPhone   Tzafrir Cohen
• [Asterisk-Users] WebPhone   Tzafrir Cohen
• [Asterisk-Users] Best GPL Gui?   Tzafrir Cohen
• [Asterisk-Users] WebPhone   Tzafrir Cohen
• [Asterisk-Users] TDM Installation error   Tzafrir Cohen
• [Asterisk-Users] The Asterisk console on a Dell D820 with Intel High Definition Audio.   Tzafrir Cohen
• [Asterisk-Users] The Asterisk console on a Dell D820 with Intel High Definition Audio.   Tzafrir Cohen
• [Asterisk-Users] SIP debug logging   Tzafrir Cohen
• [Asterisk-Users] Asterisk-Addons compile problem (cdr_addon_mysql.c)   Tzafrir Cohen
• [Asterisk-Users] Calling Extensions generates congestion when call answered   Tzafrir Cohen
• [Asterisk-Users] Zaptel 1.2.6 / Upgrade Problem   Tzafrir Cohen
• [Asterisk-Users] Zaptel 1.2.6 / Upgrade Problem   Tzafrir Cohen
• [Asterisk-Users] vserver (Debian) - no tty: howto use /usr/sbin/safe_asterisk with "-c" for color CLI?   Tzafrir Cohen
• [Asterisk-Users] Re: vserver with no /dev/tty* how to run "asterisk -c" for a colored CLI?   Tzafrir Cohen
• [Asterisk-Users] Need help with config-files   Tzafrir Cohen
• [asterisk-users] Possible Bug?   Tzafrir Cohen
• [asterisk-users] Tired of fax calls... :-/   Tzafrir Cohen
• [asterisk-users] Possible Bug?   Tzafrir Cohen
• [asterisk-users] mISDN configuration   Tzafrir Cohen
• [asterisk-users] intel vs amd motherboards   Tzafrir Cohen
• [asterisk-users] Best practices with Asterisk   Tzafrir Cohen
• [asterisk-users] Zap Channel not hanging up on Telco side   Tzafrir Cohen
• [asterisk-users] ztmonitor in numeric mode   Tzafrir Cohen
• [Asterisk-Users] Channel bank not work   Tzafrir Cohen
• [asterisk-users] intel vs amd motherboards   Tzafrir Cohen
• [asterisk-users] ztmonitor in numeric mode   Tzafrir Cohen
• [asterisk-users] prob with debian and chan_zap   Tzafrir Cohen
• [asterisk-users] prob with debian and chan_zap   Tzafrir Cohen
• [asterisk-users] Fonality vs TrixBox UI   Tzafrir Cohen
• [asterisk-users] [tip]semicolon trouble: System($(sleep 4; cp 1.call out)&) not working, but System($( sleep 4 && cp 1.call out)&) ; )   Tzafrir Cohen
• [asterisk-users] prob with debian and chan_zap   Tzafrir Cohen
• [asterisk-users] prob with debian and chan_zap   Tzafrir Cohen
• [asterisk-users] Uninstalling Asterisk? No make uninstall?   Tzafrir Cohen
• [asterisk-users] Uninstalling Asterisk? No make uninstall?   Tzafrir Cohen
• [asterisk-users] Uninstalling Asterisk? No make uninstall?   Tzafrir Cohen
• [Asterisk-Users] vserver (Debian) - no tty: howto use /usr/sbin/safe_asterisk with "-c" for color CLI?   Tzafrir Cohen
• [asterisk-users] Uninstalling Asterisk? No make uninstall?   Tzafrir Cohen
• [Asterisk-Users] vserver (Debian) - no tty: howto use /usr/sbin/safe_asterisk with "-c" for color CLI?   Tzafrir Cohen
• [Asterisk-Users] vserver (Debian) - no tty: howto use /usr/sbin/safe_asterisk with "-c" for color CLI?   Tzafrir Cohen
• [Asterisk-Users] vserver (Debian) - no tty: howto use /usr/sbin/safe_asterisk with "-c" for color CLI?   Tzafrir Cohen
• 'cpmv' instead of 'cp' Re: [asterisk-users] [tip]semicolon trouble: System($(sleep 4; cp 1.call out)&) not working, but System($( sleep 4 && cp 1.call out)&) ; )   Tzafrir Cohen
• 'cpmv' instead of 'cp' Re: [asterisk-users] [tip]semicolon trouble: System($(sleep 4; cp 1.call out)&) not working, but System($( sleep 4 && cp 1.call out)&) ; )   Tzafrir Cohen
• [asterisk-users] intel vs amd motherboards   Tzafrir Cohen
• [asterisk-users] intel vs amd motherboards   Tzafrir Cohen
• [asterisk-users] Urgent Upgrade   Tzafrir Cohen
• [asterisk-users] setting up an email to fax with asterisk   Tzafrir Cohen
• [asterisk-users] Memory leak res_perl   Tzafrir Cohen
• [asterisk-users] Mandriva 2006 Cooker RPM for Asterisk 1.2.9   Tzafrir Cohen
• [asterisk-users] outgoing call problem   Tzafrir Cohen
• [asterisk-users] Text priority labels not working for me   Tzafrir Cohen
• [asterisk-users] New Asterisk server crashes daily   Tzafrir Cohen
• [asterisk-users] PCMCIA card support   Tzafrir Cohen
• [asterisk-users] 1000s of extensions in one context?   Tzafrir Cohen
• [asterisk-users] 1000s of extensions in one context?   Tzafrir Cohen
• [asterisk-users] FW: Are FreePBX Extensions not being created in asterisk? & FOP question.   Tzafrir Cohen
• [asterisk-users] How do you harden an Asterisk install?   Tzafrir Cohen
• [asterisk-users] Asterisk instances on VPS   Tzafrir Cohen
• R: [asterisk-users] Called number on ISDN   Tzafrir Cohen
• [asterisk-users] Blog about asterisk and voip techology   Tzafrir Cohen
• [asterisk-users] zaptel on dual processor, How?   Tzafrir Cohen
• [asterisk-users] Regression testing dialplan changes   Tzafrir Cohen
• [asterisk-users] zaptel on dual processor, How?   Tzafrir Cohen
• [asterisk-users] Two security holes fixed in latest versions of Asterisk   Tzafrir Cohen
• [asterisk-users] Setting a threshold for asterisk to take ZAP line off hook ?   Tzafrir Cohen
• [asterisk-users] Re: CentOS 4.3 and Zaptel-1.2.7   Tzafrir Cohen
• [asterisk-users] Bad luck installing bristuff on multiple Linux'es. Any one got a good-luck story I can repeat?   Tzafrir Cohen
• [asterisk-users] Astribank?   Tzafrir Cohen
• [asterisk-users] Astribank?   Tzafrir Cohen
• [asterisk-users] Setting a threshold for asterisk to take ZAP line off hook ?   Tzafrir Cohen
• [asterisk-users] Bad luck installing bristuff on multiple Linux'es. Any one got a good-luck story I can repeat?   Tzafrir Cohen
• [asterisk-users] Unicall, not HOW but WHY   Tzafrir Cohen
• [asterisk-users] Automating the registration process   Tzafrir Cohen
• [asterisk-users] Ztdummy   Tzafrir Cohen
• [asterisk-users] Unicall, not HOW but WHY   Tzafrir Cohen
• [asterisk-users] X100P clone not working   Tzafrir Cohen
• [asterisk-users] X100P clone not working   Tzafrir Cohen
• [asterisk-users] Error in ubuntu dapper   Tzafrir Cohen
• [asterisk-users] Asterisk autoloading of card modules   Tzafrir Cohen
• [asterisk-users] Solution init.d scripts for CentOS 4.3   Tzafrir Cohen
• [asterisk-users] Install asterisk-bristuff for Debian Linux   Tzafrir Cohen
• [asterisk-users] Re: need a pointer regarding scripting asterisk   Tzafrir Cohen
• [asterisk-users] If you prefer to read this mail list as a forum ...   Tzafrir Cohen
• [asterisk-users] termcap support not found   Tzafrir Cohen
• [asterisk-users] reboots itlself   Tzafrir Cohen
• [asterisk-users] Asterisk/GPL and G.729 licensing   Tzafrir Cohen
• [asterisk-users] Binary/unreadable configuration files?   Tzafrir Cohen
• [asterisk-users] Binary/unreadable configuration files?   Tzafrir Cohen
• [asterisk-users] Reload of wct4xxp without restarting of Asterisk?   Tzafrir Cohen
• [asterisk-users] Asterisk 1.2.10 - Continually Restarting Logger   Tzafrir Cohen
• [asterisk-users] Zaptel trunk failed to compile   Tzafrir Cohen
• [asterisk-users] where to read stderr.out from an agi script   Tzafrir Cohen
• [asterisk-users] If you prefer to read this mail list as a forum ...   Tzafrir Cohen
• [asterisk-users] Trouble configuring TDM400P on Dell SC420   Tzafrir Cohen
• [asterisk-users] Voicemail not sent via email   Tzafrir Cohen
• [asterisk-users] TDM01B -1 FXO card not working.   Tzafrir Cohen
• [asterisk-users] Disable native bridge between two zap trunks   Tzafrir Cohen
• [Asterisk-Users] How to retrieve voicemail   Tzafrir Cohen
• [asterisk-users] Re: Re: Re: TE420P/TE415P?   Tzafrir Cohen
• [asterisk-users] Re: dropping calls in the middle of conversation   John D. Coleman
• [asterisk-users] Re: Grand stream 2000 will not dial *xx   John D. Coleman
• [Asterisk-Users] Putting a call recording into a mailbox   Dean Collins
• [asterisk-users] Feasability of using * for small appartment building?   Dean Collins
• [asterisk-users] Re: Feasability of using * for small appartmentbuilding?   Dean Collins
• [asterisk-users] Re: Feasability of using * for smallappartmentbuilding?   Dean Collins
• [asterisk-users] Re: Feasability of using * for smallappartmentbuilding?   Dean Collins
• [asterisk-users] RE: [Asterisk-video] Asterisk as an MCU   Dean Collins
• [asterisk-users] RE: [Asterisk-video] Asterisk as an MCU   Dean Collins
• [asterisk-users] Using DUNDi with TrixBox mini HOWTO   Dean Collins
• [asterisk-users]unhappy-about-VoIP-providers@googlegroups.comfounded   Dean Collins
• [asterisk-users] New York city Asterisk consultants   Dean Collins
• [asterisk-users] Just bought a Polycom 501 - I feellike myGXP-2000 was better...   Dean Collins
• [asterisk-users] FS: 2 x Asterisk X100M (red) daughterboard cards - brand new.   Dean Collins
• [asterisk-users] Mobile SIP Client   Dean Collins
• [asterisk-users] FXS adapters and Polycom phones   Jamin W. Collins
• [asterisk-users] an operational scenario   Jamin W. Collins
• [asterisk-users] PAP2 TUI Configuration Menu   Jamin W. Collins
• [asterisk-users] PAP2 TUI Configuration Menu   Jamin W. Collins
• [Asterisk-Users] flash button on asterisk + legacy pbx system   Michael Collins
• [asterisk-users] NOT logging Callerid/Call Data?   Michael Collins
• [asterisk-users] Setvar=var=val in sip.conf   Michael Collins
• [asterisk-users] emulating key system - pick up so and so on line1   Michael Collins
• [asterisk-users] ISDN Protocol   Michael Collins
• [asterisk-users] PRI vs "Digital Trunk"   Michael Collins
• [asterisk-users] Setvar=var=val in sip.conf   Joshua Colp
• [asterisk-users] Codec Negotiation   Joshua Colp
• [asterisk-users] Caller ID on Transfers   Joshua Colp
• [asterisk-users] Caller ID on Transfers   Joshua Colp
• [asterisk-users] Caller ID on Transfers   Joshua Colp
• [asterisk-users] Caller ID on Transfers   Joshua Colp
• [asterisk-users] Developing VoIP with Asterisk   Joshua Colp
• [Asterisk-Users] Which ATA to test T.38 ? What about Linksys 3102   Joshua Colp
• [asterisk-users] SIP is not working sometimes. IAX is working fine.Why?   Joshua Colp
• [asterisk-users] Message waiting question...   Joshua Colp
• [asterisk-users] Strange Error when calling   Joshua Colp
• [asterisk-users] Message waiting question...   Joshua Colp
• [asterisk-users] DTMF relay   Joshua Colp
• [asterisk-users] Message waiting question...   Joshua Colp
• [asterisk-users] SIP Woes   Joshua Colp
• [asterisk-users] SIP Woes   Joshua Colp
• [asterisk-users] Getting no Audio with G729   Joshua Colp
• [asterisk-users] One extension to ring on multiple outside lines   Joshua Colp
• [asterisk-users] Voicmail Question   Joshua Colp
• [asterisk-users] registration process   Joshua Colp
• [asterisk-users] Canreinvite   Joshua Colp
• [asterisk-users] Transfer call in SIP   Joshua Colp
• [asterisk-users] CDR IP Authorization   Joshua Colp
• [asterisk-users] PAP2T always busy on incoming calls with zaptel   Joshua Colp
• [asterisk-users] Multiple Outbound SIP Trunks   Joshua Colp
• [asterisk-users] One extension to ring on multiple outside lines   Joshua Colp
• [asterisk-users] CSTA support for asterisk   Joshua Colp
• [asterisk-users] stream file outputs only silence, even with asterisk example gsm file   Joshua Colp
• [asterisk-users] registration process   Joshua Colp
• [asterisk-users] SendText() & displaying text messages on a SIPhandset's screen   Joshua Colp
• [asterisk-users] SendText() & displaying text messages on aSIPhandset's screen   Joshua Colp
• [asterisk-users] SendText() & displaying text messages onaSIPhandset's screen   Joshua Colp
• [asterisk-users] CSTA support for asterisk   Joshua Colp
• [asterisk-users] registration process   Joshua Colp
• [asterisk-users] registration process   Joshua Colp
• [asterisk-users] MeetMe recordings in mp3 format.   Joshua Colp
• [asterisk-users] SIP channel problem   Joshua Colp
• [asterisk-users] Canreinvite and remotely registered devices   Joshua Colp
• [asterisk-users] DNS lookups failing for SIP register   Joshua Colp
• [asterisk-users] asterisk 1.4 download   Joshua Colp
• [asterisk-users] CDR IP Authorization   Joshua Colp
• [asterisk-users] Got SUBSCRIBE for extensions without hint   Joshua Colp
• [asterisk-users] music ring (CRBT)   Joshua Colp
• [asterisk-users] music ring (CRBT)   Joshua Colp
• [asterisk-users] RemoveQueueMember isn't working.   Joshua Colp
• [asterisk-users] Voicemail Getting Cut Off after 5 seconds   Josh Coltrane
• [asterisk-users] CallerID in UK on TalkTalk - different to BT?   Angus Comber
• [asterisk-users] CallerID in UK on TalkTalk - different to BT?   Angus Comber
• [asterisk-users] Weird E1 problem   Derek Conniffe
• [asterisk-users] Cisco 7960 SIP 8-3-0 getting "Got SIP response 400"   Tim Connolly
• [asterisk-users] Cisco 7960 SIP 8-3-0   Tim Connolly
• [asterisk-users] DTMF in QUEUES dont work   Tim Connolly
• [asterisk-users] Agent penality for dynamic agents   Andre Courchesne - Consultant
• [asterisk-users] Two phone numbers, one SIP provider   David Cook
• [asterisk-users] Looking for an asterisk guru   Sean Cook
• [asterisk-users] Looking for an asterisk guru   Sean Cook
• [asterisk-users] Asterisk with ISDN Fritz PCI card   Guy Corbaz
• [asterisk-users] Asterisk with ISDN Fritz PCI card   Guy Corbaz
• [asterisk-users] Asterisk with ISDN Fritz PCI card   Guy Corbaz
• [asterisk-users] timing sources   David Cornejo
• [asterisk-users] Ztdummy   David Cornejo
• [Asterisk-Users] HP Proliant server?   Luca Corti
• SV: [Asterisk-Users] HP Proliant server?   Luca Corti
• [asterisk-users] audio session start delay   Luca Corti
• [asterisk-users] audio session start delay   Luca Corti
• [Asterisk-Users] can't dial Scotland ...   Dave Cotton
• [asterisk-users] for you guys setting up customer offices...   Dave Cotton
• [asterisk-users] 2.6.18 Kernels   Dave Cotton
• [asterisk-users] OT: Sipura SPA-3000 ATA Directing Calls toAsterisk   Dave Cotton
• [asterisk-users] 2 NICs; Asterisk receives on eth1 and replies on eth0   Dave Cotton
• [Asterisk-Users] Very bad quality withAVMFritz!cardPCIandchan_capi   Dave Cotton
• [asterisk-users] call forwarding to mobile phone   Dave Cotton
• [asterisk-users] Sipura ATA's Forwarding PSTN Calls to Asterisk   Dave Cotton
• [asterisk-users] Sipura ATA's Forwarding PSTN Calls to Asterisk   Dave Cotton
• [asterisk-users] Source Directory of ASterisk   Dave Cotton
• [Asterisk-Users] H.264 and Asterik?   Jonathan k. Creasy
• [asterisk-users] Re: Polycom compatible phone for Asterisk   Jonathan k. Creasy
• [asterisk-users] Selective Router (route based on Caller-ID) configuration   Matthew Crocker
• [Asterisk-Users] DTMF and ivr systems   Doug Crompton
• [asterisk-users] Sipura ATA's Forwarding PSTN Calls to Asterisk   Doug Crompton
• [asterisk-users] Re: What's the story with X10*P FXO cards?   The Masked Cucumber
• [asterisk-users] Call information on blind transfers   Alistair Cunningham
• [asterisk-users] Canreinvite and remotely registered devices   Alistair Cunningham
• [asterisk-users] DNS lookups failing for SIP register   Alistair Cunningham
• [asterisk-users] Canreinvite and remotely registered devices   Alistair Cunningham
• [asterisk-users] Canreinvite and remotely registered devices   Alistair Cunningham
• [asterisk-users] DNS lookups failing for SIP register   Alistair Cunningham
• [asterisk-users] hardware question (X100P X101P clones)   Alexandre DELAY
• [Asterisk-Users] isdn-data over iax   DRi at b-w-computer.de
• [asterisk-users] asterisk 1.2.9.1 and spandsp and rxfax   DRi at b-w-computer.de
• [asterisk-users] RE: Asterisk in Seattle   Dal
• [asterisk-users] Looking for an asterisk guru   Aaron Daniel
• [asterisk-users] Looking for an asterisk guru   Aaron Daniel
• SV: [asterisk-users] Cisco 7941/7961/7971 wont register with asterisk   Aaron Daniel
• [asterisk-users] Cisco SIP Firmware   Aaron Daniel
• [asterisk-users] Text priority labels not working for me   Aaron Daniel
• [asterisk-users] Server redundancy   Aaron Daniel
• [asterisk-users] Server redundancy   Aaron Daniel
• [asterisk-users] Testers for ISDN AOC (Advice of Charge (Gespraechsgebuehren)) needed   Klaus Darilion
• [asterisk-users] Testers for ISDN AOC (Advice of Charge (Gespraechsgebuehren)) needed   Klaus Darilion
• [asterisk-users] Testers for ISDN AOC (Advice of Charge (Gespraechsgebuehren)) needed   Klaus Darilion
• [asterisk-users] X100P clone not working   Frank Darner
• [asterisk-users] X100P clone not working   Frank Darner
• [asterisk-users] X100P clone not working   Frank Darner
• [asterisk-users] X100P clone not working   Frank Darner
• [asterisk-users] X100P clone not working   Frank Darner
• [asterisk-users] X100P clone not working   Frank Darner
• [asterisk-users] X100P clone not working   Frank Darner
• [asterisk-users] X100P clone not working   Frank Darner
• [asterisk-users] X100P clone not working   Frank Darner
• [Asterisk-Users] Dial Macro timeout fails   David
• [Asterisk-Users] Dial Macro timeout fails   David
• [Asterisk-Users] Aastra phones - disable call waiting   Steve Davies
• [Asterisk-Users] Aastra phones - disable call waiting   Steve Davies
• [asterisk-users] spa941 and sip "bye"   Steve Davies
• [asterisk-users] Certain fax types cause problems   Steve Davies
• [asterisk-users] Certain fax types cause problems   Steve Davies
• [asterisk-users] Certain fax types cause problems   Steve Davies
• [asterisk-users] Having trouble to receive fax from samsung sf3200   Steve Davies
• [asterisk-users] Asterisk + fax   Steve Davies
• [Asterisk-Users] Aastra phones - disable call waiting   Steve Davies
• [asterisk-users] priority problem   Steve Davies
• [asterisk-users] priority problem   Steve Davies
• [asterisk-users] priority problem   Steve Davies
• [asterisk-users] priority problem   Steve Davies
• [asterisk-users] Warm transfer issues in 1.2.10   Steve Davies
• [asterisk-users] Warm transfer issues in 1.2.10   Steve Davies
• [asterisk-users] Extension planning   Steve Davies
• [asterisk-users] Nokia E61/E70 not always answering voip calls   Steve Davies
• [asterisk-users] Sip phone settings set when user registers   Steve Davies
• [asterisk-users] SNOM 360   Steve Davies
• [asterisk-users] SNOM 360   Steve Davies
• [asterisk-users] SNOM 360   Steve Davies
• [asterisk-users] Voicemails randomly not deleting in 1.2.9.1 ??   Anthony Davis
• [asterisk-users] Operator in Voicemail   Anthony Davis
• [asterisk-users] Echo cancellation doesn't work after inbound calls are transferred to another extension   Jared Davison
• [Asterisk-Users] attended transfer issue   Mike Dawson
• [Asterisk-Users] attended transfer issue   Mike Dawson
• [asterisk-users] Transfers - No ringback or moh   Mike Dawson
• [asterisk-users] Serveremail Setting Does Not Work for Text Messages   Robert DeVries
• [asterisk-users] Queue RoundRobin   Delca
• [asterisk-users] Queue RoundRobin   Delca
• [asterisk-users] Queue RoundRobin   Delca
• [asterisk-users] Can't get blind transfer to work   Delca
• [asterisk-users] Can't get blind transfer to work   Delca
• [asterisk-users] Overriding # at the end   Delca
• [asterisk-users] Re: Overriding # at the end   Delca
• [asterisk-users] Re: Overriding # at the end   Delca
• [asterisk-users] DTMF Dial Tone   Delca
• [asterisk-users] What's the story with X10*P FXO cards?   Vincent Delporte
• [asterisk-users] Re: What's the story with X10*P FXO cards?   Vincent Delporte
• [Asterisk-Users] Sangoma A200 hangup detection   Mike Dent
• [Asterisk-Users] trixbox 1.1 download   Mike Dent
• [asterisk-users] Sangoma A200 and hangup detection with Asterisk.   Mike Dent
• [asterisk-users] Cisco Buddies   Mike Dent
• [asterisk-users] Hangup detection with Sangoma A200 in the UK?   Mike Dent
• [asterisk-users] sip, dbsecret, and dundi   Tod Detre
• [asterisk-users] Automatic Hangup problem on IAX2 communication to Asterisk   Rajiv Dhir
• [asterisk-users] What is ZapRas used for ?   Angel Diaz
• [asterisk-users] MWI from Octel to Asterisk   Mike Diehl
• [asterisk-users] MWI from Octel to Asterisk   Mike Diehl
• [asterisk-users] B2BUA Webbased and Click 2 dial apps   Dinesh
• [asterisk-users] B2BUA Webbased and Click 2 dial apps   Dinesh
• [asterisk-users] B2BUA Webbased and Click 2 dial apps   Dinesh
• [asterisk-users] Problems after upgrade asterisk   Iuri Gomes Diniz
• [asterisk-users] Fritz!Box Fon ATA   Manuel Dominguez
• [asterisk-users] Re: Fritz!Box Fon ATA   Manuel Dominguez
• [asterisk-users] Re: Fritz!Box Fon ATA   Manuel Dominguez
• [asterisk-users] Re: Fritz!Box Fon ATA   Manuel Dominguez
• [asterisk-users] intel vs amd motherboards   Don
• [asterisk-users] intel vs amd motherboards   Don
• [asterisk-users] Re: $3,000 server   Don
• [asterisk-users] Hammer Tone   Don
• [asterisk-users] Macro call uniqueid   Don
• [asterisk-users] Simple But important question (for me)   Don
• [asterisk-users] re:Simple But important question (for me)   Don
• [asterisk-users] create custom cdr's   Don
• [asterisk-users] accessing dialplan global variables in agi   Don
• [asterisk-users] accessing dialplan global variables in agi   Don
• [asterisk-users] VoipNow 1.2.0 Beta   Don
• [asterisk-users] Attempting native bridge   Vincenzo VD. Di Donna
• [asterisk-users] Disable native bridge between two zap trunks   Vincenzo VD. Di Donna
• [asterisk-users] Disable native bridge between two zap trunks   Vincenzo VD. Di Donna
• [asterisk-users] OT: Sipura SPA-3000 ATA Directing Calls to Asterisk   Michael Van Donselaar
• [asterisk-users] vegastream 50 FXO DTMF Problem   Peter Doyle
• [asterisk-users] vegastream 50 FXO DTMF Problem   Peter Doyle
• [asterisk-users] How to do load balancing (1:1) with IAX and two different ISPs   Ken Dresdell
• [asterisk-users] How to do load balancing (1:1) with IAX and twodifferent ISPs   Ken Dresdell
• [asterisk-users] asterisk+ooh323.. one way audio issue   Joseph Dudash
• [asterisk-users] 7970 SIP configs   Paul Duffy
• [asterisk-users] Fast busy after one digit dialled? - 7970 SIP 8.0.3   Paul Duffy
• [asterisk-users] Polycom phone cycles between UNREACHABLE and REACHABLE   Rana Dutt
• [asterisk-users] Simple But important question (for me)   Camilo Echeverry
• [asterisk-users] news   Kris Edwards
• [asterisk-users] packet8 dta 310 power supply question   Scott Edwards
• [asterisk-users] AGI tutorials   Steve Edwards
• [asterisk-users] accessing dialplan global variables in agi   Steve Edwards
• [asterisk-users] Dialing timeouts   Dan Elder
• [asterisk-users] RE: asterisk-users Digest, Vol 24, Issue 86   Dan Elder
• [asterisk-users] Goldmine CRM softphone + asterisk   Dan Elder
• [asterisk-users] Goldmine CRM softphone + asterisk   Dan Elder
• [asterisk-users] RE: dropping calls in the middle of conversation   Dan Elder
• [asterisk-users] Goldmine SIP client/softphone questions continued:   Dan Elder
• [asterisk-users] SIP phone w/ 'modem/data' port?   Dan Elder
• [asterisk-users] Re: Goldmine SIP client/softphone questions continued: (Dan Elder)   Dan Elder
• [asterisk-users] Goldmine sip client revisited   Dan Elder
• [asterisk-users] Extension planning   Nik Engel
• [asterisk-users] Extension planning   Nik Engel
• [asterisk-users] Sip phone settings according to logged in user   Nik Engel
• [asterisk-users] Sip phone settings set when user registers   Nik Engel
• [asterisk-users] Sip phone settings set when user registers   Nik Engel
• [asterisk-users] Force peer source ip   Robin Ericsson
• [asterisk-users] Re: Load balenced (ADSL) network connections, is it possible?   Erik
• [asterisk-users] G729 License to Bridge calls through VOIP provider?   Erik
• [asterisk-users] Realtime, ODBC Voicemail, and multiple asterisk servers?   Damon Estep
• [asterisk-users] not reaching at the destination number I dialed   Innocent Evil
• [asterisk-users] not reaching at the destination number I dialed   Innocent Evil
• [Asterisk-Users] Call waiting using free PBX   Dumpolid Exeplish
• [Asterisk-Users] Best GPL Gui?   C F
• [Asterisk-Users] BLINDTRANSFER   C F
• [Asterisk-Users] How to continue after a match in an include   C F
• [Asterisk-Users] channel shows to be in use   C F
• [Asterisk-Users] s / i extension difficulty   C F
• [Asterisk-Users] flash button on asterisk + legacy pbx system   C F
• [Asterisk-Users] Best GPL Gui?   C F
• [Asterisk-Users] BLINDTRANSFER   C F
• [Asterisk-Users] flash button on asterisk + legacy pbx system   C F
• [Asterisk-Users] flash button on asterisk + legacy pbx system   C F
• [Asterisk-Users] flash button on asterisk + legacy pbx system   C F
• [Asterisk-Users] time variable   C F
• [Asterisk-Users] H.264 and Asterik?   C F
• [asterisk-users] intel vs amd motherboards   C F
• [asterisk-users] for you guys setting up customer offices...   C F
