March 2006 Archives by author

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Starting: Wed Mar 1 00:13:13 MST 2006
Ending: Fri Mar 31 23:52:58 MST 2006
Messages: 4543

• [Asterisk-Users] [Fwd: Over 40 destinations for FREE!]   Francesco Peeters (Asterisk)
• [Asterisk-Users] [Fwd: Over 40 destinations for FREE!]   Francesco Peeters (Asterisk)
• [Asterisk-Users] IAX choppy sound   Francesco Peeters (Asterisk)
• [Asterisk-Users] G.729 codec licencing   Francesco Peeters (Asterisk)
• [Asterisk-Users] Zap<-->IAX codec?   Francesco Peeters (Asterisk)
• [Asterisk-Users] Zap<-->IAX codec?   Francesco Peeters (Asterisk)
• [Asterisk-Users] Call terminated after 60 seconds   Francesco Peeters (Asterisk)
• [Asterisk-Users] Junghanns, Germany ISDN settings   Chris Earle (CBL)
• [Asterisk-Users] Junghanns, Germany ISDN settings   Chris Earle (CBL)
• [Asterisk-Users] module load order for Junghanns qozap and TDM card   Chris Earle (CBL)
• [Asterisk-Users] module load order for Junghanns qozap and TDM card   Chris Earle (CBL)
• [Asterisk-Users] module load order for Junghanns qozap and TDMcard   Chris Earle (CBL)
• [Asterisk-Users] qozap drops -- possible to bridge BRIstuff ISDN to analog zaptel phone?   Chris Earle (CBL)
• [Asterisk-Users] Junghanns and Digium TDM400?   Chris Earle (CBL)
• [Asterisk-Users] pickup problem   Chris Earle (CBL)
• [Asterisk-Users] zapata configuration & parsing   David Cook (Canada)
• [Asterisk-Users] Get no busy signal on my analog line   Andre Rodrigues (Cheyenne)
• [Asterisk-Users] new beta Grandstream firmware HT488_496_386   Andre Rodrigues (Cheyenne)
• [Asterisk-Users] new beta Grandstream firmware HT488_496_386   Andre Rodrigues (Cheyenne)
• [Asterisk-Users] new beta Grandstream firmware HT488_496_386   Andre Rodrigues (Cheyenne)
• [Asterisk-Users] new beta Grandstream firmware HT488_496_386   Andre Rodrigues (Cheyenne)
• {Filename?} [Asterisk-Users] hotel vmail and iax trouble   Chris Mason (Lists)
• [Asterisk-Users] Call go on hold for no reason   Chris Mason (Lists)
• [Asterisk-Users] Sipura 3000 DMTF   Chris Mason (Lists)
• [Asterisk-Users] An FXO version of IAXy?   Chris Mason (Lists)
• [Asterisk-Users] Bizzare DTMF on channel bank   Chris Mason (Lists)
• [Asterisk-Users] g729 and latency measures   Chris Mason (Lists)
• [Asterisk-Users] Zaptel will not build   Chris Mason (Lists)
• [Asterisk-Users] ADT/Brinks alarm dialing through Asterisk   Chris Mason (Lists)
• [Asterisk-Users] ADT/Brinks alarm dialing through Asterisk   Chris Mason (Lists)
• [Asterisk-Users] Billing from CDR files   Chris Mason (Lists)
• [Asterisk-Users] FXS channel banks   Chris Mason (Lists)
• [Asterisk-Users] FXS channel banks   Chris Mason (Lists)
• [Asterisk-Users] Asterisk billing from CDR database   Chris Mason (Lists)
• [Asterisk-Users] Asterisk billing from CDR database   Chris Mason (Lists)
• [Asterisk-Users] tsu-600   Chris Mason (Lists)
• [Asterisk-Users] Dropped calls   Chris Mason (Lists)
• [Asterisk-Users] Best budget IP phone at the moment?   Bill McCready (PCPhoneline.com)
• [Asterisk-Users] OT: Snom 320, displaying text on the screen from *   Sven Fischer (support)
• [Asterisk-Users] problems compiling zaptel on FC5   Raul Elizondo (wizardteam)