• [asterisk-users] intel vs amd motherboards   C F
• [asterisk-users] intel vs amd motherboards   C F
• [asterisk-users] intel vs amd motherboards   C F
• [asterisk-users] intel vs amd motherboards   C F
• [asterisk-users] intel vs amd motherboards   C F
• [Asterisk-Users] flash button on asterisk + legacy pbx system   C F
• [asterisk-users] Intercom mode on Polycom and/or SPA9xx   C F
• [asterisk-users] Intercom mode on Polycom and/or SPA9xx   C F
• [asterisk-users] NuFone, please send the log file   C F
• [asterisk-users] NuFone, please send the log file   C F
• [Asterisk-Users] Re: TE420P/TE415P?   C F
• [asterisk-users] NuFone, please send the log file   C F
• [asterisk-users] Intercom mode on Polycom and/or SPA9xx   C F
• [asterisk-users] Intercom mode on Polycom and/or SPA9xx   C F
• [asterisk-users] Re: Re: TE420P/TE415P?   C F
• [asterisk-users] Re: Re: Re: TE420P/TE415P?   C F
• [asterisk-users] OT: Skype protocol cracked?   C F
• [asterisk-users] sending flash using DTMF   C F
• [asterisk-users] Testing 911?   C F
• [asterisk-users] Using dproxy to solve "no DNS hangs everything"problem?   C F
• [asterisk-users] Astribank?   C F
• [asterisk-users] Using dproxy to solve "no DNS hangs everything"problem?   C F
• [asterisk-users] Astribank?   C F
• [asterisk-users] Simple But important question (for me)   C F
• [asterisk-users] Astribank?   C F
• [asterisk-users] RE: $3,000 server   C F
• [asterisk-users] Can't get blind transfer to work   C F
• [asterisk-users] Intercom feature on Polycom phones   C F
• [asterisk-users] Intercom feature on Polycom phones   C F
• [asterisk-users] Just bought a Polycom 501 - I feel like my GXP-2000 was better...   C F
• [asterisk-users] Just bought a Polycom 501 - I feel like myGXP-2000 was better...   C F
• [asterisk-users] Voicemail Forwarding   C F
• [asterisk-users] Caller ID on Transfers   C F
• [asterisk-users] Caller ID on Transfers   C F
• [asterisk-users] Caller ID on Transfers   C F
• [asterisk-users] Caller ID on Transfers   C F
• [asterisk-users] Cisco 7960 Call Waiting Beep   C F
• [asterisk-users] Polycom 501 provisioning : how to secure values located in plein text files   C F
• [asterisk-users] Strange behaviour Panasonic KX-TD1232   C F
• [Asterisk-Users] How to configure NOKIA N70 with Asterisk?   FaberK
• [Asterisk-Users] How to configure NOKIA N70 with Asterisk?   FaberK
• [Asterisk-Users] Help with IVR menu.   Fabio
• [asterisk-users] intel vs amd motherboards   Fabio
• [asterisk-users] for you guys setting up customer offices...   Fabio
• [asterisk-users] audio session start delay   Fabio
• [asterisk-users] intel vs amd motherboards   Fabio
• [asterisk-users] IVR - Automatic Attendant database query   Fabio
• [asterisk-users] intel vs amd motherboards   Fabio
• [asterisk-users] intel vs amd motherboards   Fabio
• [asterisk-users] Manager action "hold" missing?   Wolfgang Falk
• [asterisk-users] Dial plan question   Jon Farmer
• [asterisk-users] Call queue drops call after 1 min   Jon Farmer
• [asterisk-users] Sony Ericsson F250m, Sipura 3000 and Asterisk   Jon Farmer
• [asterisk-users] Sony Ericsson F250m, Sipura 3000 and Asterisk   Jon Farmer
• [asterisk-users] Provider UNREACHABLE   Barry Fawthrop
• [asterisk-users] Rate or rank ITSP   Barry Fawthrop
• [asterisk-users] Provider UNREACHABLE   Barry Fawthrop
• [asterisk-users] Provider UNREACHABLE   Barry Fawthrop
• [asterisk-users] New Asterisk GUI   Barry Fawthrop
• [asterisk-users] Writing own applications for asterisk - read CALLERIDNUM   Matthias Fechner
• [asterisk-users] Writing own applications for asterisk - read CALLERIDNUM   Matthias Fechner
• [asterisk-users] Germany VOIP provider   Matthias Fechner
• [asterisk-users] gxp-2000 configure line appearances   Matthias Fechner
• [asterisk-users] voice format changed to 4   Ken Fegb
• [asterisk-users] Connecting branch offices through IPsec tunnel -- latency effects?   Manrique Feoli
• [asterisk-users] transfers from an E1 using 2b-channel or similar anyone? (QSIG?)   Manrique Feoli
• [asterisk-users] transfers from an E1 using 2b-channel or similar anyone? (QSIG?)   Manrique Feoli
• [asterisk-users] How to send a signal via E1/T1 ISDN to asterisk, to ask the call to be moved.   Manrique Feoli
• [asterisk-users] Dial exited non-zero, only if PSTN/ZAP/E1 hangs up first. not if voip hangs up.   Manrique Feoli
• [asterisk-users] Dial exited non-zero, only if PSTN/ZAP/E1 hangs up first. not if voip hangs up.   Manrique Feoli
• [asterisk-users] long distance ethernet & Asterisk   Manrique Feoli
• [asterisk-users] long distance ethernet & Asterisk   Manrique Feoli
• [asterisk-users] Trunk transferring?   Manrique Feoli
• [asterisk-users] Trunk transferring?   Manrique Feoli
• [asterisk-users] can't retake call after dialing through Zap/E1 wich doesn't answer   Manrique Feoli
• [asterisk-users] can't retake call after dialing through Zap/E1 wich doesn't answer   Manrique Feoli
• [Asterisk-Users] to.gsm and the.gsm   Robert La Ferla
• [Asterisk-Users] how to ask for number to dial and then dial it?   Robert La Ferla
• [Asterisk-Users] Latest SVN of asterisk-addons doesn't compile   Robert La Ferla
• [asterisk-users] No ringing on outgoing SIP calls.   Robert La Ferla
• [asterisk-users] an operational scenario   Bruce Ferrell
• [asterisk-users] an operational scenario   Bruce Ferrell
• [asterisk-users] Asterisk and Vigortalk problem   Roberto Fichera
• [asterisk-users] Cisco 2610 RTP port forwarding   Roberto Fichera
• [asterisk-users] I need help patching source   Bart Fisher
• [asterisk-users] Urgent source code changes needed   Bart Fisher
• [asterisk-users] How do you recompile individual source modules?   Bart Fisher
• [asterisk-users] How do you recompile individual source modules?   Bart Fisher
• [asterisk-users] IAX over two T1 connections bad quality   Bart Fisher
• [asterisk-users] New mailing list: asterisk-speech-rec   Kevin P. Fleming
• [Asterisk-Users] multiple authentication realms   Kevin P. Fleming
• [Asterisk-Users] TE420P/TE415P?   Kevin P. Fleming
• [Asterisk-Users] Re: TE420P/TE415P?   Kevin P. Fleming
• [Asterisk-Users] TE420P/TE415P?   Kevin P. Fleming
• [Asterisk-Users] Playing sound before dialing   Kevin P. Fleming
• [Asterisk-Users] Re: TE420P/TE415P?   Kevin P. Fleming
• [Asterisk-Users] dtmfmode=inband but SDP also indicates rfc2833   Kevin P. Fleming
• [asterisk-users] NuFone, please send the log file   Kevin P. Fleming
• [Asterisk-Users] struggling with the "g" flag   Kevin P. Fleming
• [Asterisk-Users] M() option to Dial   Kevin P. Fleming
• Octasic for TDM2400P and TDM400P? was: [Asterisk-Users] TE420P/TE415P?   Kevin P. Fleming
• [Asterisk-Users] How to continue after a match in an include   Kevin P. Fleming
• [asterisk-users] Re: Re: Re: TE420P/TE415P?   Kevin P. Fleming
• [asterisk-users] Queue RoundRobin   Kevin P. Fleming
• [asterisk-users] Codec Negotiation   Kevin P. Fleming
• [asterisk-users] Alternative (?) ways to handle G.729 and annexb   Kevin P. Fleming
• [asterisk-users] Re: Re: Re: TE420P/TE415P?   Kevin P. Fleming
• [asterisk-users] Agent Attended Transfer Without DTMF   Kevin P. Fleming
• [Asterisk-Users] app_conference DTMFs?   Matt Florell
• [Asterisk-Users] app_conference DTMFs?   Matt Florell
• [Asterisk-Users] app_conference DTMFs?   Matt Florell
• [Asterisk-Users] app_conference DTMFs?   Matt Florell
• [asterisk-users] IAX2 vs TDMoE   Matt Florell
• [asterisk-users] PRI dropouts   Matt Florell
• [asterisk-users] Vicidial + Unicall mfcr2   Matt Florell
• [asterisk-users] Passing Variables with IAX   Matt Florell
• [asterisk-users] Re: Re: Re: TE420P/TE415P?   Matt Florell
• [asterisk-users] Typical Asterisk Company   Matt Florell
• [asterisk-users] Typical Asterisk Company   Matt Florell
• [asterisk-users] Typical Asterisk Company   Matt Florell
• [asterisk-users] Typical Asterisk Company   Matt Florell
• [asterisk-users] Typical Asterisk Company   Matt Florell
• [asterisk-users] where to read stderr.out from an agi script   Matt Florell
• [asterisk-users] Re: Re: Re: TE420P/TE415P?   Matt Florell
• [asterisk-users] Re: Re: Re: TE420P/TE415P?   Matt Florell
• [Asterisk-Users] Re: Digium Hardware Reliability   El Flynn
• [Asterisk-Users] how to ask for number to dial and then dial it?   El Flynn
• [Asterisk-Users] Sangoma A200 woes   El Flynn
• [asterisk-users] DTMF   El Flynn
• [asterisk-users] menu system - configurator   El Flynn
• [asterisk-users] AGI tutorials   El Flynn
• [asterisk-users] DTMF detection and Sangoma cards   El Flynn
• [asterisk-users] SIP configuration by group   El Flynn
• [asterisk-users] Keep Zap Channel from answering   El Flynn
• [asterisk-users] Sangoma Stops Receiving Calls   El Flynn
• [asterisk-users] New Asterisk GUI   El Flynn
• [asterisk-users] New Asterisk GUI   El Flynn
• [asterisk-users] New Asterisk GUI   El Flynn
• [asterisk-users] mISDN configuration   Alessio Focardi
• [Asterisk-Users] setting cdr userfield in .call file   Cory Forsyth
• [Asterisk-Users] how to ask for number to dial and then dial it?   Cory Forsyth
• [asterisk-users] problems with IAX, extension recognition and Asterisk 1.2.9.1   Cory Forsyth
• [asterisk-users] asterisk cdr shows "FAILED"   Cory Forsyth
• [Asterisk-Users] How to configure NOKIA N70 with Asterisk?   Lele Forzani
• [asterisk-users] rxfax Got hangup   Jan Fousek
• [asterisk-users] Error: Dropping incompatible voice frame   Jan Fousek
• [asterisk-users] Asterisk fails to register, when the full logging is turned on   Jan Fousek
• [asterisk-users] transfers from an E1 using 2b-channel or similar anyone? (QSIG?)   Matthew Fredrickson
• [asterisk-users] Dlink DVG 1120S/Asterisk VoIP to PSTN   David Freeman
• [asterisk-users] Trunk transferring?   David Freeman
• [asterisk-users] Trunk transferring?   David Freeman
• [Asterisk-Users] Polycom Soundpoint IP 301 w/ MGCP   Jim Freeze
• [Asterisk-Users] H.264 and Asterik?   Chaim Fried
• [asterisk-users] is ooh323 RAS/ASN.1 broken?   Chaim Fried
• [asterisk-users] Polycom_acd_functions SIP trouble   James Fromm
• [asterisk-users] Polycom_acd_functions SIP trouble   James Fromm
• [asterisk-users] Polycom_acd_functions SIP trouble   James Fromm
• [Asterisk-Users] performance & reliabulity of asterisk voicemail using odbc storage   Dmitry Furmanov
• [Asterisk-Users] performance & reliabulity of asterisk voicemail using odbc storage   Dmitry Furmanov
• [Asterisk-Users] performance & reliabulity of asterisk voicemail using odbc storage   Dmitry Furmanov
• [Asterisk-Users] performance & reliabulity of asterisk voicemail using odbc storage   Dmitry Furmanov
• [asterisk-users] Connecting branch offices through IPsec tunnel --latency effects?   Brandon Galbraith
• [asterisk-users] Message waiting question...   Brandon Galbraith
• [asterisk-users] long distance ethernet & Asterisk   Brandon Galbraith
• [asterisk-users] IAX over two T1 connections bad quality   Brandon Galbraith
• [asterisk-users] sip phone networking question [possibly OT]   Brandon Galbraith
• [asterisk-users] outgoing call problem   Ganbaa
• [asterisk-users] outgoing call problem   Ganbaa
• [asterisk-users] outgoing call problem   Ganbaa
• [Asterisk-Users] IVR menus on different DIDs   Christian Gansberger
• [Asterisk-Users] IVR menus on different DIDs   Christian Gansberger
• [Asterisk-Users] IVR menus on different DIDs   Christian Gansberger
• [Asterisk-Users] IVR menus on different DIDs   Christian Gansberger
• [Asterisk-Users] Asterisk -x option in 1.2.9.1   Douglas Garstang
• [asterisk-users] 'sip debug'   Douglas Garstang
• [asterisk-users] 'sip debug'   Douglas Garstang
• [asterisk-users] 'sip debug'   Douglas Garstang
• [asterisk-users] 'sip debug'   Douglas Garstang
• [asterisk-users] Asterisk in Seattle   Douglas Garstang
• [asterisk-users] Asterisk in Seattle   Douglas Garstang
• [asterisk-users] Asterisk in Seattle   Douglas Garstang
• [asterisk-users] RE: Asterisk in Seattle   Douglas Garstang
• [asterisk-users] for you guys setting up customer offices...   Douglas Garstang
• [asterisk-users] OT: Sipura SPA-3000 ATA Directing Calls to Asterisk   Douglas Garstang
• [asterisk-users] OT: Sipura SPA-3000 ATA Directing Calls toAsterisk   Douglas Garstang
• [asterisk-users] Incoming Call matching to peer   Douglas Garstang
• [asterisk-users] Server redundancy   Douglas Garstang
• [asterisk-users] Server redundancy   Douglas Garstang
• [asterisk-users] 2 NICs; Asterisk receives on eth1 and replieson eth0   Douglas Garstang
• [asterisk-users] Server redundancy   Douglas Garstang
• [asterisk-users] Server redundancy   Douglas Garstang
• [asterisk-users] Server redundancy   Douglas Garstang
• [asterisk-users] where the bottleneck lies ? (was: Serverredundancy)   Douglas Garstang
• [asterisk-users] DUNDi 'Unable to Find Key'   Douglas Garstang
• [asterisk-users] Transfer ACCEPT followed by DECLINE   Douglas Garstang
• [asterisk-users] config files   Douglas Garstang
• [asterisk-users] Transferring out of Queues   Douglas Garstang
• [asterisk-users] config files   Douglas Garstang
• [asterisk-users] config files   Douglas Garstang
• [asterisk-users] Hitting # to Transfer out of a Queue   Douglas Garstang
• [asterisk-users] Hitting # to Transfer out of a Queue   Douglas Garstang
• [asterisk-users] Polycom config file location   Douglas Garstang
• [asterisk-users] MOH With Asterisk Controlled Transfers   Douglas Garstang
• [asterisk-users] Queue Transfers   Douglas Garstang
• [asterisk-users] Codec Negotiation   Douglas Garstang
• [asterisk-users] Setvar=var=val in sip.conf   Douglas Garstang
• [asterisk-users] Codec Negotiation   Douglas Garstang
• [asterisk-users] Setvar=var=val in sip.conf   Douglas Garstang
• [asterisk-users] Passing Variables with IAX   Douglas Garstang
• [asterisk-users] Passing Variables with IAX   Douglas Garstang
• [asterisk-users] Hitting # to Transfer out of a Queue   Douglas Garstang
• [asterisk-users] Unable to find extension in context ''   Douglas Garstang
• [asterisk-users] Hitting # to Transfer out of a Queue   Douglas Garstang
• [asterisk-users] Hitting # to Transfer out of a Queue   Douglas Garstang
• [asterisk-users] Hitting # to Transfer out of a Queue   Douglas Garstang
• [asterisk-users] Hitting # to Transfer out of a Queue   Douglas Garstang
• [asterisk-users] Reload clears queue stats   Douglas Garstang
• [asterisk-users] Polycom config file location   Douglas Garstang
• [asterisk-users] Hitting # to Transfer out of a Queue   Douglas Garstang
• [asterisk-users] Hitting # to Transfer out of a Queue   Douglas Garstang
• [asterisk-users] Hitting # to Transfer out of a Queue   Douglas Garstang
• [asterisk-users] emulating key system - pick up so and so on line 1   Douglas Garstang
• [asterisk-users] Hitting # to Transfer out of a Queue   Douglas Garstang
• [asterisk-users] Reload clears queue stats   Douglas Garstang
• [asterisk-users] Stuck ACD Agents   Douglas Garstang
• [asterisk-users] Queue Stats   Douglas Garstang
• [asterisk-users] Redundant Ethernet   Douglas Garstang
• [asterisk-users] Codec Negotiation   Douglas Garstang
• [asterisk-users] Codec Negotiation   Douglas Garstang
• [asterisk-users] Queue Stats   Douglas Garstang
• [asterisk-users] Codec Negotiation   Douglas Garstang
• [asterisk-users] Typical Asterisk Company   Douglas Garstang
• [asterisk-users] Typical Asterisk Company   Douglas Garstang
• [asterisk-users] Typical Asterisk Company   Douglas Garstang
• [asterisk-users] Typical Asterisk Company   Douglas Garstang
• [asterisk-users] Typical Asterisk Company   Douglas Garstang
• [asterisk-users] asterisk database   Douglas Garstang
• [asterisk-users] Typical Asterisk Company   Douglas Garstang
• [asterisk-users] Typical Asterisk Company   Douglas Garstang
• [asterisk-users] Queue Persistence with queue.log   Douglas Garstang
• [asterisk-users] Sipura ATA's Forwarding PSTN Calls to Asterisk   Douglas Garstang
• [asterisk-users] Typical Asterisk Company   Douglas Garstang
• [asterisk-users] Sipura ATA's Forwarding PSTN Calls to Asterisk   Douglas Garstang
• [asterisk-users] Codec Negotiation   Douglas Garstang
• [asterisk-users] How to connect 2 AAH   Douglas Garstang
• [asterisk-users] Queue Persistence with queue.log   Douglas Garstang
• [asterisk-users] Typical Asterisk Company   Douglas Garstang
• [asterisk-users] Sipura ATA's Forwarding PSTN Calls to Asterisk   Douglas Garstang
• [asterisk-users] Codec Negotiation   Douglas Garstang
• [asterisk-users] Codec Negotiation   Douglas Garstang
• [asterisk-users] Sipura ATA's Forwarding PSTN Calls to Asterisk   Douglas Garstang
• [asterisk-users] Sipura ATA's Forwarding PSTN Calls to Asterisk   Douglas Garstang
• [asterisk-users] Codec Negotiation   Douglas Garstang
• [asterisk-users] Codec Negotiation   Douglas Garstang
• [asterisk-users] Transfers - No ringback or moh   Douglas Garstang
• [asterisk-users] Sipura ATA's Forwarding PSTN Calls to Asterisk   Douglas Garstang
• [asterisk-users] Transfers - No ringback or moh   Douglas Garstang
• [asterisk-users] Transfers - No ringback or moh   Douglas Garstang
• [asterisk-users] Transfers - No ringback or moh   Douglas Garstang
• [asterisk-users] Transfers - No ringback or moh   Douglas Garstang
• [asterisk-users] Transfers - No ringback or moh   Douglas Garstang
• [asterisk-users] Transfers - No ringback or moh   Douglas Garstang
• [asterisk-users] Transfers - No ringback or moh   Douglas Garstang
• [asterisk-users] RDNIS and IAX2   Douglas Garstang
• [asterisk-users] Just bought a Polycom 501 - I feel likemyGXP-2000 was better...   Douglas Garstang
• [asterisk-users] RDNIS and IAX2   Douglas Garstang
• [asterisk-users] IAX2 Variables   Douglas Garstang
• [asterisk-users] Caller ID on Transfers   Douglas Garstang
• [asterisk-users] Caller ID on Transfers   Douglas Garstang
• [asterisk-users] Caller ID on Transfers   Douglas Garstang
• [asterisk-users] Caller ID on Transfers   Douglas Garstang
• [asterisk-users] RDNIS and IAX2   Douglas Garstang
• [asterisk-users] Caller ID on Transfers   Douglas Garstang
• [asterisk-users] Caller ID on Transfers   Douglas Garstang
• [asterisk-users] Caller ID on Transfers   Douglas Garstang
• [asterisk-users] Caller ID on Transfers   Douglas Garstang
• [asterisk-users] Caller ID on Transfers   Douglas Garstang
• [asterisk-users] Caller ID on Transfers   Douglas Garstang
• [asterisk-users] Caller ID on Transfers   Douglas Garstang
• [asterisk-users] Caller ID on Transfers   Douglas Garstang
• [asterisk-users] Caller ID on Transfers   Douglas Garstang
• [asterisk-users] Caller ID on Transfers   Douglas Garstang
• [asterisk-users] Just bought a Polycom 501 - I feel likemyGXP-2000was better...   Douglas Garstang
• [asterisk-users] Polycom 501 provisioning : how to secure valueslocated in plein text files   Douglas Garstang
• [asterisk-users] bugs.digium.com   Douglas Garstang
• [asterisk-users] bugs.digium.com   Douglas Garstang
• [asterisk-users] Re: bugs.digium.com   Douglas Garstang
• [asterisk-users] AEL2 Looping   Douglas Garstang
• [asterisk-users] Looking for carrier grade redundant solution   Douglas Garstang
• [asterisk-users] AEL2 Looping   Douglas Garstang
• [asterisk-users] Polycom 1.6.7 Firmware Messages Button   Douglas Garstang
• [Asterisk-Users] file.c: Unexpected control subclass '14'   Stefan-Michael. Guenther (in-put GbR)
• [asterisk-users] Dial command option D(digits)   Jerry Geis
• [asterisk-users] Option D in dial doesnt seem to be working   Jerry Geis
• [asterisk-users] ttp question getting connection timeout.   Jerry Geis
• [asterisk-users] quad T1 pri   Jerry Geis
• [asterisk-users] sms on wifi phones   Jerry Geis
• [asterisk-users] sdp multipart information nortel   Jerry Geis
• [asterisk-users] wip-300 question on audio dial out with tdm2402e   Jerry Geis
• [asterisk-users] sendtext or sip message - where in RFC   Jerry Geis
• [asterisk-users] Compiling zaptel on CentOS x86_64   Jerry Geis
• [asterisk-users] IAX over two T1 connections bad quality   Jerry Geis
• [asterisk-users] Tadiran Coral IP PBX to Asterisk   Bill Gibbs
• [asterisk-users] Tadiran Coral IP PBX to Asterisk   Bill Gibbs
• [asterisk-users] Choppy MOH (Cisco gateway)   Bill Gibbs
• [asterisk-users] Choppy MOH (Cisco gateway)   Bill Gibbs
• [asterisk-users] Choppy MOH (Cisco gateway)   Bill Gibbs
• [asterisk-users] Choppy MOH (Cisco gateway)   Bill Gibbs
• [asterisk-users] Choppy MOH (Cisco gateway)   Bill Gibbs
• [asterisk-users] Server Optimization and Load Balancing   Bill Gibbs
• [asterisk-users] Provider UNREACHABLE   Bill Gibbs
• [asterisk-users] Polycom compatible phone for Asterisk   Bill Gibbs
• [asterisk-users] emulating key system - pick up so and so on line 1   Bill Gibbs
• [asterisk-users] emulating key system - pick up so and so on line1   Bill Gibbs
• [asterisk-users] One way "screech" or tone   Bill Gibbs
• [asterisk-users] One way "screech" or tone   Bill Gibbs
• [asterisk-users] One way "screech" or tone   Bill Gibbs
• [Asterisk-Users] Howto: Gentoo + Hudlite + Scratch Asterisk Install   Matt Gibson
• [asterisk-users] NOT logging Callerid/Call Data?   Matt Gibson
• [asterisk-users] NOT logging Callerid/Call Data?   Matt Gibson
• [asterisk-users] Asterisk Jobs.com Updates   Matt Gibson
• [asterisk-users] zaptel on dual processor, How?   Dennis Gilmore
• [asterisk-users] asterisk and sip nat problems   Alexander Ginzburg
• [Asterisk-Users] WebPhone   Jean-Denis Girard
• [Asterisk-Users] WebPhone   Jean-Denis Girard
• [asterisk-users] [OT] Windows softphone with handset support?   Jean-Denis Girard
• [asterisk-users] G729 Softphone   Jean-Denis Girard
• [Asterisk-Users] Help with IVR menu.   Lior Goikhburg
• [Asterisk-Users] Help with IVR menu.   Lior Goikhburg
• [Asterisk-Users] ZAP channel for outbound calls.   Lior Goikhburg
• [asterisk-users] E1 connectivity question   Mohamed A. Gombolaty
• [asterisk-users] E1 connectivity question   Mohamed A. Gombolaty
• [asterisk-users] Strange Error when calling   Mohamed A. Gombolaty
• [asterisk-users] E1 connectivity question   Mohamed A. Gombolaty
• [asterisk-users] Strange Error when calling   Mohamed A. Gombolaty
• [asterisk-users] Strange Error when calling   Mohamed A. Gombolaty
• [asterisk-users] bugs.digium.com   Mohamed A. Gombolaty
• [asterisk-users] Strange Error when calling   Mohamed A. Gombolaty
• [asterisk-users] Queue Penalty   David Gomillion
• [asterisk-users] Asterisk instances on VPS   Gustavo Alejandro Gonzalez
• [asterisk-users] Called number on ISDN   Giordano Grandis
• R: [asterisk-users] Called number on ISDN   Giordano Grandis
• R: R: [asterisk-users] Called number on ISDN   Giordano Grandis
• [asterisk-users] TxFax   Giordano Grandis
• [asterisk-users] ZapRAS   Giordano Grandis
• [asterisk-users] Canreinvite   Giordano Grandis
• R: [asterisk-users] Canreinvite   Giordano Grandis
• [asterisk-users] 2 Handsets, Same extension   Michael Graves
• [asterisk-users] Re: What's the story with X10*P FXO cards?   Michael Graves
• [asterisk-users] Encrypting the Conversation   Michael Graves
• [asterisk-users] Astribank?   Michael Graves
• [asterisk-users] Astribank?   Michael Graves
• [asterisk-users] Just bought a Polycom 501 - I feel like myGXP-2000 was better...   Michael Graves
• [asterisk-users] Just bought a Polycom 501 - I feellike myGXP-2000 was better...   Michael Graves
• [asterisk-users] Recommend hard phone which supports IAX2?   Michael Graves
• [asterisk-users] Polycom 501 provisioning : how to secure values located in plein text files   Michael Graves
• [asterisk-users] Please suggest me Best VoIP Service Provider   Gbenga Great
• [asterisk-users] FW: $3,000 server   Joe Greco
• [asterisk-users] Zip code, city and area codes   Joe Greco
• [asterisk-users] Digium Zaptel volume issues   Charles K Green
• [asterisk-users] Digium Zaptel volume issues   Charles K Green
• [asterisk-users] Clicking noise when load XP100 zaptel driver (at boot time)   Andres Gregori
• [asterisk-users] Freeware sip/iax client windows mobile   Attilla De Groot
• [asterisk-users] Queue menu   Attilla De Groot
• [asterisk-users] Queue menu   Attilla De Groot
• [asterisk-users] HP DL380 and the TE4xxP cards   Edwin Groothuis
• [asterisk-users] Automation of call initiation   Nitin Gupta
• [asterisk-users] Upgrading my office - Need help   Gary Guthary
• [asterisk-users] tormenta2 drivers   BSD Guy
• [asterisk-users] FW: $3,000 server   Randall H.
• [asterisk-users] Inc.com Names Mark Spencer of Digium to its “30 Under 30: America’s Coolest Young Entrepreneurs”   Randall H.
• [asterisk-users] Zip code, city and area codes   Chris HARIGA
• [asterisk-users] Identifying invoking party for a feature   Wayne P. HIll
• [asterisk-users] Identifying invoking party for a feature   Wayne P. HIll
• [asterisk-users] Identifying invoking party for a feature   Wayne P. HIll
• [Asterisk-Users] RRMEMORY / Queues Not Working Right   Kyle Hagan
• [asterisk-users] Asterisk and NFS   Kyle Hagan
• [asterisk-users] Voicemails randomly not deleting in 1.2.9.1 ??   Kyle Hagan
• [Asterisk-Users] Voicemail options   Paul Hales
• [Asterisk-Users] Libpri + Zaptel + Asterisk polycom_acd_functions error message   Paul Hales
• [asterisk-users] Melbourne Asterisk meeting tonight   Paul Hales
• [asterisk-users] setting up an email to fax with asterisk   Paul Hales
• [asterisk-users] setting up an email to fax with asterisk   Paul Hales
• [asterisk-users] Problem with GotoIf in dialplan   Paul Hales
• [asterisk-users] Asterisk and NEC NEAX 2000 IPS   Paul Hales
• [asterisk-users] Polycom compatible phone for Asterisk   Paul Hales
• [asterisk-users] PRI dropouts   Paul Hales
• [asterisk-users] PRI dropouts - solution I hope...   Paul Hales
• [asterisk-users] zaptel on dual processor, How?   Paul Hales
• [asterisk-users] PRI and Asterisk   Paul Hales
• [asterisk-users] Unicall in Australia   Paul Hales
• [asterisk-users] RE: [asterisk-dev] How to send DNIS(B-party number) in IAX trunk   Paul Hales
• [asterisk-users] Voicemail not sent via email   Paul Hales
• [asterisk-users] Extension planning   Paul Hales
• [asterisk-users] Mobile SIP Client   Jim Hanlon
• [asterisk-users] Mobile SIP Client   Jim Hanlon
• [asterisk-users] mISDN configuration   James Harper
• [asterisk-users] Asterisk with ISDN Fritz PCI card   James Harper
• [Asterisk-Users] Very bad quality withAVMFritz!cardPCIandchan_capi   James Harper
• [Asterisk-Users] Very bad qualitywithAVMFritz!cardPCIandchan_capi   James Harper
• [Asterisk-Users] ISDN BRI NetJet   James Harper
• [asterisk-users] Using dproxy to solve "no DNS hangs everything"problem?   James Harper
• [asterisk-users] Using dproxy to solve "no DNS hangseverything"problem?   James Harper
• [asterisk-users] Load balenced (ADSL) network connections, is it possible?   James Harper
• [asterisk-users] Problem with making outgoing calls   Darrick Hartman
• [asterisk-users] PRI vs "Digital Trunk"   Barry D. Hassler
• [asterisk-users] PRI vs "Digital Trunk"   Barry D. Hassler
• [asterisk-users] Outgoing MSNs and chan_misdn   Gary Hawkins
• [asterisk-users] Outgoing MSNs and chan_misdn   Gary Hawkins
• [asterisk-users] Troubleshooting Random PRI disconnects   James Hawks
• [asterisk-users] Dropped Calls Need Help   James Hawks
• [asterisk-users] Test E1 channel   James Hawks
• [asterisk-users] Asterisk stops accepting calls   James Hawks
• [asterisk-users] CDRTools please help   Guido Hecken
• [asterisk-users] Redundant Ethernet   Guido Hecken
• [asterisk-users] NAT and externip problem or bug   Guido Hecken
• [asterisk-users] Strange behaviour Panasonic KX-TD1232   Gary G. Hendershot
• [asterisk-users] Operator in Voicemail   Henk
• [asterisk-users] Operator in Voicemail   Henk
• [asterisk-users] fax to HP machine   William Henrique
• [asterisk-users] RemoveQueueMember isn't working.   Keith Herrington
• [Asterisk-Users] dial if   Paul Hewlett
• [Asterisk-Users] Asterisk Native Sound Distortion (ulaw)   Paul Hewlett
• [Asterisk-Users] quadBRI in bri_net mode - t3 timer expired   Paul Hewlett
• [asterisk-users] Cisco 7960 Softkey templates   Scott Higginbotham
• [asterisk-users] Issues with MeetMe   Scott Higginbotham
• [asterisk-users] T.38 call with t38 in original SDP fails   David Hindmarsh
• [asterisk-users] T.38 call with t38 in original SDP fails   David Hindmarsh
• [Asterisk-Users] Quintum A400 Call Establishment Prob   Rizwan Hisham
• [Asterisk-Users] Need help with config-files   Rizwan Hisham
• [Asterisk-Users] Quintum A400 Configuration   Rizwan Hisham
• [Asterisk-Users] Asterisk Shutdown !!!   Rizwan Hisham
• [Asterisk-Users] Bridging Prob:::I guess   Rizwan Hisham
• [asterisk-users] Invite someone to Conference   Rizwan Hisham
• [asterisk-users] Invite someone to Conference   Rizwan Hisham
• [asterisk-users] DTMF   Rizwan Hisham
• [asterisk-users] Invite someone to Conference   Rizwan Hisham
• [asterisk-users] Invite someone to Conference   Rizwan Hisham
• [asterisk-users] Invite someone to Conference   Rizwan Hisham
• [asterisk-users] Problem With Transfering Calls.   Rizwan Hisham
• [asterisk-users] DTMF   Rizwan Hisham
• [asterisk-users] AGI tutorials   Rizwan Hisham
• [asterisk-users] AGI tutorials   Rizwan Hisham
• [asterisk-users] Channel Redirect   Rizwan Hisham
• [asterisk-users] Re: Channel Redirect   Rizwan Hisham
• [asterisk-users] Transfer Application   Rizwan Hisham
• [asterisk-users] DTMF in QUEUES dont work   Rizwan Hisham
• [asterisk-users] DTMF   Rizwan Hisham
• [asterisk-users] Transfering a caller_in_queue to a conference room   Rizwan Hisham
• [asterisk-users] Hitting # to Transfer out of a Queue   Rizwan Hisham
• [asterisk-users] Transfering Problem   Rizwan Hisham
• [asterisk-users] g729.1 + g723.1 codec conversion   Jean-Michel Hiver
• [asterisk-users] CDR Call Status   Jean-Michel Hiver
• [asterisk-users] How to do load balancing (1:1) with IAX and two different ISPs   Jean-Michel Hiver
• [asterisk-users] 1000s of extensions in one context?   Jean-Michel Hiver
• [asterisk-users] How do you harden an Asterisk install?   Jean-Michel Hiver
• [asterisk-users] SIP and NAT   Jean-Michel Hiver
• [asterisk-users] Re: Hints to help me debug cdr_odbc not inserting   Jean-Michel Hiver
• [asterisk-users] IGNORE: test email   Nick Hoffman
• [asterisk-users] Codec Negotiation   Nick Hoffman
• [asterisk-users] Asterisk/GPL and G.729 licensing   Nick Hoffman
• [asterisk-users] Problem incoming calls from sipphone/giztmo   Andres Holguin
• [asterisk-users] Invalid module format (ztdummy)   Ben Holt
• [asterisk-users] SIP Woes   Dave Hope
• [asterisk-users] SIP Woes   Dave Hope
• [asterisk-users] SIP Woes   Dave Hope
• [asterisk-users] SIP Woes   Dave Hope
• [asterisk-users] problem with iax -> sip across 2 asterisks   Guus Houtzager
• [asterisk-users] Which Fax Solution really works on IAX or SIP?   Lee Howard
• [asterisk-users] Certain fax types cause problems   Lee Howard
• [asterisk-users] Zap channel faxing in or out fails but phone calls work.   Lee Howard
• [asterisk-users] Zap channel faxing in or out fails but phone calls work.   Lee Howard
• [asterisk-users] Zap channel faxing in or out fails but phone calls work.   Lee Howard
• [asterisk-users] Asterisk, IAXModem and Hylafax   Lee Howard
• [asterisk-users] Regression testing dialplan changes   Nic Hughes
• [asterisk-users] Regression testing dialplan changes   Nic Hughes
• [asterisk-users] SNOM 360   Robbie Hughes
• [Asterisk-Users] Re: Digium Hardware Reliability   Bruno Wolff III
• [asterisk-users] Re: Digium Hardware Reliability   Bruno Wolff III
• [Asterisk-Users] Libpri + Zaptel + Asterisk polycom_acd_functions error message   Dean at INKnBITs
• [Asterisk-Users] Libpri + Zaptel + Asterisk polycom_acd_functionserror message   Dean at INKnBITs
• [asterisk-users] Polycom with Asterisk   Dean at INKnBITs
• [Asterisk-Users] Libpri + Zaptel + Asterisk polycom_acd_functions error message   Dean at INKnBITs
• [Asterisk-Users] Libpri + Zaptel + Asterisk polycom_acd_functionserror message   Dean at INKnBITs
• [Asterisk-Users] Libpri + Zaptel + Asterisk polycom_acd_functionserror message   Dean at INKnBITs
• [asterisk-users] Polycom ACD, Asterisk, Kernel 2.6   Dean at INKnBITs
• [asterisk-users] Polycom ACD, Asterisk, Kernel 2.6   Dean at INKnBITs
• [asterisk-users] Polycom ACD, Asterisk, Kernel 2.6 - now SIP does not register   Dean at INKnBITs
• [asterisk-users] Possible polycom_acd_functions BUG   Dean at INKnBITs
• [asterisk-users] RE: Possible polycom_acd_functions BUG   Dean at INKnBITs
• [asterisk-users] ACD rejected calls with out going to Voicemail   Dean at INKnBITs
• [asterisk-users] Call queue drops call after 1 min   Dean at INKnBITs
• [asterisk-users] Call queue drops call after 1 min   Dean at INKnBITs
• [asterisk-users] Polycom IP301 and Queues   Dean at INKnBITs
• [asterisk-users] Polycom IP301 and Queues   Dean at INKnBITs
• [asterisk-users] Voicemail and Polycom ip301   Dean at INKnBITs
• [asterisk-users] Polycom IP301 and Queues   Dean at INKnBITs
• [asterisk-users] SIP ATA Channels for outbound calls - How to select in dialplan   Dean at INKnBITs
• [asterisk-users] Asterisk internal extensions caller ID   Dean at INKnBITs
• [asterisk-users] Voicemail not sent via email   Dean at INKnBITs
• [asterisk-users] Voicemail not sent via email   Dean at INKnBITs
• [asterisk-users] Sipura ATA's Forwarding PSTN Calls to Asterisk   Dean at INKnBITs
• [asterisk-users] Polycom_acd_functions SIP trouble   Dean at INKnBITs
• [asterisk-users] Asterisk with Linksys SPA-3000   Dean at INKnBITs
• [asterisk-users] Polycom_acd_functions SIP trouble   Dean at INKnBITs
• [asterisk-users] Sending email after voicemail   Dean at INKnBITs
• [asterisk-users] Change the from@ using the voicemail.conf   Dean at INKnBITs
• [asterisk-users] Change the from@ using the voicemail.conf   Dean at INKnBITs
• [asterisk-users] wav49 for voicemail attachment not playing   Dean at INKnBITs
• [asterisk-users] Asterisk Polycom_acd_functions and G729   Dean at INKnBITs
• [asterisk-users] Asterisk Polycom_acd_functions and G729   Dean at INKnBITs
• [Asterisk-Users] Latest SVN of asterisk-addons doesn't compile   Vahram Igityan
• [asterisk-users] Asterisk 1.2.10 - Continually Restarting Logger   Koen Van Impe
• [Asterisk-Users] flash button on asterisk + legacy pbx system   Giorgio Incantalupo
• [Asterisk-Users] flash button on asterisk + legacy pbx system   Giorgio Incantalupo
• [Asterisk-Users] flash button on asterisk + legacy pbx system   Giorgio Incantalupo
• [Asterisk-Users] flash button on asterisk + legacy pbx system   Giorgio Incantalupo
• [Asterisk-Users] how to send flash command from asterisk to old pbx when pressing button on phone   Giorgio Incantalupo
• [Asterisk-Users] flash button on asterisk + legacy pbx system   Giorgio Incantalupo
• [asterisk-users] WARNING[30954]: chan_sip.c:2734 sip_indicate: Don't know how to indicate condition 9   Giorgio Incantalupo
• [Asterisk-Users] flash button on asterisk + legacy pbx system   Giorgio Incantalupo
• [Asterisk-Users] flash button on asterisk + legacy pbx system   Giorgio Incantalupo
• [asterisk-users] asterisk dual servers through iax: Accepting UNAUTHENTICATED call   Giorgio Incantalupo
• [asterisk-users] Re: asterisk dual servers through iax: Accepting UNAUTHENTICATED call   Giorgio Incantalupo
• [asterisk-users] Again on ISDN - MSN in Italy   Giorgio Incantalupo
• [asterisk-users] Again on ISDN - MSN in Italy   Giorgio Incantalupo
• [asterisk-users] Called party cannot hear caller   Giorgio Incantalupo
• [asterisk-users] asterisk extra sounds: what for?   Giorgio Incantalupo
• [asterisk-users] Problem with GotoIf in dialplan   Ira
• [asterisk-users] Provider UNREACHABLE   Ira
• [asterisk-users] Provider UNREACHABLE   Ira
• [asterisk-users] "Legacy" analog data modems and Asterisk   Ira
• [asterisk-users] SIP configuration by group   Ira
• [asterisk-users] Setvar=var=val in sip.conf   Ira
• [asterisk-users] emulating key system - pick up so and so on line 1   Ira
• [asterisk-users] New message   Ira
• [asterisk-users] New message   Ira
• [asterisk-users] Basic Asterisk Setup   K Y Iyer
• [asterisk-users] Basic Asterisk Setup   K Y Iyer
• [asterisk-users] Server Optimization and Load Balancing   Mitch Jackson
• [asterisk-users] "Legacy" analog data modems and Asterisk   Erik Jacobs
• [Asterisk-Users] Need help with config-files   Thomas Jacobsen
• [Asterisk-Users] Need help with config-files   Thomas Jacobsen
• [Asterisk-Users] Need help with config-files   Thomas Jacobsen
• [Asterisk-Users] Need help with config-files   Thomas Jacobsen
• [Asterisk-Users] Need help with config-files   Thomas Jacobsen
• [Asterisk-Users] Need help with config-files   Thomas Jacobsen
• [Asterisk-Users] Need help with config-files   Thomas Jacobsen
• [Asterisk-Users] Need help with config-files   Thomas Jacobsen
• [asterisk-users] PAP2 TUI Configuration Menu   Nabeel Jafferali
• [asterisk-users] NAT and externip problem or bug   Robert Jenkins
• [asterisk-users] NAT and externip problem or bug   Robert Jenkins
• [asterisk-users] NAT and externip problem or bug   Robert Jenkins
• [asterisk-users] NAT and externip problem or bug   Robert Jenkins
• [asterisk-users] Solved: NAT and externip problem or bug   Robert Jenkins
• [asterisk-users] Asterisk/GPL and G.729 licensing   Robert Jenkins
• SV: [Asterisk-Users] Nokia E61   Fredrik Emil Jensen
• SV: [Asterisk-Users] Nokia E61   Fredrik Emil Jensen
• [asterisk-users] Asterisk version: 1.2.9.1 or older?   Johann
• [asterisk-users] Stuck ACD Agents   Johann
• [asterisk-users] Queue Stats   Johann
• [Asterisk-Users] How to configure NOKIA N70 with Asterisk?   Olle E Johansson
• [Asterisk-Users] SIP debug logging   Olle E Johansson
• [asterisk-users] Bug in chan_sip mysql support and canreinvite?   Olle E Johansson
• [asterisk-users] SIP conf   Olle E Johansson
• [asterisk-users] control during registration process   Olle E Johansson
• [asterisk-users] Cisco 7960 SIP 8-3-0   Daryl Johnson
• [asterisk-users] zaptel errors   Justin Johnson
• [asterisk-users] Polycom 1.6.7 Firmware Messages Button   Peter Johnson
• [asterisk-users] Issues with MeetMe   Chris Jones
• [Asterisk-Users] Switchtype   Jerry Jones
• [Asterisk-Users] Aastra phones - disable call waiting   Jerry Jones
• [Asterisk-Users] Channel bank not work   Jerry Jones
• [Asterisk-Users] Channel bank not work   Jerry Jones
• [asterisk-users] Voicemail & CallerID   Jerry Jones
• [asterisk-users] CT3 cards   Jerry Jones
• R: [asterisk-users] Called number on ISDN   Jerry Jones
• [asterisk-users] Polycom config file location   Jerry Jones
• [asterisk-users] Polycom phone cycles between UNREACHABLE and REACHABLE   Jerry Jones
• [asterisk-users] Zap channel faxing in or out fails but phone calls work.   Jerry Jones
• [asterisk-users] Zap channel faxing in or out fails but phone calls work.   Jerry Jones
• [asterisk-users] Zap channel faxing in or out fails but phone calls work.   Jerry Jones
• [asterisk-users] Aastra 9133i w/NAT and Asterisk   Jerry Jones
• [asterisk-users] ftp setup for Polycom phones   Jerry Jones
• [asterisk-users] Operator Console(s)/Shared Call Appearances   Jerry Jones
• [asterisk-users] Just bought a Polycom 501 - I feel like my GXP-2000 was better...   Jerry Jones
• [asterisk-users] vegastream 50 FXO DTMF Problem   Jerry Jones
• [asterisk-users] vegastream 50 FXO DTMF Problem   Jerry Jones
• [asterisk-users] Just bought a Polycom 501 - I feel likemyGXP-2000 was better...   Jerry Jones
• [asterisk-users] Polycom 501 - How to set handset Volume   Jerry Jones
• [asterisk-users] long distance ethernet & Asterisk   Jerry Jones
• [asterisk-users] Operator Console(s)/Shared Call Appearances   Mr. Jones
• [asterisk-users] Operator Console(s)/Shared Call Appearances   Mr. Jones
• [asterisk-users] Queue announcement issues   Phil Jordan
• [asterisk-users] Queue announcement issues   Phil Jordan
• [asterisk-users] Fonality vs TrixBox UI   Senad Jordanovic
• [asterisk-users] Fonality vs TrixBox UI   Senad Jordanovic
• [asterisk-users] Server redundancy   Senad Jordanovic
• [asterisk-users] VoipNow 1.2.0 Beta   Senad Jordanovic
• [asterisk-users] DNS lookups failing for SIP register   Senad Jordanovic
• [asterisk-users] VoipNow 1.2.0 Beta   Senad Jordanovic
• [asterisk-users] VoipNow 1.2.0 Beta   Senad Jordanovic
• [asterisk-users] Can not check voicemail from outside   Joseph
• [Asterisk-Users] dtmfmode=inband but SDP also indicates rfc2833   Martin Joseph
• [Asterisk-Users] Now that Nufone is dead...   Martin Joseph
• [asterisk-users] RE: Asterisk in Seattle   Martin Joseph
• [asterisk-users] intel vs amd motherboards   Martin Joseph
• [asterisk-users] Re: Feasability of using * for smallappartmentbuilding?   Martin Joseph
• [Asterisk-Users] vserver (Debian) - no tty: howto use /usr/sbin/safe_asterisk with "-c" for color CLI?   Martin Joseph
• [Asterisk-Users] Do you need a licence to connect a Ciscohardphone to Asterisk ?   Martin Joseph
• [Asterisk-Users] Do you need a licence to connect a Ciscohardphone to Asterisk ?   Martin Joseph
• [asterisk-users] Re: What's the story with X10*P FXO cards?   Martin Joseph
• [asterisk-users] Re: What's the story with X10*P FXO cards?   Martin Joseph
• [asterisk-users] FXS: No ringtone   Martin Joseph
• [asterisk-users] Choppy MOH (Cisco gateway)   Martin Joseph
• [asterisk-users] Mutiple Homes one asterisk box   Martin Joseph
• [asterisk-users] Provider UNREACHABLE   Martin Joseph
• [asterisk-users] Problem of Quality   Martin Joseph
• [asterisk-users] NuFone, please send the log file   Martin Joseph
• [Asterisk-Users] dtmfmode=inband but SDP also indicates rfc2833   Martin Joseph
• [asterisk-users] New Asterisk server crashes daily   Martin Joseph
• [asterisk-users] Asterisk version: 1.2.9.1 or older?   Martin Joseph
• [asterisk-users] Provider UNREACHABLE   Martin Joseph
• [asterisk-users] comcast info -- somewhat offtopic   Martin Joseph
• [asterisk-users] Provider UNREACHABLE   Martin Joseph
• [asterisk-users] Asterisk version: 1.2.9.1 or older?   Martin Joseph
• [asterisk-users] Setting a threshold for asterisk to take ZAP line off hook ?   Martin Joseph
• [asterisk-users] SRTP enabling   Martin Joseph
• [asterisk-users] Testing 911?   Martin Joseph
• [asterisk-users] Re: Asterisk 1.2.10 and Zaptel 1.2.7 released!   Martin Joseph
• [asterisk-users] Codec Negotiation   Martin Joseph
• [asterisk-users] LinkSys SPA 2002 ATA hardphone UNREACHABLE...!!!   Martin Joseph
• [asterisk-users] Please suggest me Best VoIP Service Provider   Martin Joseph
• [asterisk-users] Please suggest me Best VoIP Service Provider   Martin Joseph
• [asterisk-users] Asterisk process run amock   Martin Joseph
• [asterisk-users] Codec Negotiation   Martin Joseph
• [asterisk-users] Codec Negotiation   Martin Joseph
• [asterisk-users] Codec Negotiation   Martin Joseph
• [asterisk-users] Unicall, not HOW but WHY   Martin Joseph
• [asterisk-users] Typical Asterisk Company   Martin Joseph
• [asterisk-users] Automating the registration process   Martin Joseph
• [asterisk-users] Automating the registration process   Martin Joseph
• [asterisk-users] Codec Negotiation   Martin Joseph
• [asterisk-users] [OT] Windows softphone with handset support?   Martin Joseph
• [asterisk-users] cannot received calls in pstn line   Martin Joseph
• [asterisk-users] NAT and externip problem or bug   Martin Joseph
• [asterisk-users] NAT and externip problem or configuration problem   Martin Joseph
• [asterisk-users] Voice with echo   Martin Joseph
• [asterisk-users] Just bought a Polycom 501 - Ifeellike myGXP-2000 was better...   Martin Joseph
• [asterisk-users] bugs.digium.com   Martin Joseph
• [asterisk-users] Getting no Audio with G729   Martin Joseph
• [asterisk-users] wav49 for voicemail attachment not playing   Martin Joseph
• [asterisk-users] Asterisk VOIP / Mikrotik   Martin Joseph
• [asterisk-users] Asterisk 1.2.10 - Continually Restarting Logger   Martin Joseph
• [asterisk-users] Trunk transferring?   Martin Joseph
• [asterisk-users] Asterisk VOIP / Mikrotik   Martin Joseph
• [asterisk-users] Re: If you prefer to read this mail list as a forum ...   Martin Joseph
• [asterisk-users] Disconnection During Incoming Call   Henry F. Camacho Jr
• [asterisk-users] All Extensions Dropped   Henry F. Camacho Jr.
• [asterisk-users] SIP connections   Justin
• [asterisk-users] Provider UNREACHABLE   KC
• [asterisk-users] sending flash using DTMF   Osama Kamal
• [asterisk-users] MFC/R2 country and carrier specific protocol variants   Kanelbullar
• [asterisk-users] MFC/R2 country and carrier specific protocol variants   Kanelbullar
• [asterisk-users] zaptel errors   Mats Karlsson
• [asterisk-users] Upgrading my office - Need help   Mats Karlsson
• [asterisk-users] FC4(2.6.16-1.2108_FC4smp) meetme 404 Not Found   Zen Kato
• [Asterisk-Users] HP Proliant server?   Alejandro Kauffmann
• [asterisk-users] setting call-limits   Alejandro Kauffmann
• [asterisk-users] setting call-limits   Alejandro Kauffmann
• [asterisk-users] Asterisk autoloading of card modules   Alejandro Kauffmann
• [asterisk-users] Asterisk autoloading of card modules   Alejandro Kauffmann
• [Asterisk-Users] Re: quadBRI in bri_net mode - t3 timer expired   Sebastian Kayser
• [asterisk-users] Troubleshooting Random PRI disconnects   Shawn Kelley
• [asterisk-users] Asterisk Console Colorization Question   Shawn Kelley
• [asterisk-users] CallerID in UK on TalkTalk - different to BT?   Steve Kennedy
• [asterisk-users] CallerID in UK on TalkTalk - different to BT?   Steve Kennedy
• [Asterisk-Users] asterisk to mobile phone   Steve Kennedy
• [asterisk-users] Blended?   Steve Kennedy
• SV: [Asterisk-Users] Nokia E61   Steve Kennedy
• [asterisk-users] PRI died and Asterisk crashed   Steve Kennedy
• [asterisk-users] HP DL380 and the TE4xxP cards   Steve Kennedy
• [Asterisk-Users] svn trunk and call hold / transfers   Thomas Kenyon
• [Asterisk-Users] Nokia E61   Thomas Kenyon
• [Asterisk-Users] More g729 calls than licenses?   Thomas Kenyon
• [Asterisk-Users] IAX Phone?   Thomas Kenyon
• [asterisk-users] Rockwell Modem   Thomas Kenyon
• [asterisk-users] OT: Sipura SPA-3000 ATA Directing Calls to Asterisk   Thomas Kenyon
• [asterisk-users] OT: Sipura SPA-3000 ATA Directing Calls to Asterisk   Thomas Kenyon
• [asterisk-users] CallerID in UK on TalkTalk - different to BT?   Thomas Kenyon
• [asterisk-users] Phone Ring   Thomas Kenyon
• [asterisk-users] Tired of fax calls... :-/   Thomas Kenyon
• [asterisk-users] 2 Handsets, Same extension   Thomas Kenyon
• [asterisk-users] Tired of fax calls... :-/   Thomas Kenyon
• [asterisk-users] Issues with making Transfers   Thomas Kenyon
• [asterisk-users] Asterisk version: 1.2.9.1 or older?   Thomas Kenyon
• [asterisk-users] ATCOM's AG-188   Thomas Kenyon
• [asterisk-users] Re: Don't Hit # after 9 to get PSTN line   Thomas Kenyon
• [asterisk-users] Load balenced (ADSL) network connections, is it possible?   Thomas Kenyon
• [Asterisk-Users] Which ATA to test T.38 ? What about Linksys 3102   Thomas Kenyon
• [asterisk-users] VoipNow 1.2.0 Beta   Thomas Kenyon
• [asterisk-users] Identifying invoking party for a feature   Mindaugas Kezys
• [asterisk-users] Automating the registration process   Kevin Khanye
• [asterisk-users] Has anyone programmed their own user\client software for asterisk?   Kevin Khanye
• [asterisk-users] Has anybody in here created their own softphones?   Kevin Khanye
• [asterisk-users] AstLinux 0.4.2 Released   Kristian Kielhofner
• [asterisk-users] DTMF relay   Jason Kim
• [Asterisk-Users] Asterisk-Addons compile problem (cdr_addon_mysql.c)   Levis Kimotho
• [Asterisk-Users] Calling Extensions generates congestion when call answered   Levis Kimotho
• [Asterisk-Users] Calling Extensions generates congestion when call answered   Levis Kimotho
• [asterisk-users] Extensions dialing but fails on pickup   Levis Kimotho
• [asterisk-users] SOLVED: Re: Extensions dialing but fails on pickup   Levis Kimotho
• SOLVED: Re: [Asterisk-Users] Calling Extensions generates congestion when call answered   Levis Kimotho
• [asterisk-users] Exclude a certain route from using a trunk   Levis Kimotho
• [asterisk-users] Using DUNDi with TrixBox mini HOWTO   Levis Kimotho
• [asterisk-users] [Solved]Re: Exclude a certain route from using a trunk   Levis Kimotho
• [asterisk-users] GSM Module not picking up DTMF digits from VOIP FXO Gateway   Levis Kimotho
• [asterisk-users] test tone   Barry King
• [asterisk-users] Weird transcoding error (SIP, local channels): sip_write: Asked to transmit frame type 64, while native formats is 256 (read/write = 64/256)   Matt King
• [asterisk-users] !! Got a UA, but i'm in state 1   Matt King
• [Asterisk-Users] Nufone Tollfree Port   John Kington
• [Asterisk-Users] Nufone Tollfree Port   John Kington
• [Asterisk-Users] Now that Nufone is dead...   John Kington
• [Asterisk-Users] Now that Nufone is dead...   John Kington
• [Asterisk-Users] Now that Nufone is dead...   John Kington
• [asterisk-users] Phones cutting out.....again - PLEASE HELP!!!   Andrew Kirch
• [asterisk-users] intel vs amd motherboards   Andrew Kirch
• [asterisk-users] Asterisk version: 1.2.9.1 or older?   Andrew Kirch
• [asterisk-users] intel vs amd motherboards   Andrew D Kirch
• [asterisk-users] Dell PowerEdge 830   Andrew D Kirch
• [asterisk-users] Dell PowerEdge 830   Andrew D Kirch
• [asterisk-users] NuFone suggests to use Vonage!!!!   Andrew D Kirch
• [asterisk-users] NuFone, please send the log file   Andrew D Kirch
• [asterisk-users] NuFone, please send the log file   Andrew D Kirch
• [asterisk-users] NuFone, please send the log file   Andrew D Kirch
• [asterisk-users] NuFone, please send the log file   Andrew D Kirch
• [asterisk-users] NuFone, please send the log file   Andrew D Kirch
• [asterisk-users] NuFone, please send the log file   Andrew D Kirch
• [asterisk-users] NuFone, please send the log file   Andrew D Kirch
• [asterisk-users] NuFone, please send the log file   Andrew D Kirch
• [asterisk-users] NuFone, please send the log file   Andrew D Kirch
• [asterisk-users] core dumps when phpagi script ends?   Dan Kirshner
• [asterisk-users] loading graphic on a Cisco 7960   Edward F. Klimowicz
• [Asterisk-Users] asterisk to mobile phone   Tigran Kocharyan
• [Asterisk-Users] Call back features   Tigran Kocharyan
• [Asterisk-Users] Re: Digium Hardware Reliability   Tigran Kocharyan
• [Asterisk-Users] Skype gateway   Tigran Kocharyan
• [asterisk-users] NuFone, please send the log file   Andrew Kohlsmith
• [asterisk-users] Setvar=var=val in sip.conf   Andrew Kohlsmith
• [asterisk-users] Clocking Multiple T1 Cards   Andrew Kohlsmith
• [asterisk-users] Clocking Multiple T1 Cards   Andrew Kohlsmith
• [asterisk-users] Binary/unreadable configuration files?   Andrew Kohlsmith
• [asterisk-users] Just bought a Polycom 501 - I feel likemy GXP-2000 was better...   Andrew Kohlsmith
• [asterisk-users] Clocking Multiple T1 Cards   Andrew Kohlsmith
• [asterisk-users] Caller ID on Transfers   Andrew Kohlsmith
• [asterisk-users] bugs.digium.com   Andrew Kohlsmith
• [asterisk-users] IAX2 Connection fails over time...   Andrew Kohlsmith
• [Asterisk-Users] Re: Out of Office Auto Reply:   Koopmann, Jan-Peter
• [asterisk-users] Snom 300 headset with static noise   Koopmann, Jan-Peter
• [asterisk-users] How many users on an asterisk box behind a dsl canyou have   Koopmann, Jan-Peter
• [asterisk-users] CDR dest question   Koopmann, Jan-Peter
• [asterisk-users] SNOM 360   Koopmann, Jan-Peter
• [asterisk-users] cmd DIAL - Who picked up the call?   Koopmann, Jan-Peter
• [asterisk-users] cmd DIAL - Who picked up the call?   Koopmann, Jan-Peter
• [asterisk-users] SNOM 360   Koopmann, Jan-Peter
• [asterisk-users] Multiple dialing   Koopmann, Jan-Peter
• [asterisk-users] cmd DIAL - Who picked up the call?   Koopmann, Jan-Peter
• [asterisk-users] cmd DIAL - Who picked up the call?   Koopmann, Jan-Peter
• [asterisk-users] Re: Load balenced (ADSL) network connections, is it possible?   Stelios Koroneos
• [Asterisk-Users] HP Proliant server?   Jim Kou
• [asterisk-users] B2BUA Webbased and Click 2 dial apps   Apollon Koutlides
• [asterisk-users] zaphfc - problem   Marcin J. Kowalczyk
• [asterisk-users] zaphfc - problem   Marcin J. Kowalczyk
• [Asterisk-Users] asterisk shutdown   Anton Krall
• [Asterisk-Users] asterisk shutdown   Anton Krall
• [asterisk-users] SIP client with video???   Blake Krone
• [asterisk-users] Memory leak res_perl   Arjan Kroon
• [asterisk-users] Memory leak res_perl   Arjan Kroon
• [asterisk-users] asterisk + centos 4.3   Arun Kumar
• [asterisk-users] Request for some help....   K.N.Arun Kumar
• [Asterisk-Users] SOLVED: IAX jitter / clocking problem   Andy Kuo
• [Asterisk-Users] Help getting International Dialing setup in extensions.conf   Von L.