• [Asterisk-Users] Re: One Extension - Two Calls?   Bromont -
• [Asterisk-Users] Asterisk--->>Autodialling   Sheeju .R.Alex
• [Asterisk-Users] Frequently Showed Info Messages   ITN Info - 11-3898-0112
• [Asterisk-Users] Incoming Asterisk SIP DID Calls   ITN Info - 11-3898-0112
• [Asterisk-Users] ISDN Protocol Unknom Error with Junghanns OctoBRI   Sébastien Mortier
• [Asterisk-Users] Information to program a new driver for Asterisk   Álvaro Palma
• [Asterisk-Users] Changing REINVITE status of the channel dynamically   Álvaro Palma
• [Asterisk-Users] Upgrading Asterisk witk G729 license installed   Álvaro Palma
• [Asterisk-Users] Extracting info from the $EXTEN variable   Álvaro Palma
• [Asterisk-Users] Speeding up the dial of DTMF's in SIP channel   Álvaro Palma
• [Asterisk-Users] Problem with INVITE's being sent   Álvaro Palma
• [Asterisk-Users] Strange second REINVITE being sent   Álvaro Palma
• [Asterisk-Users] ChanSpy   Adrià Vidal
• [Asterisk-Users] G729 License questions   Adrià Vidal
• [Asterisk-Users] simple perl-agi - where's the error?   Benoît Mérouze
• [Asterisk-Users] Which Mac OSX softphone with IAX2 support?   Benoît Mérouze
• [Asterisk-Users] Re: Get no busy signal on my analog line   Bruno de Assumpção Loureiro
• [Asterisk-Users] Softphone accepting URL   Bruno de Assumpção Loureiro
• [Asterisk-Users] Reporting?   Bruno de Assumpção Loureiro
• [Asterisk-Users] mISDN Problem   Christian Gröger
• [Asterisk-Users] Polycom IP 301 is slow   Denis Galvão - iSolve
• [Asterisk-Users] Who hangup.   José Luis Gómez
• [Asterisk-Users] Problems with wcte11xp module   Josué Conti
• [Asterisk-Users] IAX problems - please help me   Josué Conti
• [Asterisk-Users] IAX problems - please help me   Josué Conti
• [Asterisk-Users] Squished faxes with txfax   Josué Conti
• [Asterisk-Users] IAX2 errors   Josué Conti
• [Asterisk-Users] Zap channels - help   Josué Conti
• [Asterisk-Users] Help! Connecting two Astersik via SIP channels   María Chóliz
• [Asterisk-Users] Using IAX   María Chóliz
• [Asterisk-Users] Timeout waiting for response to Originate   María Chóliz
• [Asterisk-Users] CDR question   Marc Patino Gómez
• [Asterisk-Users] CDR question   Marc Patino Gómez
• [Asterisk-Users] SPA3000 and callerID   Mickaël Cissé
• [Asterisk-Users] pickup a call in queue   Nicolás Gudiño
• [Asterisk-Users] OT: FOP and reverse_transfer   Nicolás Gudiño
• [Asterisk-Users] Call Monitoring / Call Takeover with Asterisk   Nicolás Gudiño
• [Asterisk-Users] Reporting?   Nicolás Gudiño
• [Asterisk-Users] Dial from php   Nicolás Gudiño
• [Asterisk-Users] Reporting?   Nicolás Gudiño
• [Asterisk-Users] Menu in queue   Poul Møller Hansen
• [Asterisk-Users] Menu in queue   Poul Møller Hansen
• [Asterisk-Users] ISDN Protocol Unknom Error with Junghanns OctoBRI   Sébastien Mortier
• [Asterisk-Users] gotoiftime with list of time range   Vincent Régnard
• [Asterisk-Users] Re: BRI cards, HFC, and bristuff - a general questionto clear up my understanding.   Filip Drągowski
• [Asterisk-Users] Howto cut the first digit   Filip Drągowski
• [Asterisk-Users] Re: BRI cards, HFC, and bristuff - a generalquestionto clear up my understanding.   Filip Drągowski
• [Asterisk-Users] I have debug off why are the logs show debug info   Filip Drągowski