• [asterisk-users] International Dialing setup in extensions.conf   Von L.
• [asterisk-users] netstats like command for sip , Is there one ?   Mr. James W. Laferriere
• [asterisk-users] Can incoming alternate rings be discriminated?   Paul Lakra
• [asterisk-users] Can incoming alternate rings be discriminated?   Paul Lakra
• [asterisk-users] Can incoming alternate rings be discriminated?   Paul Lakra
• [asterisk-users] test tone   Edwin Lam
• [asterisk-users] zapata.conf pri   Edwin Lam
• [asterisk-users] call forwarding to mobile phone   Lito Lampitoc
• [asterisk-users] call forwarding to mobile phone   Lito Lampitoc
• [asterisk-users] call forwarding to mobile phone   Lito Lampitoc
• [asterisk-users] cannot received calls in pstn line   Lito Lampitoc
• [Asterisk-Users] can't dial Scotland ...   Steve Langstaff
• [Asterisk-Users] can't dial Scotland ...   Steve Langstaff
• [asterisk-users] WARNING[30954]: chan_sip.c:2734 sip_indicate: Don't know how to indicate condition 9   Steve Langstaff
• [asterisk-users] SIP To: header   Steve Langstaff
• [asterisk-users] SIP To: header   Steve Langstaff
• [asterisk-users] Re: Voice with echo   Steve Langstaff
• [Asterisk-Users] H323 Asterisk best practices   Joshua Laroff
• [asterisk-users] FW: Are FreePBX Extensions not being created inasterisk? & FOP question.   Niklas Larsson
• [asterisk-users] Suggesstion Required   Abdul Lateef
• [asterisk-users] Asterisk and VAD   Abdul Lateef
• [asterisk-users] SRTP enabling   Abdul Lateef
• [asterisk-users] Finding far end echo in Verizon network   Andrew Latham
• [asterisk-users] CSTA support for asterisk   Andrew Latham
• [Asterisk-Users] 0000491...   Christophorus Laube
• [asterisk-users] 2 NICs; Asterisk receives on eth1 and replies on eth0   Daniel Lawson
• [Asterisk-Users] callwaiting   Lenz
• [Asterisk-Users] Queues and annoucements   Lenz
• [asterisk-users] AGI tutorials   Lenz
• [asterisk-users] QueueMetrics 1.2.1 released today   Lenz
• [asterisk-users] QueueMetrics 1.2.1 released today   Lenz
• [asterisk-users] QueueMetrics 1.2.1 released today   Lenz
• [asterisk-users] Queue Persistence with queue.log   Lenz
• [Asterisk-Users] Asterisk -x option in 1.2.9.1   Tim C. Lewis
• [asterisk-users] taskset with asterisk   Tim C. Lewis
• [asterisk-users] FW: How to send DNIS(B-party number) in IAX trunk   Sam Liang
• [asterisk-users] RE: [asterisk-dev] How to send DNIS(B-party number) in IAX trunk   Sam Liang
• [asterisk-users] RE: [asterisk-dev] How to send DNIS(B-party number) in IAX trunk   Sam Liang
• [asterisk-users] Test E1 channel   Ralph Liebessohn
• [asterisk-users] Test E1 channel   Ralph Liebessohn
• [asterisk-users] Test E1 channel   Ralph Liebessohn
• [asterisk-users] Test E1 channel   Ralph Liebessohn
• [asterisk-users] Ringing timer   Ralph Liebessohn
• [Asterisk-Users] DTMF and ivr systems   Monty Lilburn
• [asterisk-users] SIP conf   Sharon Lim
• [asterisk-users] SIP conf   Sharon Lim
• [asterisk-users] No ringing on outgoing SIP calls.   Sharon Lim
• [asterisk-users] SIP configuration by group   Sharon Lim
• [asterisk-users] SIP configuration by group   Sharon Lim
• [asterisk-users] Voicemail dial pattern from old pbx   Sharon Lim
• [asterisk-users] Voicemail dial pattern from old pbx   Sharon Lim
• [asterisk-users] NAT problem   Jose Limeres
• [asterisk-users] Problem with NAT   Jose Limeres
• [asterisk-users] Problem with NAT   Jose Limeres
• [asterisk-users] Problem with NAT   Jose Limeres
• [asterisk-users] Problem with NAT   Jose Limeres
• [asterisk-users] Asterisk in Seattle   Philippe Lindheimer
• [asterisk-users] Asterisk in Seattle   Philippe Lindheimer
• [asterisk-users] Cisco 7960 SIP 8-3-0   Mailing List
• [asterisk-users] Odd SIP timeout   Mailing List
• [asterisk-users] Source Directory of ASterisk   Mailing List
• [Asterisk-Users] callwaiting   Mailing Lists
• [asterisk-users] Asterisk dead-air issues with Digium TE110P and IVR/meetme/internal directory-   Maxx Lobo
• [asterisk-users] Asterisk dead-air issues with Digium TE110P and IVR/meetme/internal directory-   Maxx Lobo
• [asterisk-users] Asterisk dead-air issues with Digium TE110 and IVR/meetme/internal directory-   Maxx Lobo
• [asterisk-users] Help with MusicOnHold   KokMeng Loh
• [asterisk-users] Asterisk in Seattle   Nilesh Londhe
• [asterisk-users] Asterisk in Seattle   Nilesh Londhe
• [asterisk-users] Freeware sip/iax client windows mobile   Nilesh Londhe
• [asterisk-users] IPKALL direct to asterisk bypassing FWD   Nilesh Londhe
• [asterisk-users] Ztdummy   Nilesh Londhe
• [Asterisk-Users] Motorola and Asterisk   Alexander Lopez
• [asterisk-users] Invite someone to Conference   Alexander Lopez
• [asterisk-users] Invite someone to Conference   Alexander Lopez
• [asterisk-users] Invite someone to Conference   Alexander Lopez
• [asterisk-users] Phones cutting out.....again - PLEASE HELP!!!   Alexander Lopez
• [asterisk-users] NuFone, please send the log file   Alexander Lopez
• [asterisk-users] NuFone, please send the log file   Alexander Lopez
• [asterisk-users] Testing 911?   Alexander Lopez
• [asterisk-users] "Legacy" analog data modems and Asterisk   Alexander Lopez
• [asterisk-users] Redundant Ethernet   Alexander Lopez
• [asterisk-users] Voicemail not sent via email   Alexander Lopez
• [asterisk-users] How to receive a phone call each time you receivean email ?   Alexander Lopez
• [asterisk-users] Connecting branch offices through IPsec tunnel --latency effects?   Alexander Lopez
• [asterisk-users] PRI vs "Digital Trunk"   Alexander Lopez
• [asterisk-users] FW: Conference   Alexander Lopez
• [asterisk-users] Ringing timer   Alexander Lopez
• [asterisk-users] Ringing timer   Alexander Lopez
• [asterisk-users] Getting no Audio with G729   Alexander Lopez
• [asterisk-users] Getting no Audio with G729   Alexander Lopez
• [asterisk-users] Asterisk AGI cmd Record   Alexander Lopez
• [asterisk-users] External call press 1   carl Lougher
• [asterisk-users] Finding far end echo in Verizon network   carl Lougher
• [asterisk-users] Macro help needed!!!!   carl Lougher
• [asterisk-users] Macro help needed!!!!   carl Lougher
• [asterisk-users] Asterisk Home on 64bit?   Al Lougher
• [asterisk-users] Help with router setup on new asterisk box   Al Lougher
• [asterisk-users] New Asterisk server crashes daily   Al Lougher
• [asterisk-users] New Asterisk server crashes daily   Al Lougher
• [asterisk-users] Choppy/Jittery playback at beginning of calls   Al Lougher
• [asterisk-users] odd sound between SIP & IAX clients   Joseph Love
• [asterisk-users] odd sound between SIP & IAX clients   Joseph Love
• [asterisk-users] odd sound between SIP & IAX clients   Joseph Love
• [Asterisk-Users] TDM400 won't answer or dial. Help?   Stewart Loving-Gibbard
• [asterisk-users] Text priority labels not working for me   Luki
• [asterisk-users] need a pointer about scripting asterisk   Luki
• [asterisk-users] LinkSys SPA 2002 ATA hardphone UNREACHABLE...!!!   Luki
• [asterisk-users] LinkSys SPA 2002 ATA hardphone UNREACHABLE...!!!   Luki
• [asterisk-users] Germany VOIP provider   Luki
• [asterisk-users] Change current working directory to /tmp   Luki
• [asterisk-users] Message waiting question...   Luki
• [Asterisk-Users] SIP debug logging   Nikolai Lusan
• [asterisk-users] asterisk and sip nat problems   Nikolai Lusan
• [asterisk-users] asterisk and sip nat problems   Nikolai Lusan
• [asterisk-users] NAT   Nikolai Lusan
• [asterisk-users] [Asteirsk-Users]TE110P configuration problem   Richard Lyman
• [asterisk-users] Hitting # to Transfer out of a Queue   Richard Lyman
• [asterisk-users] Associate manager events to a previous Originate action   Richard Lyman
• [asterisk-users] Re: Associate manager events to a previous Originate action   Richard Lyman
• [Asterisk-Users] Sangoma A200 woes   Jim Lynch
• [Asterisk-Users] Sangoma A200 woes   Jim Lynch
• [asterisk-users] Provider UNREACHABLE   Mike Lynchfield
• [asterisk-users] Server redundancy   Mike Lynchfield
• [asterisk-users] Rate or rank ITSP   Mike Lynchfield
• [asterisk-users] So many configuration files!   Mike Lynchfield
• [asterisk-users] Provider UNREACHABLE   Mike Lynchfield
• [asterisk-users] Server redundancy   Mike Lynchfield
• [asterisk-users] 2 legs and cdr's   Mike Lynchfield
• [Asterisk-Users] svn trunk and call hold / transfers   Julian Lyndon-Smith
• [asterisk-users] Help! Zap Startup failure: why is libpri not defined ?   Julian Lyndon-Smith
• [asterisk-users] voicemail realtime and MWI   Julian Lyndon-Smith
• [Asterisk-Users] Avaya 4610sw SIP setup problem   Tom Lynn
• [Asterisk-Users] Avaya 4610sw SIP setup problem   Tom Lynn
• [Asterisk-Users] Avaya 4610sw SIP setup problem   Tom Lynn
• [asterisk-users] Asterisk in Seattle   Tom Lynn
• [asterisk-users] RE: Asterisk in Seattle   Tom Lynn
• [asterisk-users] RE: Asterisk in Seattle   Tom Lynn
• [asterisk-users] Mutiple Homes one asterisk box   Tom Lynn
• [asterisk-users] 489 Bad Event   Tom Lynn
• [asterisk-users] Help with sip debug?   Tom Lynn
• [asterisk-users] X100P clone not working   Tom Lynn
• [asterisk-users] Re: Operator Console(s)/Shared Call Appearances   Tom Lynn
• [asterisk-users] Problem with GotoIf in dialplan   Eric Lyons
• [Asterisk-Users] Page() command and file playback   Doug Lytle
• [Asterisk-Users] request @default   Doug Lytle
• [asterisk-users] Cisco 7941/7961/7971 wont register with asterisk   Doug Lytle
• SV: [asterisk-users] Cisco 7941/7961/7971 wont register with asterisk   Doug Lytle
• [asterisk-users] Zap Channel not hanging up on Telco side   Doug Lytle
• [asterisk-users] Zap Channel not hanging up on Telco side   Doug Lytle
• [asterisk-users] NOT logging Callerid/Call Data?   Doug Lytle
• [asterisk-users] Help with MusicOnHold   Doug Lytle
• [asterisk-users] Voicemails randomly not deleting in 1.2.9.1 ??   Doug Lytle
• [asterisk-users] Help with MusicOnHold   Doug Lytle
• 'cpmv' instead of 'cp' Re: [asterisk-users] [tip]semicolon trouble:   Doug Lytle
• [asterisk-users] setting up an email to fax with asterisk   Doug Lytle
• [asterisk-users] Certain fax types cause problems   Doug Lytle
• [asterisk-users] Certain fax types cause problems   Doug Lytle
• [asterisk-users] Mandriva 2006 Cooker RPM for Asterisk 1.2.9   Doug Lytle
• [asterisk-users] Mandriva 2006 Cooker RPM for Asterisk 1.2.9   Doug Lytle
• [asterisk-users] Dialing timeouts   Doug Lytle
• [asterisk-users] Mandriva 2006 Cooker RPM for Asterisk 1.2.9   Doug Lytle
• [asterisk-users] Mandriva 2006 Cooker RPM for Asterisk 1.2.9   Doug Lytle
• [asterisk-users] Certain fax types cause problems   Doug Lytle
• [asterisk-users] Re: Asterisk Database   Doug Lytle
• [asterisk-users] Re: Asterisk Database   Doug Lytle
• [asterisk-users] Hitting # to Transfer out of a Queue   Doug Lytle
• [asterisk-users] Asterisk 1.2.7.1 Crashing   Doug Lytle
• [asterisk-users] Asterisk 1.2.7.1 Crashing   Doug Lytle
• [asterisk-users] Issues with MeetMe   Doug Lytle
• [asterisk-users] Transfers - No ringback or moh   Doug Lytle
• [asterisk-users] Circuit/channel Congestion   Doug Lytle
• [asterisk-users] Clocking Multiple T1 Cards   Doug Lytle
• [asterisk-users] Circuit/channel Congestion   Doug Lytle
• [asterisk-users] Transfers - No ringback or moh   Doug Lytle
• [asterisk-users] Circuit/channel Congestion   Doug Lytle
• [asterisk-users] Transfers - No ringback or moh   Doug Lytle
• [asterisk-users] Circuit/channel Congestion   Doug Lytle
• [asterisk-users] Transfers - No ringback or moh   Doug Lytle
• [asterisk-users] Circuit/channel Congestion   Doug Lytle
• [asterisk-users] Transfers - No ringback or moh   Doug Lytle
• [asterisk-users] Transfers - No ringback or moh   Doug Lytle
• [asterisk-users] Transfers - No ringback or moh   Doug Lytle
• [asterisk-users] Caller ID on Transfers   Doug Lytle
• [asterisk-users] Caller ID on Transfers   Doug Lytle
• [asterisk-users] Caller ID on Transfers   Doug Lytle
• [asterisk-users] Zap Problem   Doug Lytle
• [asterisk-users] G729 Softphone   Guillermo Salas M.
• [asterisk-users] NAT and externip problem or bug   Julian J. M.
• [asterisk-users] NAT and externip problem or bug   Julian J. M.
• [asterisk-users] Sony Ericsson F250m, Sipura 3000 and Asterisk   Julian J. M.
• [asterisk-users] app background   Julian J. M.
• [Asterisk-Users] Re: Digium Hardware Reliability   M.Hockings
• [Asterisk-Users] Re: Digium Hardware Reliability   M.Hockings
• [Asterisk-Users] Re: Digium Hardware Reliability   M.Hockings
• [Asterisk-Users] Re: Digium Hardware Reliability   M.Hockings
• [Asterisk-Users] Re: Digium Hardware Reliability   M.Hockings
• [asterisk-users] Problems with freePBX and Fax reception   M.Hockings
• [asterisk-users] Re: Still voice with echo   M.Hockings
• [asterisk-users] PAP2T always busy on incoming calls with zaptel   Olivier MONNET
• [Asterisk-Users] can't dial Scotland ...   Colin MacMillan
• [Asterisk-Users] can't dial Scotland ...   Colin MacMillan
• [Asterisk-Users] can't dial Scotland ...   Colin MacMillan
• [Asterisk-Users] SRTP   Mikael Magnusson
• [asterisk-users] SRTP   Mikael Magnusson
• [asterisk-users] (no subject)   Maloney, Michael
• [asterisk-users] PCMCIA card support   Mauricio Mantilla
• [asterisk-users] Re: PCMCIA card support   Mauricio Mantilla
• [asterisk-users] Re: PCMCIA card support   Mauricio Mantilla
• [asterisk-users] THOR-2 support   Mauricio Mantilla
• [asterisk-users] phpagi problem   Mauricio Mantilla
• [asterisk-users] phpagi problem   Mauricio Mantilla
• [asterisk-users] Simple But important question (for me)   Mauricio Mantilla
• [asterisk-users] sounds format   Mauricio Mantilla
• [Asterisk-Users] Qsig-Link * to Meridian 81c   Marcus.Rothe at bertelsmann.de
• [asterisk-users] dropping calls in the middle of conversation   Giannis Margaritis
• [asterisk-users] Disable the flash hook hold capability on a SIP-to-SIP or SIP-to-ZAP call?   Ricardo Martins
• [asterisk-users] Caller Prompts in a Queue??   Matt
• [Asterisk-Users] RRMEMORY / Queues Not Working Right   Matt
• [asterisk-users] Metermaid phone compatibility   Matt
• [asterisk-users] Voicemails randomly not deleting in 1.2.9.1 ??   Matt
• [asterisk-users] Can I register multiple TERMINATORS to a single account on IAX?   Matt
• [asterisk-users] Can I register multiple TERMINATORS to a single account on IAX?   Matt
• [asterisk-users] Asterisk version: 1.2.9.1 or older?   Matt
• [asterisk-users] Re: Asterisk version: 1.2.9.1 or older?   Matt
• [asterisk-users] Can I register multiple TERMINATORS to a single account on IAX?   Matt
• [Asterisk-Users] Nufone Tollfree Port   trixter aka Bret McDanel
• [Asterisk-Users] please remove the autoresponder   trixter aka Bret McDanel
• [Asterisk-Users] RE: Is there a search feature?   trixter aka Bret McDanel
• [Asterisk-Users] More g729 calls than licenses?   trixter aka Bret McDanel
• [Asterisk-Users] Re: vserver with no /dev/tty* how to run "asterisk -c" for a colored CLI?   trixter aka Bret McDanel
• [Asterisk-Users] Skype gateway   trixter aka Bret McDanel
• [Asterisk-Users] Skype gateway   trixter aka Bret McDanel
• [asterisk-users] What's the story with X10*P FXO cards?   trixter aka Bret McDanel
• [asterisk-users] Encrypting the Conversation   trixter aka Bret McDanel
• [asterisk-users] Asterisk stops abruptly   trixter aka Bret McDanel
• [asterisk-users] Provider UNREACHABLE   trixter aka Bret McDanel
• [asterisk-users] Provider UNREACHABLE   trixter aka Bret McDanel
• [asterisk-users] Rate or rank ITSP   trixter aka Bret McDanel
• [asterisk-users] NuFone, please send the log file   trixter aka Bret McDanel
• [asterisk-users] NuFone, please send the log file   trixter aka Bret McDanel
• [asterisk-users] NuFone, please send the log file   trixter aka Bret McDanel
• [asterisk-users] NuFone, please send the log file   trixter aka Bret McDanel
• [asterisk-users] NuFone, please send the log file   trixter aka Bret McDanel
• [asterisk-users] NuFone, please send the log file   trixter aka Bret McDanel
• [asterisk-users] Re: Asterisk 1.2.10 and Zaptel 1.2.7 released!   trixter aka Bret McDanel
• [asterisk-users] Re: Asterisk 1.2.10 and Zaptel 1.2.7 released!   trixter aka Bret McDanel
• [asterisk-users] [Fwd: where is the error?]   trixter aka Bret McDanel
• [asterisk-users] Re: Asterisk 1.2.10 and Zaptel 1.2.7 released!   trixter aka Bret McDanel
• [asterisk-users] Tf.voipmich.com - Broken?   trixter aka Bret McDanel
• [asterisk-users] Cisco 7941/7961/7971 wont register with asterisk   Dr Rodney G. McDuff
• [Asterisk-Users] Re: Digium Hardware Reliability   Raymond McKay
• [Asterisk-Users] Re: Digium Hardware Reliability   Raymond McKay
• [Asterisk-Users] Call waiting using free PBX   Raymond McKay
• [Asterisk-Users] tdm04b strange noise when answering calls   Raymond McKay
• [asterisk-users] sip codec convertion on the fly   Raymond McKay
• [asterisk-users] Possible Bug?   Raymond McKay
• [asterisk-users] menu system - configurator   Raymond McKay
• [asterisk-users] Encrypting the Conversation   Raymond McKay
• [asterisk-users] Encrypting the Conversation   Raymond McKay
• [asterisk-users] Typical Asterisk Company   Raymond McKay
• [asterisk-users] Very OT: For the Record   Jeremy McNamara
• [asterisk-users] Re: $3,000 server   Jeremy McNamara
• [asterisk-users] Re: $3,000 server   Jeremy McNamara
• [asterisk-users] FW: $3,000 server   Jeremy McNamara
• [asterisk-users] sip, dbsecret, and dundi   Jeremy McNamara
• [asterisk-users] Re: $3,000 server   Jeremy McNamara
• [asterisk-users] One extension to ring on multiple outside lines   Jeremy McNamara
• [asterisk-users] call forwarding to mobile phone   Rodrigo Mercado
• [asterisk-users] Asterisk v/s other Telephonic plants   Rodrigo Mercado
• [asterisk-users] Asterisk dead-air issues with Digium TE110P and IVR/meetme/internal directory-   Rodrigo Mercado
• [asterisk-users] Asterisk / Avaya 70XX   Rodrigo Mercado
• [asterisk-users] Tired of fax calls... :-/   Evert Meulie
• [asterisk-users] Re: Tired of fax calls... :-/   Evert Meulie
• [Asterisk-Users] can't run "cat $filename" inside scripts with system()   Robert Michel
• [solved] sorry... ; ) Re: [Asterisk-Users] can't run "cat $filename" inside scripts with system()   Robert Michel
• Mercie:) how to debug "System( script.sh )" Re: [solved] sorry... ; ) Re: [Asterisk-Users] can't run "cat $filename" inside scripts with system()   Robert Michel
• vim with syntax highlighting Re: [Asterisk-Users] Best GPL Gui?   Robert Michel
• [Asterisk-Users] vserver (Debian) - no tty: howto use /usr/sbin/safe_asterisk with "-c" for color CLI?   Robert Michel
• [Asterisk-Users] vserver with no /dev/tty* how to run "asterisk -c" for a colored CLI?   Robert Michel
• [Asterisk-Users] Re: vserver with no /dev/tty* how to run "asterisk -c" for a colored CLI?   Robert Michel
• [asterisk-users] [tip]semicolon trouble: System($(sleep 4; cp 1.call out)&) not working, but System($( sleep 4 && cp 1.call out)&) ; )   Robert Michel
• [Asterisk-Users] vserver (Debian) - no tty: howto use /usr/sbin/safe_asterisk with "-c" for color CLI?   Robert Michel
• [Asterisk-Users] vserver (Debian) - no tty: howto use /usr/sbin/safe_asterisk with "-c" for color CLI?   Robert Michel
• 'cpmv' instead of 'cp' Re: [asterisk-users] [tip]semicolon trouble: System($(sleep 4; cp 1.call out)&) not working, but System($( sleep 4 && cp 1.call out)&) ; )   Robert Michel
• 'cpmv' instead of 'cp' Re: [asterisk-users] [tip]semicolon trouble: System($(sleep 4; cp 1.call out)&) not working, but System($( sleep 4 && cp 1.call out)&) ; )   Robert Michel
• Ok "\; " works - let us modify asterisk.vim ; ) Re: [asterisk-users] [tip]semicolon trouble:   Robert Michel
• Ok "\; " works - let us modify asterisk.vim ; ) Re: [asterisk-users] [tip]semicolon trouble:   Robert Michel
• [Asterisk-Users] vserver (Debian) - no tty: howto use /usr/sbin/safe_asterisk with "-c" for color CLI?   Robert Michel
• [Asterisk-Users] vserver (Debian) - no tty: howto use /usr/sbin/safe_asterisk with "-c" for color CLI?   Robert Michel
• CLI> color (on|of) ? Re: [Asterisk-Users] vserver (Debian) - no tty: howto use /usr/sbin/safe_asterisk with "-c" for color CLI?   Robert Michel
• 'cpmv' instead of 'cp' Re: [asterisk-users] [tip]semicolon trouble: System($(sleep 4; cp 1.call out)&) not working, but System($( sleep 4 && cp 1.call out)&) ; )   Robert Michel
• 'cpmv' instead of 'cp' Re: [asterisk-users] [tip]semicolon trouble: System($(sleep 4; cp 1.call out)&) not working, but System($( sleep 4 && cp 1.call out)&) ; )   Robert Michel
• [asterisk-users] Asterisk instances on VPS   Robert Michel
• [asterisk-users] Asterisk instances on VPS   Robert Michel
• [asterisk-users] Callback: Dial(dummy) 10 seconds rining without costs?   Robert Michel
• [asterisk-users] Re: Callback: Dial(dummy) 10 seconds ringing without costs?   Robert Michel
• [asterisk-users] FXS adapters and Polycom phones   Mike
• [asterisk-users] SIP adapters questions   Mike
• [asterisk-users] Just bought a Polycom 501 - I feel like my GXP-2000 was better...   Mike
• [asterisk-users] Just bought a Polycom 501 - I feel likemy GXP-2000 was better...   Mike
• [asterisk-users] RE: Just bought a Polycom 501 - I feel likemyGXP-2000 was   Mike
• [asterisk-users] Just bought a Polycom 501 - Ifeellike myGXP-2000 was better...   Mike
• [asterisk-users] Just bought a Polycom 501 - Ifeellike myGXP-2000was better...   Mike
• [asterisk-users] Polycom 501 provisioning : how to secure values located in plein text files   Mike
• [Asterisk-Users] Now that Nufone is dead...   Jay Milk
• [Asterisk-Users] Do you need a licence to connect a Ciscohardphone to Asterisk ?   Jay Milk
• [asterisk-users] Global variables and AGI   Jay Milk
• [asterisk-users] Re: Overriding # at the end   Jay Milk
• [asterisk-users] Germany VOIP provider   Jay Milk
• [asterisk-users] 2 NICs; Asterisk receives on eth1 and replieson eth0   Avi Miller
• [asterisk-users] sending out fax using asterisk   Avi Miller
• [asterisk-users] sending out fax using asterisk   Avi Miller
• [asterisk-users] Polycom config file location   Avi Miller
• [asterisk-users] Issue with g729 codec   Chris Miller
• [asterisk-users] Re: $3,000 server   Harvey Miller
• [asterisk-users] Re: $3,000 server   Harvey Miller
• [asterisk-users] Re: $3,000 server   Harvey Miller
• [asterisk-users] Polycom IP301 and Queues   Michael Miller
• [asterisk-users] Polycom IP301 and Queues   Michael Miller
• [asterisk-users] Polycom IP301 and Queues   Michael Miller
• [asterisk-users] Polycom_acd_functions SIP trouble   Michael Miller
• [asterisk-users] Zap channel faxing in or out fails but phone calls work.   Gregory L Miller-Kramer
• [asterisk-users] Re: Zap channel faxing in or out fails but phone calls work.   Gregory L Miller-Kramer
• [asterisk-users] Net::CSTA on CPAN   Gabriel Millerd
• [asterisk-users] Asterisk 1.2.10 - Continually Restarting Logger   Kenny Millington
• [asterisk-users] Asterisk 1.2.10 - Continually Restarting Logger   Kenny Millington
• [asterisk-users] Asterisk 1.2.10 - Continually Restarting Logger   Kenny Millington
• [asterisk-users] Asterisk 1.2.10 - Continually Restarting Logger   Kenny Millington
• [asterisk-users] stuck/phantom zap channels   Louis-David Mitterrand
• [asterisk-users] Unicall reload problem   Jesus Mogollon
• [asterisk-users] Any Polycom dealers willing help out?   Mojo with Horan & Company, LLC
• [asterisk-users] Setting a threshold for asterisk to take ZAP line off hook ?   Mojo with Horan & Company, LLC
• [asterisk-users] Injecting prerecorded audio into active call   Mojo with Horan & Company, LLC
• [asterisk-users] Polycom - simpler transfers?   Mojo with Horan & Company, LLC
• [asterisk-users] Polycom config file location   Mojo with Horan & Company, LLC
• [asterisk-users] Polycom config file location   Mojo with Horan & Company, LLC
• [asterisk-users] OT: FOP examples   Mojo with Horan & Company, LLC
• [asterisk-users] ftp setup for Polycom phones   Mojo with Horan & Company, LLC
• [asterisk-users] ftp setup for Polycom phones   Mojo with Horan & Company, LLC
• [asterisk-users] ftp setup for Polycom phones   Mojo with Horan & Company, LLC
• [asterisk-users] Caller ID on Transfers   Mojo with Horan & Company, LLC
• [asterisk-users] Ringing timer   Mojo with Horan & Company, LLC
• [asterisk-users] Invite someone to Conference   Jimmy Montano
• [asterisk-users] Delay on ring after dial Out   Pablo Mora
• [Asterisk-Users] how to decrease answer time !   Pablo Mora
• [asterisk-users] asterisk + centos 4.3   Pablo Mora
• [asterisk-users] Don't Hit # after 9 to get PSTN line   Pablo Mora
• [asterisk-users] Re: Don't Hit # after 9 to get PSTN line   Pablo Mora
• [asterisk-users] Don't Hit # after 9 to get PSTN line   Pablo Mora
• [asterisk-users] Re: Don't Hit # after 9 to get PSTN line   Pablo Mora
• [asterisk-users] Strange behaviour Panasonic KX-TD1232   Pablo Mora
• [asterisk-users] Transfer call in SIP   Victor Moreno
• [asterisk-users] One extension to ring on multiple outside lines   Dave Morrow
• [asterisk-users] One extension to ring on multiple outside lines   Dave Morrow
• [asterisk-users] Problem With Transfering Calls.   Shad Mortazavi
• [asterisk-users] Mobile SIP Client   Shad Mortazavi
• [asterisk-users] Mobile SIP Client   Shad Mortazavi
• [asterisk-users] Merlin Legend PRI not quite there yet   Sterling Moses
• [asterisk-users] ztmonitor in numeric mode   Robert Moskowitz
• [asterisk-users] ztmonitor in numeric mode   Robert Moskowitz
• [asterisk-users] zap and fax   Robert Moskowitz
• [asterisk-users] Re: CentOS 4.3 and Zaptel-1.2.7   Tony Mountifield
• [asterisk-users] SIP Registration conundrum   Tony Mountifield
• [asterisk-users] Re: SIP Registration conundrum   Tony Mountifield
• [asterisk-users] Re: Associate manager events to a previous Originate action   Tony Mountifield
• [asterisk-users] Possible Bug?   Marco Mouta
• [asterisk-users] Possible Bug?   Marco Mouta
• [asterisk-users] B2BUA Webbased and Click 2 dial apps   Marco Mouta
• [asterisk-users] How to collect Call duration, Dialout Call files?   Marco Mouta
• [asterisk-users] mISDN configuration   Marco Mouta
• [asterisk-users] How to collect Call duration, Dialout Call files?   Marco Mouta
• [asterisk-users] How to collect Call duration, Dialout Call files?   Marco Mouta
• [asterisk-users] How to collect Call duration, Dialout Call files?   Marco Mouta
• [asterisk-users] Test E1 channel   Marco Mouta
• [asterisk-users] Test E1 channel   Marco Mouta
• [asterisk-users] Outgoing MSNs and chan_misdn   Marco Mouta
• [asterisk-users] CDR calls started via AstManProxy   Marco Mouta
• [asterisk-users] outgoing call problem   Marco Mouta
• [asterisk-users] Called number on ISDN   Marco Mouta
• [asterisk-users] Linksys SPA941 - low Static Noise? or some parameter in hands   Marco Mouta
• [asterisk-users] Extensions Register but don't ring when called, can call others though   Marco Mouta
• [asterisk-users] Queue hold position in other language?   Marco Mouta
• [asterisk-users] Queue hold position in other language?   Marco Mouta
• [asterisk-users] If you prefer to read this mail list as a forum ...   Marco Mouta
• [asterisk-users] Codec Negotiation   Marco Mouta
• [asterisk-users] Problem with NAT   Marco Mouta
• [asterisk-users] Problem with NAT   Marco Mouta
• [asterisk-users] Problem with NAT   Marco Mouta
• [asterisk-users] Sipura ATA's Forwarding PSTN Calls to Asterisk   Marco Mouta
• [asterisk-users] Re: Voice with echo   Marco Mouta
• [asterisk-users] G729 License to Bridge calls through VOIP provider?   Marco Mouta
• [asterisk-users] Still voice with echo   Marco Mouta
• [asterisk-users] Still voice with echo   Marco Mouta
• [asterisk-users] SIP and podcasts   Marco Mouta
• [asterisk-users] New message   Marco Mouta
• [asterisk-users] G729 License to Bridge calls through VOIP provider?   Marco Mouta
• [asterisk-users] Invalid Conference Number - Meetme Created via FreePBX GUI   Marco Mouta
• [Asterisk-Users] Nokia E61   Devraj Mukherjee
• [Asterisk-Users] Nokia E61   Devraj Mukherjee
• SV: [Asterisk-Users] Nokia E61   Devraj Mukherjee
• [asterisk-users] Trouble configuring TDM400P on Dell SC420   Devraj Mukherjee
• [asterisk-users] Asterisk autoloading of card modules   Devraj Mukherjee
• [asterisk-users] Solution init.d scripts for CentOS 4.3   Devraj Mukherjee
• [asterisk-users] Asterisk autoloading of card modules   Devraj Mukherjee
• [asterisk-users] SIP response 400 Bad request   Devraj Mukherjee
• [asterisk-users] Asterisk SIP problems with Nokia E61   Devraj Mukherjee
• [asterisk-users] Re: Asterisk SIP problems with Nokia E61   Devraj Mukherjee
• [asterisk-users] NAT   Atif Munir
• [asterisk-users] compain   Atif Munir
• [asterisk-users] Polycom config file location   Stephen Murphy
• [asterisk-users] config files   Stephen Murphy
• [asterisk-users] ftp setup for Polycom phones   Stephen Murphy
• [asterisk-users] ftp setup for Polycom phones   Stephen Murphy
• [asterisk-users] ftp setup for Polycom phones   Stephen Murphy
• [asterisk-users] ftp setup for Polycom phones   Stephen Murphy
• [asterisk-users] ftp setup for Polycom phones   Stephen Murphy
• [asterisk-users] Intercom feature on Polycom phones   Stephen Murphy
• [asterisk-users] Intercom feature on Polycom phones   Stephen Murphy
• [asterisk-users] (no subject)   Ramya Murthy
• [asterisk-users] MFC/R2 country and carrier specific protocol variants   Dennis Nacino
• [asterisk-users] TE110P configuration problem   Dennis Nacino
• [Asterisk-Users] VoIP Cheap & Asterisk   Massimo De Nadal
• [Asterisk-Users] Work required - modify Asterisk + SEMS   Dinesh Nair
• [Asterisk-Users] WebPhone   Dinesh Nair
• [Asterisk-Users] WebPhone   Dinesh Nair
• [Asterisk-Users] WebPhone   Dinesh Nair
• [Asterisk-Users] WebPhone   Dinesh Nair
• [Asterisk-Users] WebPhone   Dinesh Nair
• SV: [Asterisk-Users] Nokia E61   Dinesh Nair
• [asterisk-users] Queue announcement issues   Dinesh Nair
• [asterisk-users] Queue announcement issues   Dinesh Nair
• [asterisk-users] VoipNow 1.2.0 Beta   Dinesh Nair
• [asterisk-users] don't hear start/begin of voiceprompts   Mein Name
• [asterisk-users] don't hear start/begin of voiceprompts   Mein Name
• [Asterisk-Users] Integrate asterisk with Database   Rajeev Natarajan
• [asterisk-users] Asterisk AGI cmd Record   Rajeev Natarajan
• [asterisk-users] Macro call uniqueid   Rajeev Natarajan
• [Asterisk-Users] WebPhone   Il Neofita
• [asterisk-users] Voicemail Contexts   Peder at NetworkOblivion
• [asterisk-users] Cisco Buddies   Peder at NetworkOblivion
• [asterisk-users] Cisco SIP Firmware   Peder at NetworkOblivion
• [asterisk-users] Problem - Can't pickup call   Peder at NetworkOblivion
• [asterisk-users] Voicemail & CallerID   Peder at NetworkOblivion
• [asterisk-users] Cisco Gateway & CallerID Name   Peder at NetworkOblivion
• [asterisk-users] RE: Cisco Gateway & CallerID Name   Peder at NetworkOblivion
• [Asterisk-Users] G729 and Cisco IOS 12.4   Peder at NetworkOblivion
• [asterisk-users] Operator Console(s)/Shared Call Appearances   Peder at NetworkOblivion
• [Asterisk-Users] Which ATA to test T.38 ? What about Linksys 3102   Peder at NetworkOblivion
• [asterisk-users] PRI Random Disconnected   chan (Alpha Trilogies Networks)
• [asterisk-users] Injecting prerecorded audio into active call   Nick
• [asterisk-users] Any Polycom dealers willing help out?   Darren Nickerson
• [Asterisk-Users] Duration for billing   Marnus van Niekerk
• [asterisk-users] B2BUA Webbased and Click 2 dial apps   Marnus van Niekerk
• [asterisk-users] setting of volume   Michael Nielsen
• [asterisk-users] trouble with * and # infront of a phonenumber   Michael Nielsen
• SV: [asterisk-users] trouble with * and # infront of a phonenumber   Michael Nielsen
• [asterisk-users] Mutiple Homes one asterisk box   Andrew Niemantsverdriet
• [asterisk-users] Mutiple Homes one asterisk box   Andrew Niemantsverdriet
• [asterisk-users] Mutiple Homes one asterisk box   Andrew Niemantsverdriet
• [Asterisk-Users] directory   John Novack
• [asterisk-users] not reaching at the destination number I dialed   John Novack
• [asterisk-users] agentcallbacklogin (logging out of)   Jordan Novak
• [asterisk-users] ACD Queues Agents logout   Jordan Novak
• [asterisk-users] Flash operator panel   Jordan Novak
• [asterisk-users] Flash operator panel   Jordan Novak
• [Asterisk-Users] Intel E7220 chipset?   Massimo Nuvoli
• [asterisk-users] IVR - Automatic Attendant database query   Massimo Nuvoli
• [asterisk-users] Hitting # to Transfer out of a Queue   Massimo Nuvoli
• [asterisk-users] Digium TE110P IRQ   Massimo Nuvoli
• [asterisk-users] overlapdial and DID not always working   Massimo Nuvoli
• [asterisk-users] overlapdial and DID not always working   Massimo Nuvoli
• [asterisk-users] overlapdial and DID not always working   Massimo Nuvoli
• SV: [Asterisk-Users] How to configure NOKIA N70 with Asterisk?   Amund Nygaard
• SV: SV: [Asterisk-Users] How to configure NOKIA N70 with Asterisk?   Amund Nygaard
• SV: SV: SV: [Asterisk-Users] How to configure NOKIA N70 with Asterisk?   Amund Nygaard
• SV: [Asterisk-Users] Nokia E61   Amund Nygaard
• SV: SV: [Asterisk-Users] Nokia E61   Amund Nygaard
• SV: [Asterisk-Users] How to configure NOKIA N70 with Asterisk?   Amund Nygaard
• [asterisk-users] Uninstalling Asterisk? No make uninstall?   Carey O'Shea
• [asterisk-users] Uninstalling Asterisk? No make uninstall?   Carey O'Shea
• [asterisk-users] Uninstalling Asterisk? No make uninstall?   Carey O'Shea
• [asterisk-users] Uninstalling Asterisk? No make uninstall?   Carey O'Shea
• [Asterisk-Users] Help with JIAXClient   Richard OSS
• [Asterisk-Users] Help with JIAXClient   Richard OSS
• [asterisk-users] Issue with g729 codec   Daniel Oakes
• [asterisk-users] Issue with g729 codec   Daniel Oakes
• [asterisk-users] Asterisk with Analogue cards   Obelix
• [Asterisk-Users] BLINDTRANSFER   Kai Ober
• [Asterisk-Users] BLINDTRANSFER   Kai Ober
• [asterisk-users] AGI tutorials   Kai Ober
• [asterisk-users] priority problem   Kai Ober
• [Asterisk-Users] Question setting up a "bat phone" extension.   Kai Ober
• [asterisk-users] don't hear start/begin of voiceprompts   Kai Ober
• [Asterisk-Users] Forward call   Kai Ober
• [asterisk-users] IVR DTMF   Kai Ober
• [Asterisk-Users] Question setting up a "bat phone" extension.   Kai Ober
• [Asterisk-Users] Question setting up a "bat phone" extension.   Kai Ober
• [asterisk-users] Bad luck installing bristuff on multiple Linux'es. Any one got a good-luck story I can repeat?   Kai Ober
• [asterisk-users] Ztdummy   Kai Ober
• [asterisk-users] Bad luck installing bristuff on multiple Linux'es. Any one got a good-luck story I can repeat?   Kai Ober
• [asterisk-users] all call forward   Kai Ober
• [asterisk-users] ACD Queues Agents logout   Kai Ober
• [asterisk-users] Binary/unreadable configuration files?   Kai Ober
• [asterisk-users] Binary/unreadable configuration files?   Kai Ober
• [asterisk-users] Binary/unreadable configuration files?   Kai Ober
• [asterisk-users] ACD Queues Agents logout   Kai Ober
• [asterisk-users] ACD Queues Agents logout   Kai Ober
• [asterisk-users] Voicmail Question   Kai Ober
• [asterisk-users] cmd DIAL - Who picked up the call?   Kai Ober
• [asterisk-users] Voicmail Question   Kai Ober
• [asterisk-users] cmd DIAL - Who picked up the call?   Kai Ober
• [asterisk-users] SRTP   Kai Ober
• [asterisk-users] cmd DIAL - Who picked up the call?   Kai Ober
• [asterisk-users] cmd DIAL - Who picked up the call?   Kai Ober
• [asterisk-users] Current radius patches   Natambu Obleton
• [asterisk-users] A very lost newbie.   Natambu Obleton
• [asterisk-users] question about asterisk DB   Natambu Obleton
• [asterisk-users] zaptel Disabled echo canceller because of tone (rx) on channel 2 work?   Vitaly Oborsky
• [asterisk-users] call-limit and problem with freezy phones. also freezy zap channels with x101p card.   Vitaly Oborsky
• [asterisk-users] what single PRI interface, from which manufacturer   Frank Ochmann
• [Asterisk-Users] How to configure NOKIA N70 with Asterisk?   Olivier
• [asterisk-users] Tired of fax calls... :-/   Olivier
• [Asterisk-Users] Do you need a licence to connect a Cisco hardphone to Asterisk ?   Olivier
• [Asterisk-Users] Do you need a licence to connect a Cisco hardphone to Asterisk ?   Olivier
• [asterisk-users] Tired of fax calls... :-/   Olivier
• [asterisk-users] Re: [Asterisk-java-users] A newbie introduction   Olivier
• [asterisk-users] Multiuser and analog port   Olivier
• [asterisk-users] How to receive a phone call each time you receive an email ?   Olivier
• [asterisk-users] MWI from Octel to Asterisk   Olivier
• [asterisk-users] CSTA support for asterisk   Olivier
• [Asterisk-Users] Which ATA to test T.38 ? What about Linksys 3102   Olivier
• [asterisk-users] alcatel ip touch 4068 ... sip?   Olivier
• [asterisk-users] Sip phone settings set when user registers   Olivier
• [asterisk-users] RE: alcatel ip touch 4068 ... sip?   Olivier
• [Asterisk-Users] Port doubler for 8 port BRI cards   Olivier
• [asterisk-users] CSTA support for asterisk   Olivier
• AW: [asterisk-users] Using dproxy to solve "no DNS hangs everything"problem?   John van Oppen
• [asterisk-users] Call Disconnects   J. Oquendo
• [Asterisk-Users] Motorola and Asterisk   Carlos Alberto Bernat Orozco
• [Asterisk-Users] PacketCable and Asterisk   Carlos Alberto Bernat Orozco
• [asterisk-users] Asterisk patches for packetcable   Carlos Alberto Bernat Orozco
• [asterisk-users] Asterisk and PacketCable PROJECT   Carlos Alberto Bernat Orozco
• [asterisk-users] Voice with echo   Carlos Alberto Bernat Orozco
• [asterisk-users] Re: Voice with echo   Carlos Alberto Bernat Orozco
• [asterisk-users] Still voice with echo   Carlos Alberto Bernat Orozco
• [asterisk-users] Still voice with echo   Carlos Alberto Bernat Orozco
• [asterisk-users] Developing VoIP with Asterisk   Carlos Alberto Bernat Orozco
• [asterisk-users] Developing VoIP with Asterisk (hardphones & softphones)   Carlos Alberto Bernat Orozco
• [asterisk-users] Developing VoIP with Asterisk   Carlos Alberto Bernat Orozco
• [Asterisk-Users] Do you need a licence to connect a Cisco hardphone to Asterisk ?   Florian Overkamp
• [Asterisk-Users] Do you need a licence to connect a Ciscohardphone to Asterisk ?   Florian Overkamp
• [Asterisk-Users] Do you need a licence to connect a Ciscohardphone to Asterisk ?   Florian Overkamp
• [asterisk-users] Extensions Register but don't ring when called, can call others though   Tim P
• [asterisk-users] 2 * servers, IAX connection between to dial extensions across IAX - not working   Tim P
• [asterisk-users] 2 * servers, IAX connection between to dial extensions across IAX - not working   Tim P
• [asterisk-users] 2 * servers, IAX connection between to dial extensions across IAX - not working   Tim P
• [asterisk-users] FreePBX Inbound Route   Tim P
• [asterisk-users] Asterisk Trunk Name Problem   Ondrej P.
• [asterisk-users] How to transfer other sessions   PSPunch
• [asterisk-users] Provider UNREACHABLE   Andres Paglayan
• [asterisk-users] Play sound mid way through call   Jonathan Palley
• [asterisk-users] Re: OT: Project Management & Collaboration Software   Jonathan Palley
• [asterisk-users] China   Jonathan Palley
• [Asterisk-Users] WebPhone   Tim Panton
• [asterisk-users] Re: for you guys setting up customer offices...   Tim Panton
• [asterisk-users] Re: Wrong account code from iax_buddies   Tim Panton
• [asterisk-users] Re: Wrong account code from iax_buddies   Tim Panton
• [asterisk-users] Re: Wrong account code from iax_buddies   Tim Panton
• [asterisk-users] Hints to help me debug cdr_odbc not inserting   Tim Panton
• [Asterisk-Users] GSM gateway flooded cell - how to detect?   Tim Panton
• [asterisk-users] Has anybody in here created their own softphones?   Tim Panton
• [asterisk-users] Has anybody in here created their own softphones?   Tim Panton
• [asterisk-users] Unicall, not HOW but WHY   Tim Panton
• [asterisk-users] H.323 an IAX   Tim Panton
• [asterisk-users] Recommend hard phone which supports IAX2?   Tim Panton
• [asterisk-users] Connecting branch offices through IPsec tunnel --latency effects?   Tim Panton
• [asterisk-users] odd sound between SIP & IAX clients   Tim Panton
• [asterisk-users] Manager interface   Tim Panton
• [asterisk-users] Manager interface   Tim Panton
• [asterisk-users] long distance ethernet & Asterisk   Tim Panton
• [asterisk-users] Manager interface   Tim Panton
• [asterisk-users] Alternative (?) ways to handle G.729 and annexb   Michail Pappas
• [asterisk-users] Rookie voicemail user question   Randy Paries
• [asterisk-users] Rookie voicemail user question   Randy Paries
• [Asterisk-Users] SRTP   Patrick
• [Asterisk-Users] Putting a call recording into a mailbox   Patrick
• [Asterisk-Users] trixbox 1.1 download   Patrick
• [Asterisk-Users] Skype gateway   Patrick
• [Asterisk-Users] Skype gateway   Patrick
• [asterisk-users] intel vs amd motherboards   Patrick
• [asterisk-users] CFWD Status with PHP   Patrick
• [asterisk-users] Voicemails randomly not deleting in 1.2.9.1 ??   Patrick
• [asterisk-users] New Asterisk server crashes daily   Patrick
• [asterisk-users] Asterisk + fax   Patrick
• [asterisk-users] Contacts for Chan_gsm_bt maintainer?   Patrick
• [asterisk-users] Setvar=var=val in sip.conf   Patrick
• [asterisk-users] Setvar=var=val in sip.conf   Patrick
• [asterisk-users] LinkSys SPA 2002 ATA hardphone UNREACHABLE...!!!   Patrick
• [asterisk-users] Hitting # to Transfer out of a Queue   Patrick
• [asterisk-users] HP DL380 and the TE4xxP cards   Patrick
• [asterisk-users] Canreinvite and remotely registered devices   Patrick
• [asterisk-users] Testers for ISDN AOC (Advice of Charge (Gespraechsgebuehren)) needed   Patrick
• [asterisk-users] Testers for ISDN AOC (Advice of Charge (Gespraechsgebuehren)) needed   Patrick
• [asterisk-users] Voice mail limit   Patrick
• [asterisk-users] Circuit/channel Congestion   Paul Traue, Jr.
• [asterisk-users] Caller Prompts in a Queue??   Aaron Paxson
• [asterisk-users] Caller Prompts in a Queue??   Aaron Paxson
• [asterisk-users] SNOM missed call.   Thomas Laurids Pedersen
• [asterisk-users] Circuit/channel Congestion   Thomas Laurids Pedersen
• [asterisk-users] SIP client with video???   Joao Pereira
• [asterisk-users] G726 codec softphone   Roberto Pereyra
• [asterisk-users] G726 codec softphone   Roberto Pereyra
• [Asterisk-Users] tdm04b strange noise when answering calls   Erick Perez
• [Asterisk-Users] tdm04b strange noise when answering calls   Erick Perez
• [asterisk-users] strange Fax or modem like tone when tdm400 answers pstn   Erick Perez
• [asterisk-users] xlite softphones: Got SUBSCRIBE for extensions without hint. Please add hint to 1001 in context   Erick Perez
• [asterisk-users] How to plot/graph fxotune -d data   Erick Perez
• [asterisk-users] Tired of fax calls... :-/   Erick Perez
• [asterisk-users] for you guys setting up customer offices...   Erick Perez
• [asterisk-users] setting up an email to fax with asterisk   Erick Perez
• [asterisk-users] setting up an email to fax with asterisk   Erick Perez
• [asterisk-users] setting up an email to fax with asterisk   Erick Perez
• [asterisk-users] setting up an email to fax with asterisk   Erick Perez
• [asterisk-users] OT: 3Com 3C10222 POE 24 Port Ethernet   Erick Perez
• [asterisk-users] Keeping stable 1.2.9.1 updated with patches   Erick Perez
• [asterisk-users] Asterisk stops abruptly   Erick Perez
• [asterisk-users] several asterisk servers questions   Erick Perez
• [asterisk-users] Asterisk stops abruptly   Erick Perez
• [asterisk-users] 1000s of extensions in one context?   Erick Perez
• [asterisk-users] an operational scenario   Erick Perez
• [asterisk-users] OT: 3Com 3C10222 POE 24 Port Ethernet   Erick Perez
• [asterisk-users] Asterisk 1.2.10 and Zaptel 1.2.7 released!   Erick Perez
• [asterisk-users] compiling zaptel 1.2.7 error: stray '\194' in program   Erick Perez
• [asterisk-users] Re: compiling zaptel 1.2.7 error: stray '\194' in program   Erick Perez
• [asterisk-users] How to create or test tone configuration to include them in zaptel   Erick Perez
• [asterisk-users] Asterisk and Phonesystems ...   Noc Phibee
• [Asterisk-Users] can't dial Scotland ...   Mark Phillips
• [asterisk-users] for you guys setting up customer offices...   Mark Phillips
• [asterisk-users] Re: $3,000 server   Aaron Picht
• [asterisk-users] CallerID   Wilson Pickett
• [asterisk-users] CallerID   Wilson Pickett
• [asterisk-users] Provider UNREACHABLE   Wilson Pickett
• [asterisk-users] Email notification of voicemail   Wilson Pickett
• [asterisk-users] Provider UNREACHABLE   Wilson Pickett
• [asterisk-users] Provider UNREACHABLE   Wilson Pickett
• [asterisk-users] zaptel on dual processor, How?   Olivier Picquenot
• [asterisk-users] zaptel on dual processor, How?   Olivier Picquenot
• [Asterisk-Users] SIP debug logging   William Piper
• [asterisk-users] Asterisk UAc / Request-URI   William Piper
• [asterisk-users] 'sip debug'   William Piper
• [asterisk-users] 'sip debug'   William Piper
• [asterisk-users] 'sip debug'   William Piper
• [asterisk-users] Re: $3,000 server   William Piper
• [asterisk-users] Asterisk Database   William Piper
• [asterisk-users] Can I register multiple TERMINATORS to a single account on IAX?   William Piper
• [asterisk-users] Can I register multiple TERMINATORS to a single account on IAX?   William Piper
• [asterisk-users] extensions.conf 4 digit dialing question   William Piper
• [asterisk-users] IP CDR   William Piper
• [asterisk-users] create custom cdr's   William Piper
• [asterisk-users] create custom cdr's   William Piper
• [asterisk-users] sip realtime   William Piper
• [asterisk-users] Rookie voicemail user question   William Piper
• [asterisk-users] Rookie voicemail user question   William Piper
• [asterisk-users] freepbx and a2billing   William Piper
• [asterisk-users] Voice mail limit   William Piper
• [asterisk-users] CDR IP Authorization   William Piper
• [asterisk-users] cmd DIAL - Who picked up the call?   William Piper
• [Asterisk-Users] PSTN Incoming Route   Pierre du Plessis
• [asterisk-users] buy X100p card in singapore   Pierre du Plessis
• [asterisk-users] MeetMe recordings in mp3 format.   Pierre du Plessis
• [asterisk-users] Asterisk version: 1.2.9.1 or older?   Don Pobanz
• [asterisk-users] Asterisk version: 1.2.9.1 or older?   Don Pobanz
• [asterisk-users] quad T1 pri   Don Pobanz
• [asterisk-users] asterisk + centos 4.3   Don Pobanz
• [asterisk-users] "Legacy" analog data modems and Asterisk   Don Pobanz
• [asterisk-users] Source Clock   Don Pobanz
• [asterisk-users] One extension to ring on multiple outside lines   Don Pobanz
• [asterisk-users] Voicemail dial pattern from old pbx   Don Pobanz
• [asterisk-users] Mitel 3300 + *   Barry Porch
• [asterisk-users] Mitel 3300 + *   Barry Porch
• [asterisk-users] long distance ethernet & Asterisk   Porier, Jeremy M.
• [asterisk-users] Invite someone to Conference   Vamsi Pottangi
• [asterisk-users] Asterisk AGI cmd Record   Prepaid
• [asterisk-users] CentOS 4.3 and Zaptel-1.2.7   Russ Price
• [asterisk-users] Bad luck installing bristuff on multiple Linux'es. Any one got a good-luck story I can repeat?   Cosmin Prund
• [asterisk-users] Bad luck installing bristuff on multiple Linux'es. Any one got a good-luck story I can repeat?   Cosmin Prund
• [asterisk-users] Bad luck installing bristuff on multiple Linux'es. Any one got a good-luck story I can repeat?   Cosmin Prund
• [asterisk-users] Bad luck installing bristuff on multiple Linux'es. Any one got a good-luck story I can repeat?   Cosmin Prund
• [asterisk-users] Bad luck installing bristuff on multiple Linux'es. Any one got a good-luck story I can repeat?   Cosmin Prund
• [asterisk-users] Install asterisk-bristuff for Debian Linux   Cosmin Prund
• [asterisk-users] Install asterisk-bristuff for Debian Linux   Cosmin Prund
• [Asterisk-Users] Work required - modify Asterisk + SEMS   Mike Puchol
• [asterisk-users] Encrypting the Conversation   Mike Puchol
• [asterisk-users] Invite someone to Conference   Joe Pukepail
• [asterisk-users] Urgent source code changes needed   Joe Pukepail
• [asterisk-users] PRI vs "Digital Trunk"   Joe Pukepail
• [asterisk-users] long distance ethernet & Asterisk   Joe Pukepail
• [asterisk-users] Postpaid   Dirgan Putra
• [asterisk-users] cdr functions change between * 1.2.4 and 1.2.9.1 (agi)   Ben Q
• [asterisk-users] Re: cdr functions change between * 1.2.4 and 1.2.9.1 (agi)   Ben Q
• [Asterisk-Users] TDM400 won't answer or dial. Help?   Mark Quitoriano
• [asterisk-users] A very lost newbie.   David R.
• [Asterisk-Users] performance & reliabulity of asterisk voicemail using odbc storage   RR
• [Asterisk-Users] performance & reliabulity of asterisk voicemail using odbc storage   RR
• [Asterisk-Users] performance & reliabulity of asterisk voicemail using odbc storage   RR
• [Asterisk-Users] performance & reliabulity of asterisk voicemail using odbc storage   RR
• [Asterisk-Users] performance & reliabulity of asterisk voicemail using odbc storage   RR
• [Asterisk-Users] RE: Is there a search feature?   RR
• [asterisk-users] Performance of Database Storage Vs Clustered File System   RR
• [asterisk-users] Server redundancy   RR
• [asterisk-users] Server redundancy   RR
• [asterisk-users] Server redundancy   RR
• [asterisk-users] 2 NICs; Asterisk receives on eth1 and replies on eth0   RR
• [asterisk-users] Asterisk Servers problem?   RR
• [asterisk-users] New Asterisk server crashes daily   RR
• [asterisk-users] Zaptel Compilation Error   RR
• [asterisk-users] Automating the registration process   RR
• [asterisk-users] meetme application doubt   RR
• [asterisk-users] Automating the registration process   RR
• [asterisk-users] Automating the registration process   RR
• [asterisk-users] question about asterisk DB   RR
• [asterisk-users] MeetMe in Realtime   RR
• [asterisk-users] Re: FW: meetme application doubt   RR
• [Asterisk-Users] Nokia E61   Antonio Rabena
• SV: [Asterisk-Users] Nokia E61   Antonio Rabena
• [asterisk-users] Connecting branch offices through IPsec tunnel --latency effects?   Kevin Ragsdale
• [asterisk-users] Hairpin Detection issues   Ramki
• [asterisk-users] Re: asterisk-users Digest, Vol 24, Issue 116   J Rangi
• [asterisk-users] Re: asterisk-users Digest, Vol 24, Issue 116   J Rangi
• [asterisk-users] Polycom_acd_functions SIP trouble   Faris Raouf
• [asterisk-users] Polycom_acd_functions SIP trouble   Faris Raouf
• [asterisk-users] Interested in IVR information   Robert Rawlinson
• [Asterisk-Users] Dynamic realtime with MWI working   Mark Reardon
• [asterisk-users] FXS adapters and Polycom phones   Tele Cost Price Reducer
• [asterisk-users] Rate or rank ITSP   Bruce Reeves
• [asterisk-users] Polycom compatible phone for Asterisk   Bruce Reeves
• [asterisk-users] config files   Bruce Reeves
• [asterisk-users] extensions.conf 4 digit dialing question   Bruce Reeves
• [asterisk-users] emulating key system - pick up so and so on line 1   Bruce Reeves
• [asterisk-users] Polycom 601 and Paging   Bruce Reeves
• [asterisk-users] Zap channel faxing in or out fails but phone calls work.   Bruce Reeves
• [asterisk-users] Typical Asterisk Company   Bruce Reeves
• [asterisk-users] ftp setup for Polycom phones   Bruce Reeves
• [asterisk-users] New message   Bruce Reeves
• [asterisk-users] New message   Bruce Reeves
• [asterisk-users] Clocking Multiple T1 Cards   Bruce Reeves
• [asterisk-users] Caller ID on Transfers   Bruce Reeves
• [asterisk-users] Anyone tried vitelity?   Bruce Reeves
• [asterisk-users] long distance ethernet & Asterisk   Bruce Reeves
• [asterisk-users] Announce queue?   Bruce Reeves
• [asterisk-users] Flash operator panel   Bruce Reeves
• [asterisk-users] Sip No Audio Both Side   Regi Reg
• [asterisk-users] inbound sip rtcp hangup   Vincent Regnard
• [asterisk-users] PRI issues with telco access codes   Josh Reineke
• [asterisk-users] RE: Asterisk in Seattle   Josh Reineke
• [asterisk-users] problems to call brazil from germany   Sebastian Reitenbach
• [asterisk-users] problems to call brazil from germany   Sebastian Reitenbach
• [asterisk-users] problems to call brazil from germany   Sebastian Reitenbach
• [asterisk-users] did sometimes not working   Sebastian Reitenbach
• [asterisk-users] did sometimes not working   Sebastian Reitenbach
• [asterisk-users] overlapdial and DID not always working   Sebastian Reitenbach
• [asterisk-users] overlapdial and DID not always working   Sebastian Reitenbach
• [asterisk-users] overlapdial and DID not always working   Sebastian Reitenbach
• [asterisk-users] overlapdial and DID not always working   Sebastian Reitenbach
• [asterisk-users] overlapdial and DID not always working   Sebastian Reitenbach
• [asterisk-users] Simple But important question (for me)   Stefan Reuter
• [asterisk-users] Associate manager events to a previous Originate action   Stefan Reuter
• [asterisk-users] bugs.digium.com   Stefan Reuter
• [asterisk-users] Manager interface   Stefan Reuter
• [asterisk-users] CSTA support for asterisk   Stefan Reuter
• [Asterisk-Users] Now that Nufone is dead...   John Reynolds
• [Asterisk-Users] RE: Is there a search feature?   Hadley Rich
• [asterisk-users] A very lost newbie.   Hadley Rich
• [asterisk-users] Information about Softphone support G729 ?   Hadley Rich
• [asterisk-users] Asterisk, IAXModem and Hylafax   JR Richardson
• [asterisk-users] Faxing over CCM SIP trunk to asterisk   John Riek
• [asterisk-users] SipAddHeaders Question   Steven Ringwald
• [asterisk-users] asterisk 1.2.9.1 and spandsp and rxfax   Rob Ristroph
• [asterisk-users] asterisk 1.2.9.1 and spandsp and rxfax   Rob Ristroph
• [asterisk-users] Re: CentOS 4.3 and Zaptel-1.2.7   Rob Ristroph
• [asterisk-users] Fonality vs TrixBox UI   Alex Robar
• [asterisk-users] NuFone, please send the log file   Alex Robar
• [asterisk-users] RE: $3,000 server   Alex Robar
• [asterisk-users] FW: $3,000 server   Alex Robar
• [asterisk-users] unhappy-about-VoIP-providers@googlegroups.com founded   Alex Robar
• [asterisk-users] DTMF detection and Sangoma cards   Alex Robar
• [asterisk-users] Polycom compatible phone for Asterisk   Alex Robar
• [asterisk-users] Using DUNDi with TrixBox mini HOWTO   Alex Robar
• [asterisk-users] Using DUNDi with TrixBox mini HOWTO   Alex Robar
• [asterisk-users] New York city Asterisk consultants   Alex Robar
• [asterisk-users] FW: Are FreePBX Extensions not being created in asterisk? & FOP question.   Alex Robar
• [asterisk-users] config files   Alex Robar
• [asterisk-users] Polycom config file location   Alex Robar
• [Asterisk-Users] Music on Hold from Soundcard   Alex Robar
• [asterisk-users] link quality is poor   Alex Robar
• [asterisk-users] MoH from Sound Card: Does it actually work?   Alex Robar
• [asterisk-users] Load balenced (ADSL) network connections, is it possible?   Alex Robar
• [asterisk-users] Load balenced (ADSL) network connections, is it possible?   Alex Robar
• [asterisk-users] How to connect 2 AAH   Alex Robar
• [asterisk-users] How to connect 2 AAH   Alex Robar
• [asterisk-users] ftp setup for Polycom phones   Alex Robar
• [asterisk-users] Germany VOIP provider   Alex Robar
• [asterisk-users] cannot received calls in pstn line   Alex Robar
• [asterisk-users] NAT   Alex Robar
• [asterisk-users] ERROR 1045 (28000): Access denied for user   Alex Robar
• [asterisk-users] Intercom feature on Polycom phones   Alex Robar
• [asterisk-users] Binary/unreadable configuration files?   Alex Robar
• [asterisk-users] FW: IP CDR   Alex Robar
• [asterisk-users] SIP and podcasts   Alex Robar
• [asterisk-users] Sangoma Stops Receiving Calls   Alex Robar
• [asterisk-users] Sangoma Stops Receiving Calls   Alex Robar
• [asterisk-users] Sangoma Stops Receiving Calls   Alex Robar
• [asterisk-users] Sangoma Stops Receiving Calls   Alex Robar
• [asterisk-users] adding a voice conversation recording on a existing PBX system   Alex Robar
• [asterisk-users] sip phone networking question [possibly OT]   Alex Robar
• [asterisk-users] Email notification of voicemail   Anthony Rodgers
• [asterisk-users] Email notification of voicemail   Anthony Rodgers
• [asterisk-users] Asterisk version: 1.2.9.1 or older?   Anthony Rodgers
• [asterisk-users] Email notification of voicemail   Anthony Rodgers
• [asterisk-users] ACD Queues Agents logout   Anthony Rodgers
• [asterisk-users] Caller ID on Transfers   Anthony Rodgers
• [asterisk-users] Strange Error when calling   Anthony Rodgers
• [asterisk-users] Extending call parking to display park extension on the handset display   Guillermo Roditi
• [asterisk-users] SendText() & displaying text messages on a SIP handset's screen   Guillermo Roditi
• [asterisk-users] SendText() & displaying text messages on a SIPhandset's screen   Guillermo Roditi
• [asterisk-users] SendText() & displaying text messages on aSIPhandset's screen   Guillermo Roditi
• [Asterisk-Users] Putting a call recording into a mailbox   Marc Rohlfing
• AW: [Asterisk-Users] Putting a call recording into a mailbox   Marc Rohlfing
• [asterisk-users] Multiple issues   Eric Rousse
• [asterisk-users] Re: Asterisk Database   Rushowr
• [asterisk-users] Re: Asterisk Database   Rushowr
• [asterisk-users] Clearing variables in the dialplan?   Rushowr
• [asterisk-users] Regression testing dialplan changes   Rushowr
• [asterisk-users] AEL2 Looping   Rushowr
• [asterisk-users] AEL2 Looping   Rushowr
• [asterisk-users] quad T1 pri   SANS
• [Asterisk-Users] Spoofing a BLF Signal?   Marc SCHAEFER
• [asterisk-users] Caller ID on a Sangoma   SQS Mail SQS
• [asterisk-users] Encrypting the Conversation   Alberto Sagredo
• [asterisk-users] Choppy MOH (Cisco gateway)   Alberto Sagredo
• [Asterisk-Users] H323 to SIP Gateway   Daniel Salama
• [Asterisk-Users] PSTN Incoming Route   Daniel Salama
• [asterisk-users] G729 Softphone   Daniel Salama
• [asterisk-users] G729 Softphone   Daniel Salama
• [asterisk-users] AGI Scripts and CDR   Daniel Salama
• [asterisk-users] AGI Scripts and CDR   Daniel Salama
• [asterisk-users] AGI Scripts and CDR   Daniel Salama
• [asterisk-users] Zaptel For new TE412P   Michael Sampson
• [asterisk-users] Text priority labels not working for me   Wes Santee
• [asterisk-users] Text priority labels not working for me   Wes Santee
• [Asterisk-Users] request @default   Olivier Saulnier
• [Asterisk-Users] request @default   Olivier Saulnier
• [Asterisk-Users] request @default   Olivier Saulnier
• [Asterisk-Users] Forward call   Olivier Saulnier
• [asterisk-users] Phone Ring   Olivier Saulnier
• [asterisk-users] Problem of Quality   Olivier Saulnier
• [asterisk-users] french promt   Olivier Saulnier
• [asterisk-users] Possible Bug?   Kevin Savoy
• [asterisk-users] RE: Asterisk in Seattle   Kevin Savoy
• [asterisk-users] Zap Channel not hanging up on Telco side   Kevin Savoy
• [asterisk-users] Zap Channel not hanging up on Telco side   Kevin Savoy
• [asterisk-users] Zap Channel not hanging up on Telco side   Kevin Savoy
• [asterisk-users] Unable to configure my DID number   Kevin Savoy
• [asterisk-users] Email notification of voicemail   Kevin Savoy
• [asterisk-users] Email notification of voicemail   Kevin Savoy
• [asterisk-users] Voicemail & CallerID   Kevin Savoy
• [asterisk-users] quad T1 pri   Kevin Savoy
• [asterisk-users] Email notification of voicemail   Kevin Savoy
• [asterisk-users] Email notification of voicemail   Kevin Savoy
• [BULK] Re: [Asterisk-Users] how to decrease answer time !   Kevin Savoy
• [asterisk-users] Email notification of voicemail   Kevin Savoy
• [asterisk-users] Operator in Voicemail   Kevin Savoy
• [asterisk-users] Operator in Voicemail   Kevin Savoy
• [asterisk-users] Operator in Voicemail   Kevin Savoy
• [asterisk-users] Choppy MOH (Cisco gateway)   John Sawa
• [Asterisk-Users] Channel bank not work   Bill Schaffer
• [asterisk-users] [Asteirsk-Users]TE110P configuration problem   Bill Schaffer
• [asterisk-users] [Asteirsk-Users]TE110P configuration problem   Bill Schaffer
• [Asterisk-Users] Channel bank not work   Bill Schaffer
• [asterisk-users] console/dsp and autoanswer   Bill Schaffer
• [asterisk-users] Dial plan question   Bill Schaffer
• [asterisk-users] what single PRI interface, from which manufacturer   Armin Schindler
• [asterisk-users] Connecting branch offices through IPsec tunnel -- latency effects?   Armin Schindler
• [asterisk-users] bugs.digium.com   Armin Schindler
• [asterisk-users] Testers for ISDN AOC (Advice of Charge (Gespraechsgebuehren)) needed   Armin Schindler
• [asterisk-users] Malformed/Missing URL Problem with Cisco Callmanager 4.1   David Schmitt
• [asterisk-users] Buch Bekanntmachung: Der Weg zu VoIP Asterisk von A bis Z   Silvio Schneider
• [asterisk-users] mgcp trouble   Christian Schnelle
• [asterisk-users] MySQL question   Schochet, Wes
• [Asterisk-Users] No reinvite - reason?   Roger Schreiter
• [asterisk-users] Bug in chan_sip mysql support and canreinvite?   Roger Schreiter
• [asterisk-users] 1000s of extensions in one context?   Roger Schreiter
• [asterisk-users] waitexten only provides one digit in chan_zap   Roger Schreiter
• [asterisk-users] 1000s of extensions in one context?   Roger Schreiter
• [asterisk-users] Reinvite and NAT -> Problem   Roger Schreiter
• [asterisk-users] Germany VOIP provider   Roger Schreiter
• [asterisk-users] SIP reinvite _and_ NAT   Roger Schreiter
• [asterisk-users] Zip code, city and area codes   Roger Schreiter
• [asterisk-users] Hardware... dimensioning ??   Javier Schussel
• [Asterisk-Users] ooh323 svn updated   Richard Scobie
• [asterisk-users] ooh323c - cdr   Richard Scobie
• [asterisk-users] Asterisk Realtime Macros   Jon Scottorn
• [asterisk-users] VoipNow 1.2.0 Beta   Miles Scruggs
• [asterisk-users] Operator Console(s)/Shared Call Appearances   Sebastian
• [asterisk-users] Asterisk with ISDN Fritz PCI card   Benjamin Sebbah
• [Asterisk-Users] Very bad quality with AVMFritz!cardPCIandchan_capi   Benjamin Sebbah
• [Asterisk-Users] Very bad quality withAVMFritz!cardPCIandchan_capi   Benjamin Sebbah
• [Asterisk-Users] Very bad quality withAVMFritz!cardPCIandchan_capi   Benjamin Sebbah
• [asterisk-users] IVR - Automatic Attendant database query   Dirk Enrique Seiffert
• [asterisk-users] New Asterisk server crashes daily   Roshan Sembacuttiaratchy
• [asterisk-users] Re: $3,000 server   Gonzalo Servat
• [asterisk-users] FW: $3,000 server   Gonzalo Servat
• [Asterisk-Users] Question setting up a "bat phone" extension.   Gonzalo Servat
• [asterisk-users] re:Simple But important question (for me)   Gonzalo Servat
• [asterisk-users] OT: Project Management & Collaboration Software   Gonzalo Servat
• [asterisk-users] Regular expression problem   Gonzalo Servat
• [asterisk-users] Recommend hard phone which supports IAX2?   Gonzalo Servat
• [asterisk-users] buy X100p card in singapore   Freddy Setiawan
• [asterisk-users] buy X100p card in singapore   Freddy Setiawan
• [asterisk-users] Re: Feasability of using * for small appartment building?   Kyle Sexton
• [asterisk-users] New GTK Gui for Monitoring and Administration   Kyle Sexton
• [asterisk-users] [announcement] kansas city asterisk user group   Kyle Sexton
• [asterisk-users] Announce queue?   Kyle Sexton
• [asterisk-users] RE: [Asterisk-video] Asterisk as an MCU   Curt Shaffer
• [asterisk-users] H323 implementation   Curt Shaffer
• [asterisk-users] SIP-> H323   Curt Shaffer
• [asterisk-users] Anyone tried vitelity?   Curt Shaffer
• [asterisk-users] Asterisk VOIP / Mikrotik   Curt Shaffer
• [asterisk-users] Asterisk VOIP / Mikrotik   Curt Shaffer
• [asterisk-users] Asterisk VOIP / Mikrotik   Curt Shaffer
• [asterisk-users] Error: Dropping incompatible voice frame   Tim Sharp
• [asterisk-users] for you guys setting up customer offices...   Shaun
• [asterisk-users] polycom 601 manual config?   Shaun
• [asterisk-users] Re: polycom 601 manual config?   Shaun
• [asterisk-users] Re: polycom 601 manual config?   Shaun
• [asterisk-users] delay and jitter issues..   T. Shaw
• [asterisk-users] for you guys setting up customer offices...   T. Shaw
• [asterisk-users] OT: FOP examples   T. Shaw
• [asterisk-users] Operator in Voicemail   T. Shaw
• [asterisk-users] Anyone tried vitelity?   T. Shaw
• [asterisk-users] sip phone networking question [possibly OT]   T. Shaw
• [Asterisk-Users] RRMEMORY / Queues Not Working Right   Shayne
• [asterisk-users] IAX2 Connection fails over time...   Stuart Sheldon
• [asterisk-users] Merlin Legend PRI not quite there yet   Stuart Sheldon
• [Asterisk-Users] dtmfmode=inband but SDP also indicates rfc2833   Stagg Shelton
• [Asterisk-Users] dtmfmode=inband but SDP also indicates rfc2833   Stagg Shelton
• [Asterisk-Users] dtmfmode=inband but SDP also indicates rfc2833   Stagg Shelton
• [Asterisk-Users] dtmfmode=inband but SDP also indicates rfc2833   Stagg Shelton
• [asterisk-users] Blog about asterisk and voip techology   Sergio Sicari
• [asterisk-users] billed calls when cellullar phone is unreachable   Andreas Sikkema
• [asterisk-users] ISDN Protocol   Lincoln Zuljewic Silva
• [asterisk-users] Zaptel Problem - Unable to create channel of type 'Zap'   Lincoln Zuljewic Silva
• [asterisk-users] Zaptel Problem - Unable to create channel of type 'Zap'   Lincoln Zuljewic Silva
• [asterisk-users] Source Clock   Lincoln Zuljewic Silva
• [asterisk-users] MFCR2 Patch   Lincoln Zuljewic Silva
• [asterisk-users] Digium TE110P IRQ   Lincoln Zuljewic Silva
• [asterisk-users] Circuit/channel Congestion   Lincoln Zuljewic Silva
• [asterisk-users] Circuit/channel Congestion   Lincoln Zuljewic Silva
• [asterisk-users] Circuit/channel Congestion   Lincoln Zuljewic Silva
• [asterisk-users] Circuit/channel Congestion   Lincoln Zuljewic Silva
• [asterisk-users] Circuit/channel Congestion   Lincoln Zuljewic Silva
• [asterisk-users] Circuit/channel Congestion   Lincoln Zuljewic Silva
• [asterisk-users] SIP and NAT   Lincoln Zuljewic Silva
• [asterisk-users] Test E1 channel   Moises Silva
• [asterisk-users] Test E1 channel   Moises Silva
• [asterisk-users] Test E1 channel   Moises Silva
• [asterisk-users] Connect to 'agi://blablabla' failed: Operation now in progress   Moises Silva
• [asterisk-users] IVR DTMF   Moises Silva
• [asterisk-users] CHANNEL STATUS of sip and iax devices   Moises Silva
• [asterisk-users] problems to call brazil from germany   Moises Silva
• [asterisk-users] show channels   Moises Silva
• [asterisk-users] Problem with MFCR2   Moises Silva
• [asterisk-users] Unicall libmfcr   Moises Silva
• [asterisk-users] Zaptel Problem - Unable to create channel of type 'Zap'   Moises Silva
• [asterisk-users] Is dmtfmode used/valid in iax.conf contexts?   Moises Silva
• [asterisk-users] Zaptel Problem - Unable to create channel of type 'Zap'   Moises Silva
• [asterisk-users] Has anybody in here created their own softphones?   Moises Silva
• [asterisk-users] Unicall in Australia   Moises Silva
• [asterisk-users] Problem with MFCR2   Moises Silva
• [asterisk-users] Unicall, not HOW but WHY   Moises Silva
• [asterisk-users] Unicall, not HOW but WHY   Moises Silva
• [asterisk-users] How to receive a phone call each time you receive an email ?   Moises Silva
• [asterisk-users] Unicall reload problem   Moises Silva
• [asterisk-users] playing a sound into a meetme conf   Moises Silva
• [asterisk-users] AGI Scripts and CDR   Moises Silva
• [Asterisk-Users] Avaya 4610sw SIP setup problem   Herchi Silviu
• [Asterisk-Users] Avaya 4610sw SIP setup problem   Herchi Silviu
• [asterisk-users] vegastream 50 FXO DTMF Problem   Issac Simchayof
• [asterisk-users] vegastream 50 FXO DTMF Problem   Issac Simchayof
• [asterisk-users] vegastream 50 FXO DTMF Problem   Issac Simchayof
• [asterisk-users] vegastream 50 FXO DTMF Problem   Issac Simchayof
• [asterisk-users] PRI channels filling up   Garth van Sittert
• [asterisk-users] Rxfax and squashed TIFF   Garth van Sittert
• [Asterisk-Users] Asterisk and HiSax   Tomasz Skwara
• [asterisk-users] Voicemails randomly not deleting in 1.2.9.1 ??   Kevin Smith
• [asterisk-users] PHP AGI   Kevin Smith
• [asterisk-users] PHP AGI   Kevin Smith
• [asterisk-users] PHP AGI   Kevin Smith
• [asterisk-users] PHP AGI   Kevin Smith
• [asterisk-users] Global variables and AGI   Kevin Smith
• [asterisk-users] Polycom, TFTP, and DHCP   Kevin Smith
• [asterisk-users] Global variables and AGI   Kevin Smith
• [asterisk-users] Polycom, TFTP, and DHCP   Kevin Smith
• [asterisk-users] Tough time getting Polycom phones to register after router reboot   Kevin Smith
• [asterisk-users] Polycom config file location   Kevin Smith
• [asterisk-users] Queue RoundRobin   Kevin Smith
• [asterisk-users] Polycom IP301 and Queues   Kevin Smith
• [asterisk-users] Queue Stats   Kevin Smith
• [Asterisk-Users] Polycom Soundpoint IP 301 w/ MGCP   Nathan C. Smith
• [asterisk-users] Got Mediatrix 1204 to work! now MWI and Poly com   Nathan C. Smith
• [Asterisk-Users] HP Proliant server?   Phil Smith
• [asterisk-users] Provider UNREACHABLE   Rick Smith
• [asterisk-users] Provider UNREACHABLE   Rick Smith
• [asterisk-users] Provider UNREACHABLE   Rick Smith
• [asterisk-users] Asterisk VOIP / Mikrotik   Rick Smith
• [asterisk-users] Asterisk VOIP / Mikrotik   Rick Smith
• [asterisk-users] Server redundancy   Christopher Snell
• [asterisk-users] Call Parking breaks suddenly   Christopher Snell
• [asterisk-users] DTMF detection and Sangoma cards   Christopher Snell
• [asterisk-users] DTMF detection and Sangoma cards   Christopher Snell
• [Asterisk-Users] Music on Hold from Soundcard   Christopher Snell
• [asterisk-users] MoH from Sound Card: Does it actually work?   Christopher Snell
• [asterisk-users] Cyberdata paging speakers - anyone use them?   Christopher Snell
• [asterisk-users] stream file outputs only silence, even with asterisk example gsm file   Guido Sohne
• [asterisk-users] Install Asterisk on VPS   Kanishka Somaratne
• [asterisk-users] mISDN configuration   Andrea Spadaccini
• [asterisk-users] mISDN configuration   Andrea Spadaccini
• [asterisk-users] mISDN configuration   Andrea Spadaccini
• [SOLVED] Re: [asterisk-users] mISDN configuration   Andrea Spadaccini
• [asterisk-users] Outgoing MSNs and chan_misdn   Andrea Spadaccini
• [asterisk-users] Asterisk version: 1.2.9.1 or older?   Andrea Spadaccini
• [asterisk-users] Again on ISDN - MSN in Italy   Andrea Spadaccini
• [asterisk-users] Again on ISDN - MSN in Italy   Andrea Spadaccini
• [asterisk-users] Again on ISDN - MSN in Italy   Andrea Spadaccini
• [asterisk-users] Again on ISDN - MSN in Italy   Andrea Spadaccini
• [asterisk-users] BudgeTone BT-102 not registering to Asterisk   Andrea Spadaccini
• [asterisk-users] sip realtime   Andrea Spadaccini
• [Asterisk-Users] voip-magazine article "Using DUNDiwithaClusterofAsterisk Servers"   Scott Spoofed
• [asterisk-users] One extension can transfer internal calls, can't transfer incoming external calls   Mat Stace
• [asterisk-users] Two phone numbers, one SIP provider   Mat Stace
• [asterisk-users] Two phone numbers, one SIP provider   Mat Stace
• [asterisk-users] Change the from@ using the voicemail.conf   Mat Stace
• [asterisk-users] Change the from@ using the voicemail.conf   Mat Stace
• [Asterisk-Users] Asterisk 1.2.7.1 with Polycom 501 on SIP -> Conf Calling   Mike Staver
• [Asterisk-Users] Asterisk 1.2.7.1 with Polycom 501 on SIP -> Conf Calling   Mike Staver
• [asterisk-users] problems to call brazil from germany   Johann Steinwendtner
• [asterisk-users] Other phone continues to ring when pick up a call with *8 on SVN HEAD   Chris Stenton
• [asterisk-users] Re: Asterisk and NFS   Steven
• [asterisk-users] Re: Re: Feasability of using * forsmallappartmentbuilding?   Steven
• [asterisk-users] Re: Metermaid phone compatibility   Steven
• [asterisk-users] Re: Re: TE420P/TE415P?   Steven
• [asterisk-users] Re: Re: Re: TE420P/TE415P?   Steven
• RE: [asterisk-users] Inc.com Names Mark Spencer of Digium to its “30 Under 30: America’s Coolest Young Entrepreneurs”   Steven
• [asterisk-users] FW: Are FreePBX Extensions not being created in asterisk? & FOP question.   Steven
• [asterisk-users] FW: Are FreePBX Extensions not being created inasterisk? & FOP question.   Steven
• [asterisk-users] Re: Don't Hit # after 9 to get PSTN line   Steven
• [asterisk-users] Re: Don't Hit # after 9 to get PSTN line   Steven
• [asterisk-users] Re: Don't Hit # after 9 to get PSTN line   Steven
• [asterisk-users] call forwarding to mobile phone   Steven
• [asterisk-users] Re: bugs.digium.com   Steven
• [asterisk-users] Re: If you prefer to read this mail list asa forum ...   Steven
• [asterisk-users] Re: FYI - first release of alarm response code.   Steven
• [asterisk-users] Re: How do you recompile individual source modules?   Steven
• [asterisk-users] Re: Voicemail dial pattern from old pbx   Steven
• [asterisk-users] Re: Automatic deletion of voicemail messages olderthan N days?   Steven
• [asterisk-users] Transfer Application   Benjamin Stocker
• [asterisk-users] Two phone numbers, one SIP provider   Benjamin Stocker
• [asterisk-users] Two phone numbers, one SIP provider   Benjamin Stocker
• [asterisk-users] Two phone numbers, one SIP provider   Benjamin Stocker
• [asterisk-users] Two phone numbers, one SIP provider   Benjamin Stocker
• [asterisk-users] question about asterisk DB   Benjamin Stocker
• [asterisk-users] Regular expression problem   Benjamin Stocker
• [asterisk-users] Voicemail not sent via email   Benjamin Stocker
• [asterisk-users] Regular expression problem   Benjamin Stocker
• [asterisk-users] Re: question about asterisk DB   Benjamin Stocker
• [asterisk-users] Trouble with call file   Rivy Strauss
• [asterisk-users] Trouble with call file   Rivy Strauss
• [asterisk-users] Snom 300 headset with static noise   Christian Stredicke
• [asterisk-users] SNOM missed call.   Christian Stredicke
• [asterisk-users] Snom 360   Christian Stredicke
• [asterisk-users] Problem with making outgoing calls   VoIP Street
• [asterisk-users] Email notification of voicemail   VoIP Street
• [asterisk-users] Email notification of voicemail   VoIP Street
• [asterisk-users] How many users on an asterisk box behind a dsl can you have   VoIP Street
• [asterisk-users] phpagi problem   VoIP Street
• [asterisk-users] Midnight Naughties   James Sturges
• [asterisk-users] PRI dropouts   James Sturges
• [asterisk-users] PRI dropouts - solution I hope...   James Sturges
• [asterisk-users] Asterisk instances on VPS   James Sturges
• [asterisk-users] PRI dropouts - solution I hope...   James Sturges
• [asterisk-users] Tired of fax calls... :-/   Technical Support
• [asterisk-users] Help with sip debug?   Shanon Swafford
• [asterisk-users] Testing 911?   Brian Swan
• [Asterisk-Users] Aastra phones - disable call waiting   I T
• [asterisk-users] Asterisk Servers problem?   ECP TECHNOLOGIES
• [asterisk-users] NOT logging Callerid/Call Data?   Administrator TOOTAI
• [asterisk-users] Freeware sip/iax client windows mobile   Administrator TOOTAI
• [asterisk-users] Zaptel trunk failed to compile   Administrator TOOTAI
• [asterisk-users] Zaptel trunk failed to compile   Administrator TOOTAI
• [asterisk-users] PSTN disconnect tones on voicemail messages   TWV
• [Asterisk-Users] asterisk to mobile phone   Sam Tam
• [asterisk-users] call forwarding to mobile phone   Sam Tam
• [asterisk-users] call forwarding to mobile phone   Sam Tam
• [asterisk-users] call forwarding to mobile phone   Sam Tam
• SV: [Asterisk-Users] Nokia E61   Sam Tam
• [asterisk-users] adding a voice conversation recording on a existing PBX system   Sam Tam
• [asterisk-users] Multi Asterisk Server to relay call request   Fadjar Tandabawana
• [asterisk-users] OT: Sipura SPA-3000 ATA DirectingCalls to Asterisk   AR Tarzi
• [Asterisk-Users] Channel bank not work   Viktor Tatianin
• [Asterisk-Users] Channel bank not work   Viktor Tatianin
• [Asterisk-Users] Channel bank not work   Viktor Tatianin
• [Asterisk-Users] Channel bank not work   Viktor Tatianin
• [Asterisk-Users] Channel bank not work   Viktor Tatianin
• [Asterisk-Users] Channel bank not work   Viktor Tatianin
• [asterisk-users] channel bank log   Viktor Tatianin
• [asterisk-users] Asterisk 1.2.10 and Zaptel 1.2.7 released!   Asterisk Development Team
• [asterisk-users] Any Polycom dealers willing help out?   MBIT Technologies
• [asterisk-users] [Asteirsk-Users]TE110P configuration problem   MBIT Technologies
• [asterisk-users] Asterisk and NEC NEAX 2000 IPS   MBIT Technologies
• [asterisk-users] Uni Call   MBIT Technologies
• [asterisk-users] Unicall libmfcr   MBIT Technologies
• [asterisk-users] Unicall in Australia   MBIT Technologies
• [asterisk-users] Recording/Monitor after xfer   Rodrigo P. Telles
• [asterisk-users] Re: Recording/Monitor after xfer   Rodrigo P. Telles
• [asterisk-users] Clocking Multiple T1 Cards   Shaw Terwilliger
• [asterisk-users] Clocking Multiple T1 Cards   Shaw Terwilliger
• [asterisk-users] Clocking Multiple T1 Cards   Shaw Terwilliger
• [asterisk-users] Clocking Multiple T1 Cards   Shaw Terwilliger
• [asterisk-users] Polycom ACD, Asterisk, Kernel 2.6   James Texter
• [asterisk-users] Playfile waiting for N digits   James Texter
• [asterisk-users] playback / stream file   Martin Schrott - Thinking-Systems
• [asterisk-users] Transfers - No ringback or moh   Martin Schrott - Thinking-Systems
• [asterisk-users] Transfers - No ringback or moh   Martin Schrott - Thinking-Systems
• [asterisk-users] Fritz!Box Fon ATA   Martin Schrott - Thinking-Systems
• [asterisk-users] Re: Fritz!Box Fon ATA   Martin Schrott - Thinking-Systems
• [asterisk-users] Re: Fritz!Box Fon ATA   Martin Schrott - Thinking-Systems
• [asterisk-users] Re: Fritz!Box Fon ATA   Martin Schrott - Thinking-Systems
• [asterisk-users] agentcallbacklogin Asterisk V1.210 and v1.4   Martin Schrott - Thinking-Systems
• [asterisk-users] Sip phone settings set when user registers   Bradley D. Thornton
• [asterisk-users] Bad luck installing bristuff on multiple Linux'es. Any one got a good-luck story I can repeat?   Mark Tinka
• [asterisk-users] Load balenced (ADSL) network connections, is it possible?   Mark Tinka
• [asterisk-users] TDM01B -1 FXO card not working.   Jan du Toit
• [asterisk-users] TDM01B -1 FXO card not working.   Jan du Toit
• [asterisk-users] MeetMe recordings in mp3 format.   Jan du Toit
• [asterisk-users] MeetMe recordings in mp3 format.   Jan du Toit
• [asterisk-users] VoipNow 1.2.0 Beta   Tom
• [asterisk-users] Polycom phone cycles between UNREACHABLE and REACHABLE   Tong
• [asterisk-users] Cisco 7960 SIP 8-3-0   Tong
• [asterisk-users] Issues with making Transfers   Tong
• [Asterisk-Users] attended transfer issue   Tong
• [asterisk-users] Tf.voipmich.com - Broken?   Brent Torrenga
• [asterisk-users] Re: Using dproxy to solve "no DNS hangs everything" problem?   Brent Torrenga
• [asterisk-users] Sipura ATA's Forwarding PSTN Calls to Asterisk   Jorge Mauricio Hernandez Torres
• [Asterisk-Users] test   Steve Totaro
• [asterisk-users] Fonality vs TrixBox UI   Steve Totaro
• [asterisk-users] Fonality vs TrixBox UI   Steve Totaro
• [asterisk-users] OT: 3Com 3C10222 POE 24 Port Ethernet   Steve Totaro
• [asterisk-users] NuFone, please send the log file   Steve Totaro
• [asterisk-users] Re: $3,000 server   Steve Totaro
• [asterisk-users] Re: $3,000 server   Steve Totaro
• [asterisk-users] Reload clears queue stats   Steve Totaro
• [asterisk-users] Agent Attended Transfer Without DTMF   Steve Totaro
• [asterisk-users] Queue Stats   Steve Totaro
• [asterisk-users] Queue Persistence with queue.log   Steve Totaro
• [asterisk-users] Asterisk Dial Plan to Play Message   Steve Totaro
• [asterisk-users] Asterisk Dial Plan to Play Message   Steve Totaro
• [asterisk-users] Asterisk Dial Plan to Play Message   Steve Totaro
• [asterisk-users] Goldmine CRM softphone + asterisk   Steve Totaro
• [asterisk-users] E1 connectivity question   Steve Totaro
• [asterisk-users] HP DL380 and the TE4xxP cards   Steve Totaro
• [asterisk-users] comcast info -- somewhat offtopic   Steve Totaro
• [asterisk-users] PRI dropouts - solution I hope...   Steven Totaro
• [asterisk-users] QueueMetrics 1.2.1 released today   Steven Totaro
• [asterisk-users] QueueMetrics 1.2.1 released today   Steven Totaro
• [asterisk-users] Nokia Primicell and Asterisk ? Hangup and Answer detection ?   Información Capa Tres
• [Asterisk-Users] Queues and annoucements   Tristan
• [Asterisk-Users] Queues and annoucements   Tristan
• [Asterisk-Users] Queues and annoucements   Tristan
• [asterisk-users] Problem With Transfering Calls.   Tristan
• [asterisk-users] Error on dial_exec_full   Tristan
• [asterisk-users] QueuePauseMember(|Agent/) question   Tristan
• [asterisk-users] QueuePauseMember(|Agent/) question   Tristan
• [asterisk-users] Queue menu   Tristan
• [asterisk-users] Asterisk 1.2.10 and Zaptel 1.2.7 released!   Tristan
• [asterisk-users] Dynamic Queue Members never called   Tristan
• [asterisk-users] create custom cdr's   Tristan
• [asterisk-users] Incoming Call matching to peer   Alyed Tzompa
• [asterisk-users] E1 additional calling party number   Alyed Tzompa
• [asterisk-users] Help with MusicOnHold!!!   Alyed Tzompa
• [asterisk-users] trouble with * and # infront of a phonenumber   Alyed Tzompa
• [asterisk-users] setting of volume   Alyed Tzompa
• [asterisk-users] So many configuration files!   Alyed Tzompa
• [asterisk-users] G729 Softphone   Alyed Tzompa
• [asterisk-users] SIP and NAT   Alyed Tzompa
• [asterisk-users] Clocking Multiple T1 Cards   Steve Underwood
• [asterisk-users] Rxfax and squashed TIFF   Steve Underwood
• [Asterisk-Users] Which ATA to test T.38 ? What about Linksys 3102   Steve Underwood
• [asterisk-users] CSTA support for asterisk   Steve Underwood
• [Asterisk-Users] callwaiting in queues   Urban
• [asterisk-users] RE: $3,000 server   VOICEIN at aol.com
• SV: [Asterisk-Users] How to configure NOKIA N70 with Asterisk?   Jens Vagelpohl
• SV: SV: [Asterisk-Users] How to configure NOKIA N70 with Asterisk?   Jens Vagelpohl
• [Asterisk-Users] How to configure NOKIA N70 with Asterisk?   Jens Vagelpohl
• [Asterisk-Users] Now that Nufone is dead...   Jens Vagelpohl
• [asterisk-users] OT: 3Com 3C10222 POE 24 Port Ethernet   Jared Valentine
• [Asterisk-Users] Mediatrix 1204 and Asterisk   Julian Varanini
• [Asterisk-Users] Mediatrix client authentication failed EFAILURE_REASON_AUTHENTICATION   Julian Varanini
• [asterisk-users] Got Mediatrix 1204 to work! now MWI and Polycom   Julian Varanini
• [asterisk-users] Got Mediatrix 1204 to work! now MWI and Polycom   Julian Varanini
• [asterisk-users] Help with MusicOnHold   Julian Varanini
• [asterisk-users] Help with MusicOnHold!!!   Julian Varanini
• [asterisk-users] Help with MusicOnHold   Julian Varanini
• [asterisk-users] Help with MusicOnHold   Julian Varanini
• [asterisk-users] Help with MusicOnHold!!!   Julian Varanini
• [asterisk-users] Help with MusicOnHold   Julian Varanini
• [asterisk-users] Mandriva 2006 Cooker RPM for Asterisk 1.2.9   Julian Varanini
• [asterisk-users] Mandriva 2006 Cooker RPM for Asterisk 1.2.9   Julian Varanini
• [asterisk-users] Mandriva 2006 Cooker RPM for Asterisk 1.2.9   Julian Varanini
• [asterisk-users] Mandriva 2006 Cooker RPM for Asterisk 1.2.9   Julian Varanini
• [asterisk-users] Mediatrix 1204 and Asterisk 1.2.9 stops working intermittently   Julian Varanini
• [asterisk-users] Asterisk 1.2.10 and Zaptel 1.2.7 released!   Julian Varanini
• [asterisk-users] Asterisk 1.2.10 and Zaptel 1.2.7 released!   Julian Varanini
• [asterisk-users] Polycom IP301 and Queues   Julian Varanini
• [asterisk-users] Voicemail and Polycom ip301   Julian Varanini
• [asterisk-users] PRI and Asterisk   Julian Varanini
• [asterisk-users] PRI and Asterisk   Julian Varanini
• [asterisk-users] Polycom IP301 and Queue questions, deployed environments   Julian Varanini
• [asterisk-users] VM integration to panasonic kx-td500   Miguel Ruiz Velasco
• [Asterisk-Users] IVR menus on different DIDs   Maxim Vexler
• [Asterisk-Users] s / i extension difficulty   Maxim Vexler
• [asterisk-users] Tired of fax calls... :-/   Maxim Vexler
• [asterisk-users] prob with debian and chan_zap   Maxim Vexler
• [asterisk-users] [tip]semicolon trouble: System($(sleep 4; cp 1.call out)&) not working, but System($( sleep 4 && cp 1.call out)&) ; )   Maxim Vexler
• 'cpmv' instead of 'cp' Re: [asterisk-users] [tip]semicolon trouble:   Maxim Vexler
• [asterisk-users] Asterisk + fax   Maxim Vexler
• [asterisk-users] How to create or test tone configuration to include them in zaptel   Maxim Vexler
• [asterisk-users] Setting a threshold for asterisk to take ZAP line off hook ?   Maxim Vexler
• [asterisk-users] Setting a threshold for asterisk to take ZAP line off hook ?   Maxim Vexler
• [asterisk-users] Setting a threshold for asterisk to take ZAP line off hook ?   Maxim Vexler
• [asterisk-users] Setting a threshold for asterisk to take ZAP line off hook ?   Maxim Vexler
• [asterisk-users] Zap channel faxing in or out fails but phone calls work.   Maxim Vexler
• [asterisk-users] PSTN disconnect tones on voicemail messages   Maxim Vexler
• [Asterisk-Users] Sangoma A200 woes   Christian Victor
• [asterisk-users] Unable to find good link to configure Polycom 501 with Asterisk (Plz send good link)   Tom Vile
• [asterisk-users] for you guys setting up customer offices...   Tom Vile
• [asterisk-users] Metermaid phone compatibility   Tom Vile
• [asterisk-users] CallerID   Tom Vile
• [asterisk-users] Re: Metermaid phone compatibility   Tom Vile
• [asterisk-users] Intercom mode on Polycom and/or SPA9xx   Tom Vile
• [asterisk-users] Intercom mode on Polycom and/or SPA9xx   Tom Vile
• [asterisk-users] How do you harden an Asterisk install?   Tom Vile
• [asterisk-users] How do you harden an Asterisk install?   Tom Vile
• [asterisk-users] Dlink DVG 1120S/Asterisk VoIP to PSTN   Tom Vile
• [asterisk-users] A very lost newbie.   Tom Vile
• [asterisk-users] How to connect 2 AAH   Tom Vile
• [asterisk-users] Grand stream 2000 will not dial *xx   Tom Vile
• [asterisk-users] Grand stream 2000 will not dial *xx   Tom Vile
• [asterisk-users] Source Directory of ASterisk   Tom Vile
• [asterisk-users] VoipNow 1.2.0 Beta   Tom Vile
• [asterisk-users] [Asteirsk-Users]TE110P configuration problem   Maximo Villamayor
• [asterisk-users] [Asteirsk-Users]TE110P configuration problem   Maximo Villamayor
• [Asterisk-Users] Cant seem to send cidname to snom 320   Terry Wade
• [asterisk-users] asterisk + nite affiliates   Terry Wade
• [asterisk-users] Asterisk version: 1.2.9.1 or older?   Terry Wade
• [asterisk-users] SMS functionality of bristuff (0.3.0-PRE-1r) with a Junghanns "duo GSM PCI" card   Chris Walker
• [asterisk-users] re:Simple But important question (for me)   Matthew Warren
• [asterisk-users] re:Simple But important question (for me)   Matthew Warren
• [asterisk-users] Re: Operator Console(s)/Shared Call Appearances   Matthew Warren
• [asterisk-users] Install H323   Wasif
• [asterisk-users] Getting no Audio with G729   Wasif
• [asterisk-users] RE: Getting no Audio with G729   Wasif
• [asterisk-users] Source Directory of ASterisk   Wasif
• [asterisk-users] No Audio   Wasif
• [asterisk-users] CFWD Status with PHP   Peter Wastl
• [Asterisk-Users] Asterisk -x option in 1.2.9.1   Watkins, Bradley
• [asterisk-users] Troubleshooting Random PRI disconnects   Watkins, Bradley
• [asterisk-users] Possible Bug?   Watkins, Bradley
• [asterisk-users] DUNDI / regcontext   Watkins, Bradley
• [asterisk-users] Testing 911?   Watkins, Bradley
• [asterisk-users] DUNDI / regcontext   Watkins, Bradley
• [asterisk-users] Redundant Ethernet   Watkins, Bradley
• [asterisk-users] Connecting Asterisk to a Metaswitch   Watkins, Bradley
• [asterisk-users] Caller ID on Transfers   Watkins, Bradley
• [asterisk-users] Caller ID on Transfers   Watkins, Bradley
• [asterisk-users] MWI from Octel to Asterisk   Watkins, Bradley
• [asterisk-users] Source Directory of ASterisk   Watkins, Bradley
• [asterisk-users] SNOM missed call.   Franklin Webb
• [asterisk-users] Issues with making Transfers   Ron Wellsted
• [asterisk-users] Asterisk Realtime Macros   Ron Wellsted
• [asterisk-users] Re: Recording/Monitor after xfer   Ron Wellsted
• [asterisk-users] Polycom, TFTP, and DHCP   Michael Welter
• [asterisk-users] Polycom, TFTP, and DHCP   Michael Welter
• [asterisk-users] AGI record_file   Michael Welter
• [Asterisk-Users] Queues and annoucements   BJ Weschke
• [Asterisk-Users] Libpri + Zaptel + Asterisk polycom_acd_functions error message   BJ Weschke
• [asterisk-users] Problem With Transfering Calls.   BJ Weschke
• [Asterisk-Users] Libpri + Zaptel + Asterisk polycom_acd_functions error message   BJ Weschke
• [asterisk-users] Polycom_acd_functions SIP trouble   BJ Weschke
• [asterisk-users] Operator Console(s)/Shared Call Appearances   BJ Weschke
• [asterisk-users] Play sounds to the callee and the callersynchronously when call begins   BJ Weschke
• [asterisk-users] Asterisk Polycom_acd_functions and G729   BJ Weschke
• [asterisk-users] Asterisk Polycom_acd_functions and G729   BJ Weschke
• [Asterisk-Users] HP Proliant server?   Derek Whitten
• [asterisk-users] NuFone suggests to use Vonage!!!!   Derek Whitten
• [asterisk-users] NuFone, please send the log file   Derek Whitten
• [asterisk-users] comcast info -- somewhat offtopic   Derek Whitten
• [asterisk-users] comcast info -- somewhat offtopic   Derek Whitten
• [asterisk-users] comcast info -- somewhat offtopic   Derek Whitten
• [asterisk-users] Re: $3,000 server   Derek Whitten
• [asterisk-users] zaptel on dual processor, How?   Derek Whitten
• [asterisk-users] need a pointer regarding scripting asterisk   Derek Whitten
• [Asterisk-Users] Polycom Soundpoint IP 301 w/ MGCP   Eric "ManxPower" Wieling
• [Asterisk-Users] Dial Macro timeout fails   Eric "ManxPower" Wieling
• [asterisk-users] test tone   Eric "ManxPower" Wieling
• [Asterisk-Users] Do you need a licence to connect a Ciscohardphone to Asterisk ?   Eric "ManxPower" Wieling
• [asterisk-users] FXS: No ringtone   Eric "ManxPower" Wieling
• [asterisk-users] Dialing timeouts   Eric "ManxPower" Wieling
• [asterisk-users] Text priority labels not working for me   Eric "ManxPower" Wieling
• [asterisk-users] Yet another problem with incoming SIP calls and 407   Eric "ManxPower" Wieling
• [asterisk-users] Polycom ACD, Asterisk, Kernel 2.6   Eric "ManxPower" Wieling
• [asterisk-users] Asterisk version: 1.2.9.1 or older?   Eric "ManxPower" Wieling
• [asterisk-users] Re: Asterisk and VAD   Eric "ManxPower" Wieling
• R: [asterisk-users] Called number on ISDN   Eric "ManxPower" Wieling
• [asterisk-users] Re: Wrong account code from iax_buddies   Eric "ManxPower" Wieling
• [asterisk-users] What is ZapRas used for ?   Eric "ManxPower" Wieling
• [Asterisk-Users] Question setting up a "bat phone" extension.   Eric "ManxPower" Wieling
• [asterisk-users] polycom 601 manual config?   Eric "ManxPower" Wieling
• [Asterisk-Users] Question setting up a "bat phone" extension.   Eric "ManxPower" Wieling
• [Asterisk-Users] Question setting up a "bat phone" extension.   Eric "ManxPower" Wieling
• [asterisk-users] Don't Hit # after 9 to get PSTN line   Eric "ManxPower" Wieling
• [asterisk-users] Don't Hit # after 9 to get PSTN line   Eric "ManxPower" Wieling
• [asterisk-users] Asterisk dead-air issues with Digium TE110P and IVR/meetme/internal directory-   Eric "ManxPower" Wieling
• [asterisk-users] Asterisk Dial Plan to Play Message   Eric "ManxPower" Wieling
• [asterisk-users] Asterisk Dial Plan to Play Message   Eric "ManxPower" Wieling
• [asterisk-users] Asterisk Dial Plan to Play Message   Eric "ManxPower" Wieling
• [asterisk-users] overlapdial and DID not always working   Eric "ManxPower" Wieling
• [asterisk-users] Regular expression problem   Eric "ManxPower" Wieling
• [asterisk-users] Voicemail not sent via email   Eric "ManxPower" Wieling
• [asterisk-users] Just bought a Polycom 501 - I feel like my GXP-2000 was better...   Eric "ManxPower" Wieling
• [asterisk-users] New message   Eric "ManxPower" Wieling
• [asterisk-users] Ringing timer   Eric "ManxPower" Wieling
• [asterisk-users] One way "screech" or tone   Eric "ManxPower" Wieling
• [asterisk-users] One way "screech" or tone   Eric "ManxPower" Wieling
• [asterisk-users] One way "screech" or tone   Eric "ManxPower" Wieling
• [asterisk-users] Re: '500 Internal Server' Error on SIP NOTIFY   James Willing
• [asterisk-users] X100P clone not working   Walter Willis
• [asterisk-users] fxo lines bridged on a new call once!   Jay Wilton
• [asterisk-users] Looking for carrier grade redundant solution   Stephen Wingfield
• [asterisk-users] sipbuddies realtime fields and latest documentation   Thomas Winter
• [asterisk-users] SIP user deny and permit for calls through Asterisk   Thomas Winter
• [asterisk-users] Server for Asterisk PCI   Thomas Winter
• [Asterisk-Users] How to continue after a match in an include   Ronald Wiplinger
• [Asterisk-Users] channel shows to be in use   Ronald Wiplinger
• [Asterisk-Users] time variable   Ronald Wiplinger
• [Asterisk-Users] I am looking for a (graphical) statistic program   Ronald Wiplinger
• [asterisk-users] ASTCC: how can I limit to xxx minutes per week?   Ronald Wiplinger
• [asterisk-users] ASTCC: inuse flag still hangs!   Ronald Wiplinger
• [asterisk-users] NuFone suggests to use Vonage!!!!   Ronald Wiplinger
• [asterisk-users] NuFone, please send the log file   Ronald Wiplinger
• [asterisk-users] NuFone, please send the log file   Ronald Wiplinger
• [asterisk-users] NuFone, please send the log file   Ronald Wiplinger
• [asterisk-users] NuFone, please send the log file   Ronald Wiplinger
• [asterisk-users] NuFone, please send the log file   Ronald Wiplinger
• [asterisk-users] NuFone, please send the log file   Ronald Wiplinger
• [asterisk-users] NuFone, please send the log file   Ronald Wiplinger
• [asterisk-users] unhappy-about-VoIP-providers@googlegroups.com founded   Ronald Wiplinger
• [asterisk-users] Zip code, city and area codes   Ronald Wiplinger
• [Asterisk-Users] Voicemail options   Kevin Withnall
• [Asterisk-Users] Voicemail options   Kevin Withnall
• [asterisk-users] SPA-3000 XML Config File   Kevin Withnall
• [asterisk-users] PRI dropouts   Kevin Withnall
• [asterisk-users] Alarmreceiver   Kevin Withnall
• [asterisk-users] PRI dropouts - solution I hope...   Kevin Withnall
• [asterisk-users] PRI dropouts - solution I hope...   Kevin Withnall
• [asterisk-users] FYI - first release of alarm response code.   Kevin Withnall
• [asterisk-users] Re: FYI - first release of alarm response code.   Kevin Withnall
• [asterisk-users] Dell PowerEdge 830   Hans Witvliet
• [Asterisk-Users] RE: Is there a search feature?   Tobias Wolf
• [asterisk-users] DUNDI / regcontext   Simon Woodhead
• [asterisk-users] DUNDI / regcontext   Simon Woodhead
• [asterisk-users] DUNDI / regcontext   Simon Woodhead
• [asterisk-users] regexten / Realtime WAS DUNDI / regcontext   Simon Woodhead
• [asterisk-users] Redundant Ethernet   Simon Woodhead
• [asterisk-users] New VEGASTREAM 400 Devices Cheap   Michael Workman
• [asterisk-users] NuFone, please send the log file   Michael Workman
• [asterisk-users] NuFone, please send the log file   Michael Workman
• [asterisk-users] NuFone, please send the log file   Michael Workman
• [asterisk-users] NuFone, please send the log file   Michael Workman
• [asterisk-users] NuFone, please send the log file   Michael Workman
• [asterisk-users] NuFone, please send the log file   Michael Workman
• [asterisk-users] Lets All Get Smart...   Michael Workman
• [asterisk-users] RE: $3,000 server   Michael Workman
• [asterisk-users] RE: $3,000 server   Michael Workman
• [asterisk-users] FW: $3,000 server   Michael Workman
• [asterisk-users] FW: $3,000 server   Michael Workman
• [asterisk-users] Re: $3,000 server   Michael Workman
• [asterisk-users] FW: $3,000 server   Michael Workman
• [asterisk-users] Phones cutting out.....again - PLEASE HELP!!!   Isaac Xiao
• [asterisk-users] Inconsistent call detail records   Tielin Xu
• [asterisk-users] Question on event AgentComplete of Manager API   Tielin Xu
• [asterisk-users] Manager interface   Tielin Xu
• [asterisk-users] Manager interface   Tielin Xu
• [asterisk-users] One way sound problem.   Tetsuya Yamamoto
• [asterisk-users] asterisk core dumps on a Sipura forwarded to a queue/moh   Vahan Yerkanian
• [asterisk-users] music ring (CRBT)   Peng Yong
• [asterisk-users] Alarmreceiver   Shane Young
• [asterisk-users] Zip code, city and area codes   Shane Young
• [asterisk-users] Do zttest results matter without telephony hardware?   Carl Youngblood
• [asterisk-users] Yate client   Stanislav Zahariev
• [asterisk-users] Having trouble to receive fax from samsung sf3200   Muhammad Zaka
• [asterisk-users] Help with MusicOnHold   Zeeshan Zakaria
• [asterisk-users] Encrypting the Conversation   Zeeshan Zakaria
• [asterisk-users] Which Fax Solution really works on IAX or SIP?   Zeeshan Zakaria
• [asterisk-users] Which Fax Solution really works on IAX or SIP?   Zeeshan Zakaria
• [asterisk-users] Choppy MOH (Cisco gateway)   Zeeshan Zakaria
• [asterisk-users] Tired of fax calls... :-/   Zeeshan Zakaria
• [asterisk-users] Sphinx and Asterisk Integration, How?   Zeeshan Zakaria
• [asterisk-users] zaptel on dual processor, How?   Zeeshan Zakaria
• [asterisk-users] zaptel on dual processor, How?   Zeeshan Zakaria
• [asterisk-users] zaptel on dual processor, How?   Zeeshan Zakaria
• [asterisk-users] Sphinx and Asterisk Integration, How?   Zeeshan Zakaria
• [asterisk-users] zaptel on dual processor, How?   Zeeshan Zakaria
• [asterisk-users] CentOS 4.3 and Zaptel-1.2.7   Zeeshan Zakaria
• [asterisk-users] Sphinx and Asterisk Integration, How?   Zeeshan Zakaria
• [asterisk-users] zapRAS   Ever Zalazar
• [asterisk-users] call forwarding   Ever Zalazar
• [asterisk-users] app-callforward   Ever Zalazar
• [asterisk-users] spandsp and app_*fax.c   Warrick Zedi
• [asterisk-users] Ringing timer   Zenone
• [asterisk-users] Ringing timer   Zenone
• [asterisk-users] Ringing timer   Zenone
• [asterisk-users] Ringing timer   Zenone
• [asterisk-users] Ringing timer   Michel Zenone
• [asterisk-users] Asterisk instances on VPS   Juergen K. Zick
• [asterisk-users] Sip voip call termination in Nigeria   Pele Zico
• [asterisk-users] Re: Sip voip call termination in Nigeria   Pele Zico
• [asterisk-users] Best practices with Asterisk   Pele Zico
• [asterisk-users] Re: Best practices with Asterisk   Pele Zico
• [asterisk-users] Bristuffed Asterisk: Hangup problems   Jeroen Zwarts
• [asterisk-users] Yet another problem with incoming SIP calls and 407   Wolfgang Zweimueller
• [asterisk-users] Yet another problem with incoming SIP calls and 407   Wolfgang Zweimueller
• [asterisk-users] Associate manager events to a previous Originate action   Johannes Zweng
• [asterisk-users] Associate manager events to a previous Originate action   Johannes Zweng
• [asterisk-users] Asterisk 1.2.10 and Zaptel 1.2.7 released!   Asterisk [Submusic]
• [asterisk-users] Can one SIP extension be used for two phones?   alunt2003 at googlemail.com
• [asterisk-users] Can one SIP extension be used for two phones?   alunt2003 at googlemail.com
• [asterisk-users] If you prefer to read this mail list as a forum ...   alunt2003 at googlemail.com
• [asterisk-users] ooh323c - cdr   antonio
• [asterisk-users] ooh323c - cdr problem   antonio
• [asterisk-users] I: ooh323c - cdr problem   antonio
• [asterisk-users] PRI got event: HDLC Abort (6) on Primary D-channel of span 1   asterisk
• [asterisk-users] Mitel 3300 + *   asterisk
• [asterisk-users] Mitel 3300 + *   asterisk
• [asterisk-users] SIP channel problem   asterisk
• [asterisk-users] Cisco 7960 - automated send DTMF digits after dialing?   asterisk at anime.net
• [asterisk-users] Cisco 7960 - automated send DTMF digits after dialing?   asterisk at anime.net
• [asterisk-users] Just bought a Polycom 501 - I feel likemy GXP-2000 was better...   asterisk at anime.net
• [asterisk-users] Just bought a Polycom 501 - I feel likemyGXP-2000 was better...   asterisk at anime.net
• [asterisk-users] can no more compile zaptel !!!   asterisk at frameweb.it
• [asterisk-users] can no more compile zaptel !!!   asterisk at frameweb.it
• [asterisk-users] PRI got event: HDLC Abort (6) on Primary D-channel of span 1 (fwd)   asterisk at nicox.org
• [asterisk-users] Double Ring on Asterisk 1.2.x (fwd)   asterisk at nicox.org
• [asterisk-users] Reload of wct4xxp without restarting of Asterisk?   asterisk at nicox.org
• [asterisk-users] Reload of wct4xxp without restarting of Asterisk?   asterisk at nicox.org
• [asterisk-users] Injecting prerecorded audio into active call   asterisk at nvworld.net
• [asterisk-users] Fonality vs TrixBox UI   asterisk at txpe.net
• [asterisk-users] Polycom 601 and Paging   asterisk at txpe.net
• [asterisk-users] Cyberdata paging speakers - anyone use them?   asterisk at txpe.net
• [asterisk-users] Caller ID on Transfers   asterisk at txpe.net
• [asterisk-users] Polycom 501 - How to set handset Volume   asterisk at txpe.net
• [asterisk-users] Please ignore ...   augustynr
• [asterisk-users] Feasability of using * for small appartment building?   augustynr
• [asterisk-users] Re: Feasability of using * for small appartment building?   augustynr
• [asterisk-users] Re: Feasability of using * for small appartment building?   augustynr
• [asterisk-users] Re: Feasability of using * for small appartmentbuilding?   augustynr
• [asterisk-users] Re: Feasability of using * for smallappartmentbuilding?   augustynr
• [asterisk-users] Re: Feasability of using * for small appartmentbuilding?   augustynr
• [asterisk-users] Re: Feasability of using * for smallappartmentbuilding?   augustynr
• [asterisk-users] If you prefer to read this mail list as a forum ...   augustynr
• [asterisk-users] 2 * servers, IAX connection between to dial extensions across IAX - not working   bails
• [asterisk-users] Can't load ztdummy   brettb at cytelcom.com
• [asterisk-users] need a pointer regarding scripting asterisk   shawn bright
• [asterisk-users] Re: need a pointer regarding scripting asterisk   shawn bright
• [asterisk-users] where to read stderr.out from an agi script   shawn bright
• [asterisk-users] where to read stderr.out from an agi script   shawn bright
• [Asterisk-Users] voip-magazine article "Using DUNDi with a Cluster of Asterisk Servers"   tijmen van den brink
• [Asterisk-Users] IAX2 failed to authenticate as priv (DUNDi)   tijmen van den brink
• [asterisk-users] Using DUNDi with TrixBox mini HOWTO   tijmen van den brink
• [asterisk-users] Using DUNDi with TrixBox mini HOWTO   tijmen van den brink
• [asterisk-users] Using DUNDi with TrixBox mini HOWTO   tijmen van den brink
• [asterisk-users] DUNDi 'Unable to Find Key'   tijmen van den brink
• [Asterisk-Users] Now that Nufone is dead...   broadbandvoice at comcast.net
• [Asterisk-Users] Best VoIP provider for Asterisk   broadbandvoice at comcast.net
• SV: [Asterisk-Users] Running 40 active calls (too much för CPU?)   broadbandvoice at comcast.net
• [asterisk-users] DTMF   broadbandvoice at comcast.net
• [asterisk-users] Phones cutting out.....again - PLEASE HELP!!!   broadbandvoice at comcast.net
• [asterisk-users] for you guys setting up customer offices...   broadbandvoice at comcast.net
• [asterisk-users] sip_poke_noanswer: Peer xxx is now unreachable   broadbandvoice at comcast.net
• [asterisk-users] Asterisk Dial Plan to Play Message   broadbandvoice at comcast.net
• [asterisk-users] Asterisk Dial Plan to Play Message   broadbandvoice at comcast.net
• [asterisk-users] Asterisk Dial Plan to Play Message   broadbandvoice at comcast.net
• [asterisk-users] Asterisk Dial Plan to Play Message   broadbandvoice at comcast.net
• [asterisk-users] Asterisk Dial Plan to Play Message   broadbandvoice at comcast.net
• [asterisk-users] Asterisk Dial Plan to Play Message   broadbandvoice at comcast.net
• [asterisk-users] ERROR 1045 (28000): Access denied for user   broadbandvoice at comcast.net
• [asterisk-users] Asterisk in Seattle   calvis
• [asterisk-users] for you guys setting up customer offices...   calvis
• [asterisk-users] OT: Project Management & Collaboration Software   calvis
• [asterisk-users] Polycom 501 - How to set handset Volume   calvis
• [asterisk-users] Zaptel Compilation Error   leonimar cape
• [asterisk-users] show channels   marek cervenka
• [asterisk-users] sip realtime   marek cervenka
• [asterisk-users] Call-limit and internal transfer   random cluster
• [asterisk-users] test to see if I can get any message through   John covici
• [asterisk-users] Injecting prerecorded audio into active call   John covici
• [Asterisk-Users] Need help with Junghanns Quadbri   Jean-Louis curty
• [asterisk-users] Cisco 7941/7961/7971 wont register with asterisk   dave
• [asterisk-users] Cisco SIP Firmware   dave
• [asterisk-users] Call-limit and internal transfer   alexandre - aldeia digital
• [asterisk-users] CDR related issue   dashy dude
• [asterisk-users] OFF-TRACK: Is VOIP -PSTN integration legal in China   dashy dude
• [asterisk-users] OFF-TRACK: Is VOIP -PSTN integration legal in China   dashy dude
• [asterisk-users] Hitting # to Transfer out of a Queue   end1r at comcast.net
• [asterisk-users] Hitting # to Transfer out of a Queue   end1r at comcast.net
• RE : [Asterisk-Users] Running 40 active calls (too much för CPU?)   f6hqz-m at hamwlan.net
• [asterisk-users] X100P clone not working   f6hqz-m at hamwlan.net
• [asterisk-users] Connecting branch offices through IPsec tunnel --latency effects?   f6hqz-m at hamwlan.net
• [Asterisk-Users] Hanging SIP Channels   asterisk fpf
• [asterisk-users] recompiling/updateing zaptel   sdgesa gaeharth
• [asterisk-users] Asterisk version: 1.2.9.1 or older?   garth at bitco.co.za
• [asterisk-users] Echo on PRI   garth at bitco.co.za
• [asterisk-users] Asterisk + fax   garth at bitco.co.za
• [asterisk-users] Asterisk + fax   al gav
• [asterisk-users] Asterisk + fax   al gav
• [asterisk-users] Asterisk + fax   al gav
• [Asterisk-Users] TDM Installation error   chawki hammoud
• [asterisk-users] Asterisk UAc / Request-URI   harrygaillac-sip at yahoo.fr
• [asterisk-users] Using outboundproxy in sip.conf   harrygaillac-sip at yahoo.fr
• [asterisk-users] Parked calls   harrygaillac-sip at yahoo.fr
• [asterisk-users] Parked calls   harrygaillac-sip at yahoo.fr
• [asterisk-users] Install H323   harrygaillac-sip at yahoo.fr
• [asterisk-users] Parked calls   harrygaillac-sip at yahoo.fr
• [asterisk-users] OH323 registration with gatekeeper problem   harrygaillac-sip at yahoo.fr
• [asterisk-users] asterisk-1.2.9 / chan-oh323.so   harrygaillac-sip at yahoo.fr
• SOLVED:RE : [asterisk-users] asterisk-1.2.9 / chan-oh323.so   harrygaillac-sip at yahoo.fr
• [asterisk-users] Asterisk and H.323   harrygaillac-sip at yahoo.fr
• [asterisk-users] Asterisk Manager Vb 6   hernany.ce
• [asterisk-users] Help troubleshooting "deadlocked" Asterisk   hugolivude
• [asterisk-users] Agent login problem with MP 124   imthiyaz at peopletech.co.in
• [asterisk-users] New GTK Gui for Monitoring and Administration   j
• [asterisk-users] New GTK Gui for Monitoring and Administration   j
• [asterisk-users] New GTK Gui for Monitoring and Administration   j
• [asterisk-users] asterisk + nite affiliates   j
• [asterisk-users] Asterisk version: 1.2.9.1 or older?   j
• [asterisk-users] Load balenced (ADSL) network connections, is it possible?   jacobso1
• [Asterisk-Users] Running 40 active calls (too much för CPU?)   jan.sarin at securia.se
• SV: [Asterisk-Users] Running 40 active calls (too much för CPU?)   jan.sarin at securia.se
• SV: [Asterisk-Users] Running 40 active calls (too much för CPU?)   jan.sarin at securia.se
• SV: SV: [Asterisk-Users] Running 40 active calls (too much för CPU?)   jan.sarin at securia.se
• SV: [Asterisk-Users] Running 40 active calls (too much för CPU?)   jan.sarin at securia.se
• [Asterisk-Users] HP Proliant server?   jan.sarin at securia.se
• SV: [Asterisk-Users] HP Proliant server?   jan.sarin at securia.se
• SV: [Asterisk-Users] HP Proliant server?   jan.sarin at securia.se
• [Asterisk-Users] Intel E7220 chipset?   jan.sarin at securia.se
• [asterisk-users] Queue hold position in other language?   jan.sarin at securia.se
• SV: [asterisk-users] Queue hold position in other language?   jan.sarin at securia.se
• [asterisk-users] Calls waiting announcement with two or more queues?   jan.sarin at securia.se
• [asterisk-users] Compiling zaptel on CentOS x86_64   jan.sarin at securia.se
• [asterisk-users] Got SUBSCRIBE for extensions without hint   jbauer at teec.com
• [asterisk-users] meetme issue with high cpu usage and "hung" conference rooms   jdutt at softelinc.com
• [asterisk-users] Tough time getting Polycom phones to register after router reboot   jdutt at softelinc.com
• [asterisk-users] Unable to find good link to configure Polycom 501 with Asterisk (Plz send good link)   jhill at asteriskit.com.au
• [asterisk-users] Sipura ATA's Forwarding PSTN Calls to Asterisk   john at cusick.ws
• [asterisk-users] Sipura ATA's Forwarding PSTN Calls to Asterisk   john at cusick.ws
• [asterisk-users] Asterisk with Linksys SPA-3000   john at cusick.ws
• [asterisk-users] How many users on an asterisk box behind a dsl can you have   ted jones
• [asterisk-users] Re: Asterisk 1.2.10 and Zaptel 1.2.7 released!   jwb at paravolve.net
• [asterisk-users] SIP and podcasts   kael
• [asterisk-users] SIP and podcasts   kael
• [asterisk-users] SIP and podcasts   kael
• [asterisk-users] long distance ethernet & Asterisk   kaze0010 at umn.edu
• [asterisk-users] Connecting Asterisk to a Metaswitch   kharris
• [asterisk-users] 2 NICs; Asterisk receives on eth1 and replies on eth0   kjcsb
• [asterisk-users] Voicemail volume patch   kjcsb
• [asterisk-users] Determining what gets written to the dst field for a CDR   kjcsb
• [asterisk-users] menu system - configurator   bram kortleven
• [asterisk-users] If you prefer to read this mail list as a forum ...   brandon kruz
• [asterisk-users] Invalid module format (ztdummy)   brandon kruz
• [asterisk-users] Error in ubuntu dapper   brandon kruz
• [asterisk-users] Error in ubuntu dapper   brandon kruz
• [asterisk-users] How to connect XLite with public IP?   brandon kruz
• [asterisk-users] NAT and externip problem or bug   brandon kruz
• [asterisk-users] Ztdummy   brandon kruz
• [asterisk-users] cannot received calls in pstn line   brandon kruz
• [asterisk-users] X100P clone not working   brandon kruz
• [asterisk-users] Ztdummy   brandon kruz
• [asterisk-users] X100P clone not working   brandon kruz
• [asterisk-users] X100P clone not working   brandon kruz
• [asterisk-users] newbbie question   brandon kruz
• [asterisk-users] Asterisk Dial Plan to Play Message   brandon kruz
• [asterisk-users] newbbie question   brandon kruz
• [asterisk-users] Asterisk Dial Plan to Play Message   brandon kruz
• [asterisk-users] Asterisk Dial Plan to Play Message   brandon kruz
• [asterisk-users] sip.conf, extensions.conf   ashok kumar
• [Asterisk-Users] Queues and annoucements   lenz
• [asterisk-users] Queue Persistence with queue.log   lenz
• [Asterisk-Users] Does asterisk support outbound fax?   root linux
• [Asterisk-Users] Does asterisk support outbound fax?   root linux
• [asterisk-users] [Asteirsk-Users]TE110P configuration problem   root linux
• [asterisk-users] sending out fax using asterisk   root linux
• [asterisk-users] sending out fax using asterisk   root linux
• [asterisk-users] kernel: Error! while loading wct4xxp module   root linux
• [asterisk-users] Possible Bug?   Warren (mailing lists)
• [asterisk-users] Sip voip call termination in Nigeria   Warren (mailing lists)
• [asterisk-users] comcast info -- somewhat offtopic   Warren (mailing lists)
• [asterisk-users] New VEGASTREAM 400 Devices Cheap   Warren (mailing lists)
• [asterisk-users] NuFone, please send the log file   Warren (mailing lists)
• [asterisk-users] New York city Asterisk consultants   Warren (mailing lists)
• [asterisk-users] Asterisk version: 1.2.9.1 or older?   Warren (mailing lists)
• [asterisk-users] How do you harden an Asterisk install?   Warren (mailing lists)
• [asterisk-users] How do you harden an Asterisk install?   Warren (mailing lists)
• [asterisk-users] Can incoming alternate rings be discriminated?   Warren (mailing lists)
• [asterisk-users] How do you harden an Asterisk install?   Warren (mailing lists)
• [asterisk-users] astbill + mysql 5   Warren (mailing lists)
• [asterisk-users] Can incoming alternate rings be discriminated?   Warren (mailing lists)
• [asterisk-users] Asterisk 1.2.10 and Zaptel 1.2.7 released!   Warren (mailing lists)
• [asterisk-users] zaptel on dual processor, How?   Warren (mailing lists)
• [asterisk-users] zaptel on dual processor, How?   Warren (mailing lists)
• [asterisk-users] Re: Polycom compatible phone for Asterisk   andrew matthews
• [asterisk-users] termcap support not found   mattwm at chairo.vic.edu.au
• [asterisk-users] RE: Cisco Gateway & CallerID Name   mavince at optonline.net
• [asterisk-users] OT: Project Management & Collaboration Software   mawali at news.icns.com
• [Asterisk-Users] Help with JIAXClient   me me
• [Asterisk-Users] Trouble Setting Up International Dialing in extensions.conf   methodvon
• [Asterisk-Users] Trouble Setting Up International Dialing in extensions.conf   methodvon
• [asterisk-users] sip codec convertion on the fly   mike
• [asterisk-users] sip codec convertion on the fly   mike
• [asterisk-users] asterisk and sip nat problems   mike
• [asterisk-users] asterisk and sip nat problems   mike
• [asterisk-users] asterisk and sip nat problems   mike
• [asterisk-users] qozap w/ 1.2.9.1   mike
• [asterisk-users] Zap Channel not hanging up on Telco side   mike
• [asterisk-users] Zap Channel not hanging up on Telco side   mike
• [asterisk-users] qozap w/ 1.2.9.1   mike
• [asterisk-users] New GTK Gui for Monitoring and Administration   mike
• [asterisk-users] Choppy MOH (Cisco gateway)   mike
• [asterisk-users] Rate or rank ITSP   mike
• [asterisk-users] Rate or rank ITSP   mike
• [asterisk-users] Injecting prerecorded audio into active call   mike
• [Asterisk-Users] Putting a call recording into a mailbox   mitcheloc at gmail.com
• [asterisk-users] Manager interface   mitcheloc at gmail.com
• [asterisk-users] prob with debian and chan_zap   nephish
• [asterisk-users] prob with debian and chan_zap   nephish
• [asterisk-users] prob with debian and chan_zap   nephish
• [asterisk-users] prob with debian and chan_zap   nephish
• [asterisk-users] prob with debian and chan_zap   nephish
• [asterisk-users] need a pointer about scripting asterisk   nephish
• [asterisk-users] need a pointer about scripting asterisk   nephish
• [asterisk-users] SIP account not available in queue ringall   nik600
• [Asterisk-Users] Meetme versus app_conference   jeff oconnell
• [Asterisk-Users] app_conference DTMFs?   jeff oconnell
• [Asterisk-Users] app_conference DTMFs?   jeff oconnell
• [Asterisk-Users] app_conference DTMFs?   jeff oconnell
• [Asterisk-Users] app_conference DTMFs?   jeff oconnell
• [Asterisk-Users] app_conference DTMFs?   jeff oconnell
• [Asterisk-Users] app_conference DTMFs?   jeff oconnell
• [Asterisk-Users] app_conference DTMFs?   jeff oconnell
• [asterisk-users] intel vs amd motherboards   olivier.taylor
• [asterisk-users] intel vs amd motherboards   olivier.taylor
• [asterisk-users] [Fwd: where is the error?]   olivier.taylor
• [asterisk-users] [Fwd: where is the error?]   olivier.taylor
• [asterisk-users] Germany VOIP provider   olivier.taylor
• [asterisk-users] Germany VOIP provider   olivier.taylor
• [asterisk-users] Codec Negotiation   olivier.taylor
• [asterisk-users] Play sound to called party ...   phil.dawson at marnock.com
• [asterisk-users] Polycom (Do Not Disturb)   phil.dawson at marnock.com
• [asterisk-users] Load balenced (ADSL) network connections, is it possible?   phil.dawson at marnock.com
• [asterisk-users] Rockwell Modem   ram
• [asterisk-users] Rockwell Modem   ram
• [asterisk-users] Please suggest me Best VoIP Service Provider   ram
• [asterisk-users] Astrisks compatable cards   ram
• [asterisk-users] CDRTools please help   ravi reddy
• [asterisk-users] CSTA support for asterisk   Vincent renaville
• [asterisk-users] SIP Woes   tim robinson
• [asterisk-users] Re: Clearing variables in the dialplan?   rushowr at phreaker.net
• [asterisk-users] don't hear start/begin of voiceprompts   rushowr at phreaker.net
• [asterisk-users] don't hear start/begin of voiceprompts   rushowr at phreaker.net
• [asterisk-users] Load balenced (ADSL) network connections, is it possible?   rushowr at phreaker.net
• [asterisk-users] Load balenced (ADSL) network connections, is it possible?   rushowr at phreaker.net
• [asterisk-users] OT: Project Management & Collaboration Software   rushowr at phreaker.net
• [asterisk-users] Fwd: Problem with chan_zap.so   ismir saljic
• [asterisk-users] CSTA support for asterisk   sanchal.singh at alliance-infotech.com
• [asterisk-users] CSTA support for asterisk   sanchal.singh at alliance-infotech.com
• [asterisk-users] CSTA support for astersik   sanchal.singh at alliance-infotech.com
• [asterisk-users] asterisk with CSTA using VAIL SIP TIM   sanchal.singh at alliance-infotech.com
• [Asterisk-Users] voip-magazine article "Using DUNDi with aCluster of Asterisk Servers"   scott
• [Asterisk-Users] voip-magazine article "Using DUNDi with aClusterof Asterisk Servers"   scott
• [Asterisk-Users] voip-magazine article "Using DUNDi withaClusterof Asterisk Servers"   scott
• [Asterisk-Users] voip-magazine article "Using DUNDiwithaClusterof Asterisk Servers"   scott
• [Asterisk-Users] voip-magazine article "Using DUNDiwithaClusterofAsterisk Servers"   scott
• [Asterisk-Users] voip-magazine article "UsingDUNDiwithaClusterofAsterisk Servers"   scott
• [Asterisk-Users] voip-magazine article"UsingDUNDiwithaClusterofAsterisk Servers"   scott
• [Asterisk-Users] voip-magazinearticle"UsingDUNDiwithaClusterofAsterisk Servers"   scott
• [Asterisk-Users]voip-magazinearticle"UsingDUNDiwithaClusterofAsterisk Servers"   scott
• [Asterisk-Users] Does asterisk support outbound fax?   scott
• [Asterisk-Users]voip-magazinearticle"UsingDUNDiwithaClusterofAsteriskServers"   scott
• [Asterisk-Users] SOLVED: IAX jitter / clocking problem   scott
• [Asterisk-Users] Does asterisk support outbound fax?   scott
• [Asterisk-Users]voip-magazinearticle"UsingDUNDiwithaClusterofAsteriskServers"   scott
• [Asterisk-Users] SOLVED: IAX jitter / clocking problem   scott
• [Asterisk-Users] Does asterisk support outbound fax?   scott
• [Asterisk-Users]voip-magazinearticle"UsingDUNDiwithaClusterofAsteriskServers"   scott
• [Asterisk-Users] SOLVED: IAX jitter / clocking problem   scott
• [Asterisk-Users] Does asterisk support outbound fax?   scott
• [Asterisk-Users]voip-magazinearticle"UsingDUNDiwithaClusterofAsteriskServers"   scott
• [Asterisk-Users] SOLVED: IAX jitter / clocking problem   scott
• [Asterisk-Users] Does asterisk support outbound fax?   scott
• [Asterisk-Users]voip-magazinearticle"UsingDUNDiwithaClusterofAsteriskServers"   scott
• [Asterisk-Users] SOLVED: IAX jitter / clocking problem   scott
• [Asterisk-Users] Does asterisk support outbound fax?   scott
• [Asterisk-Users] Mediatrix client authentication failedEFAILURE_REASON_AUTHENTICATION   scott
• [Asterisk-Users]voip-magazinearticle"UsingDUNDiwithaClusterofAsteriskServers"   scott
• [Asterisk-Users] SOLVED: IAX jitter / clocking problem   scott
• [Asterisk-Users] time variable   scott
• [Asterisk-Users] Does asterisk support outbound fax?   scott
• [Asterisk-Users] Mediatrix client authenticationfailedEFAILURE_REASON_AUTHENTICATION   scott
• [Asterisk-Users]voip-magazinearticle"UsingDUNDiwithaClusterofAsteriskServers"   scott
• [Asterisk-Users] SOLVED: IAX jitter / clocking problem   scott
• [Asterisk-Users] time variable   scott
• [Asterisk-Users] Does asterisk support outbound fax?   scott
• [Asterisk-Users] Mediatrix clientauthenticationfailedEFAILURE_REASON_AUTHENTICATION   scott
• [Asterisk-Users]voip-magazinearticle"UsingDUNDiwithaClusterofAsteriskServers"   scott
• [Asterisk-Users] SOLVED: IAX jitter / clocking problem   scott
• [Asterisk-Users] Re: Digium Hardware Reliability   scott
• [Asterisk-Users] time variable   scott
• [Asterisk-Users] Does asterisk support outbound fax?   scott
• [Asterisk-Users] MediatrixclientauthenticationfailedEFAILURE_REASON_AUTHENTICATION   scott
• [Asterisk-Users]voip-magazinearticle"UsingDUNDiwithaClusterofAsteriskServers"   scott
• [Asterisk-Users] SOLVED: IAX jitter / clocking problem   scott
• [Asterisk-Users] Re: Digium Hardware Reliability   scott
• [Asterisk-Users] time variable   scott
• [Asterisk-Users] Does asterisk support outbound fax?   scott
• [Asterisk-Users]MediatrixclientauthenticationfailedEFAILURE_REASON_AUTHENTICATION   scott
• [Asterisk-Users]voip-magazinearticle"UsingDUNDiwithaClusterofAsteriskServers"   scott
• [Asterisk-Users] SOLVED: IAX jitter / clocking problem   scott
• [Asterisk-Users] Re: Digium Hardware Reliability   scott
• [Asterisk-Users] time variable   scott
• [Asterisk-Users] Does asterisk support outbound fax?   scott
• [Asterisk-Users]MediatrixclientauthenticationfailedEFAILURE_REASON_AUTHENTICATION   scott
• [Asterisk-Users]voip-magazinearticle"UsingDUNDiwithaClusterofAsteriskServers"   scott
• [Asterisk-Users] SOLVED: IAX jitter / clocking problem   scott
• [Asterisk-Users] Re: Digium Hardware Reliability   scott
• [Asterisk-Users] time variable   scott
• [Asterisk-Users] Does asterisk support outbound fax?   scott
• [Asterisk-Users] vserver with no /dev/tty* how to run "asterisk -c"for a colored CLI?   scott
• [Asterisk-Users]MediatrixclientauthenticationfailedEFAILURE_REASON_AUTHENTICATION   scott
• [Asterisk-Users]MediatrixclientauthenticationfailedEFAILURE_REASON_AUTHENTICATION   scott
• [Asterisk-Users]voip-magazinearticle"UsingDUNDiwithaClusterofAsteriskServers"   scott
• [Asterisk-Users] SOLVED: IAX jitter / clocking problem   scott
• [Asterisk-Users] Re: Digium Hardware Reliability   scott
• [Asterisk-Users] time variable   scott
• [Asterisk-Users] Does asterisk support outbound fax?   scott
• [Asterisk-Users] vserver with no /dev/tty* how to run "asterisk-c"for a colored CLI?   scott
• [Asterisk-Users]MediatrixclientauthenticationfailedEFAILURE_REASON_AUTHENTICATION   scott
• [Asterisk-Users] Page() command and file playback   scott
• [Asterisk-Users]MediatrixclientauthenticationfailedEFAILURE_REASON_AUTHENTICATION   scott
• [asterisk-users] ATCOM's AG-188   sdcharly at gmail.com
• [asterisk-users] IAX ATA with FXO   sdcharly at gmail.com
• [asterisk-users] intel vs amd motherboards   shadowym
• [asterisk-users] Metermaid phone compatibility   shadowym
• [asterisk-users] Re: for you guys setting up customer offices...   shadowym
• [asterisk-users] Re: Metermaid phone compatibility   shadowym
• [asterisk-users] How do you harden an Asterisk install?   shadowym
• [asterisk-users] How do you harden an Asterisk install?   shadowym
• [asterisk-users] How do you harden an Asterisk install?   shadowym
• [asterisk-users] Hardware/Software suggestions, Supermicro 6024H-TR?   shadowym
• [asterisk-users] emulating key system - pick up so and so on line1   shadowym
• [asterisk-users] Redundant Ethernet   shadowym
• [asterisk-users] Operator Console(s)/Shared Call Appearances   shadowym
• [asterisk-users] Just bought a Polycom 501 - I feel like myGXP-2000 was better...   shadowym
• [asterisk-users] Just bought a Polycom 501 - I feellike myGXP-2000 was better...   shadowym
• [asterisk-users] Sangoma Stops Receiving Calls   shadowym
• [asterisk-users] SIP To: header   sip
• [asterisk-users] Germany VOIP provider   sip
• [asterisk-users] Cisco 7912 IP Phone - Convert SIP to SCCP   sipResearcher
• [asterisk-users] Play sounds to the callee and the caller synchronously when call begins   snlee
• [asterisk-users] SPA-3000 XML Config File   soren at silikone.com
• [Asterisk-Users] Need help with Junghanns Quadbri   stoffell
• [asterisk-users] Zap Channel not hanging up on Telco side   stoffell
• [asterisk-users] qozap w/ 1.2.9.1   stoffell
• [asterisk-users] Redundant Ethernet   stoffell
• [asterisk-users] CentOS 4.3 and Zaptel-1.2.7   tracinet
• [Asterisk-Users] What does it mean?   unplug
• [asterisk-users] control during registration process   unplug
• [asterisk-users] control during registration process   unplug
• [asterisk-users] Server redundancy   unplug
• [asterisk-users] Server redundancy   unplug
• [asterisk-users] Server redundancy   unplug
• [asterisk-users] register process flow   unplug
• [asterisk-users] Server redundancy   unplug
• [asterisk-users] dial plan -- help   unplug
• [asterisk-users] priority problem   unplug
• [asterisk-users] priority problem   unplug
• [asterisk-users] priority problem   unplug
• [asterisk-users] priority problem   unplug
• [asterisk-users] priority problem   unplug
• [asterisk-users] question ast db   unplug
• [asterisk-users] priority problem   unplug
• [asterisk-users] usage of ast db   unplug
• [asterisk-users] header replacement   unplug
• [asterisk-users] header replacement   unplug
• [asterisk-users] header replacement   unplug
• [asterisk-users] question about function realtime   unplug
• [asterisk-users] asterisk database   unplug
• [asterisk-users] all call forward   unplug
• [asterisk-users] asterisk database   unplug
• [asterisk-users] realtime function   unplug
• [asterisk-users] asterisk database   unplug
• [asterisk-users] question about asterisk DB   unplug
• [asterisk-users] question about asterisk DB   unplug
• [asterisk-users] question about asterisk DB   unplug
• [asterisk-users] question about asterisk DB   unplug
• [asterisk-users] registration process   unplug
• [asterisk-users] registration process   unplug
• [asterisk-users] registration process   unplug
• [asterisk-users] intel vs amd motherboards   varun
• [asterisk-users] intel vs amd motherboards   varun
• [asterisk-users] intel vs amd motherboards   varun
• [asterisk-users] intel vs amd motherboards   varun
• [asterisk-users] intel vs amd motherboards   varun
• [asterisk-users] astbill + mysql 5   varun
• [asterisk-users] asterisk + centos 4.3   varun
• [asterisk-users] asterisk + centos 4.3   varun
• [asterisk-users] astbill + mysql 5   varun
• [asterisk-users] CentOS 4.3 and Zaptel-1.2.7   varun
• [asterisk-users] Asterisk + fax   ven
• [asterisk-users] Asterisk + fax   ven
• [asterisk-users] asterisk 1.4 download   vivek at staff.ownmail.com
• [asterisk-users] app background   vivek at staff.ownmail.com
• [asterisk-users] setting call-limits   voip at gmx-topmail.de
• [asterisk-users] setting call-limits   voip at gmx-topmail.de
• [asterisk-users] setting call-limits   voip at gmx-topmail.de
• [Asterisk-Users] More g729 calls than licenses?   voiplist
• [Asterisk-Users] More g729 calls than licenses?   voiplist
• [asterisk-users] Mutiple Homes one asterisk box   voiplist
• [asterisk-users] Wrong account code from iax_buddies   voiplist
• [asterisk-users] Re: Wrong account code from iax_buddies   voiplist
• [asterisk-users] Re: Wrong account code from iax_buddies   voiplist
• [asterisk-users] Re: Wrong account code from iax_buddies   voiplist
• [asterisk-users] Re: Wrong account code from iax_buddies   voiplist
• [asterisk-users] Re: Wrong account code from iax_buddies   voiplist
• [asterisk-users] Re: Wrong account code from iax_buddies   voiplist
• [asterisk-users] Testing 911?   voiplist
• [asterisk-users] LinkSys SPA 2002 ATA hardphone UNREACHABLE...!!!   voiplist
• [asterisk-users] LinkSys SPA 2002 ATA hardphone UNREACHABLE...!!!   voiplist
• [asterisk-users] SPA-2000, Asterisk 1.2.4 & Incoming call success? Anyone?   voiplist
• [asterisk-users] Keep Zap Channel from answering   voiplist
• [asterisk-users] Two phone numbers, one SIP provider   voiplist
• [asterisk-users] Rate engine AGI?   voiplist
• [asterisk-users] Rate engine AGI?   voiplist
• [Asterisk-Users] callwaiting in queues   whois wes
• [Asterisk-Users] Help with IVR menu.   whois wes
• [Asterisk-Users] Dial Macro timeout fails   whois wes
• [Asterisk-Users] Dial Macro timeout fails   whois wes
• [asterisk-users] Phones cutting out.....again - PLEASE HELP!!!   whois wes
• [asterisk-users] Phones cutting out.....again - PLEASE HELP!!!   whois wes
• [asterisk-users] Phones cutting out.....again - PLEASE HELP!! !   whois wes
• [asterisk-users] Phones cutting out.....again - PLEASE HELP!! !   whois wes
• [asterisk-users] Server locking up again   whois wes
• [asterisk-users] setting call-limits   whois wes
• [asterisk-users] 2 * servers, IAX connection between to dial extensions across IAX - not working   whois wes
• [asterisk-users] 2 * servers, IAX connection between to dial extensions across IAX - not working   whois wes
• [asterisk-users] dropping calls in the middle of conversation   whois wes
• [Asterisk-Users] H323 Asterisk best practices   yusuf
• [asterisk-users] AGI: Channel status   yusuf
• [asterisk-users] FXS: No ringtone   yusuf
• [asterisk-users] [Fwd: [Fwd: polarityswitch: no ringback]]   yusuf
• [asterisk-users] Voismart GSM - no billsecs   yusuf
• [asterisk-users] Voismart GSM - no billsecs   yusuf
• [asterisk-users] Server Optimization and Load Balancing   zoa
• [asterisk-users] If you prefer to read this mail list as a forum ...   zoa
• [asterisk-users] LinkSys SPA 2002 ATA hardphone UNREACHABLE...!!!   | Rurouni Alucard |