• [Asterisk-Users] Video phone failed on Asterisk-1.2.4   蘇玉華
• [Asterisk-Users] asterisk and DDI   René Enskat [Teamware GmbH]
• AW: [Asterisk-Users] simple perl-agi - where's the error?   René Enskat [Teamware GmbH]
• [Asterisk-Users] Realtime mapping problem after svn upgrade   René Enskat [Teamware GmbH]
• SV: [Asterisk-Users] re: Sound issues on SIP-SIP calls   Bjørn O
• SV: [Asterisk-Users] re: Sound issues on SIP-SIP calls   Bjørn O
• SV: [Asterisk-Users] IAX - only one way traffic   Bjørn O
• SV: [Asterisk-Users] IAX - only one way traffic   Bjørn O
• [Asterisk-Users] Polling Asterisk for Life   Håkan Källberg
• [Asterisk-Users] Thinking of moving from pure VoIP to PRI - thoughts?   Michaël Gaudette
• [Asterisk-Users] Limiting the number of concurrent calls for a group of SIP devices   Michaël Gaudette
• [Asterisk-Users] Unable to make hints function properly   Per Møller
• SV: [Asterisk-Users] Unable to make hints function properly   Per Møller
• [Asterisk-Users] SMS in Spain (it seems Protocol 2)   Sergio García Murillo
• [Asterisk-Users] Asterisk sip and radius authentication   Sergio Iñigo Ibáñez
• [Asterisk-Users] Help: Using asterisk and mysql for a university project   Sergio Iñigo Ibáñez
• [Asterisk-Users] Caller ID forwarding with Pickup() application?   Tamás Bondár
• [Asterisk-Users] automatic callback when busy   Tamás Bondár
• [Asterisk-Users] automatic callback when busy   Tamás Bondár
• [Asterisk-Users] automatic callback when busy   Tamás Bondár
• [Asterisk-Users] ooh323 codec's - alaw   Tomislav Parčina
• [Asterisk-Users] Re: How to check if transcoding is setup to work properly   Tomislav Parčina
• [Asterisk-Users] Cisco 7905 - vad, cng   Tomislav Parčina
• [Asterisk-Users] MOH native files   Tomislav Parčina
• [Asterisk-Users] Info about F1000G   Tomislav Parčina
• [Asterisk-Users] Re: MOH native files   Tomislav Parčina
• [Asterisk-Users] Re: MOH native files   Tomislav Parčina
• [Asterisk-Users] Native music on hold - Error   Tomislav Parčina
• [Asterisk-Users] Re: Agents, queues and Pentalties   Tomislav Parčina
• [Asterisk-Users] Re: Info about F1000G   Tomislav Parčina
• [Asterisk-Users] Re: res_features pickupexten   Tomislav Parčina
• [Asterisk-Users] Re: Re: MOH native files   Tomislav Parčina
• [Asterisk-Users] Re: Native music on hold - Error   Tomislav Parčina
• [Asterisk-Users] Re: Get no busy signal on my analog line   Tomislav Parčina
• [Asterisk-Users] Re: Native music on hold - Error   Tomislav Parčina
• [Asterisk-Users] No new mails   Tomislav Parčina
• [Asterisk-Users] Set(LANGUAGE()=language) - for queue   Tomislav Parčina
• [Asterisk-Users] Asterisk add-ons - H323   Tomislav Parčina
• [Asterisk-Users] Gmane - Asterisk Users Mailing List   Tomislav Parčina
• [Asterisk-Users] Send One Touch Record to mail   Tomislav Parčina
• [Asterisk-Users] Send One Touch Record to mail   Tomislav Parčina
• [Asterisk-Users] Can't hear busy tone   Tomislav Parčina
• [Asterisk-Users] Cisco phones and Linksys SRW224P   Tomislav Parčina
• [Asterisk-Users] Re: Cisco phones and Linksys SRW224P   Tomislav Parčina
• [Asterisk-Users] RE: Re: Cisco phones and Linksys SRW224P   Tomislav Parčina
• [Asterisk-Users] Re: RE: Re: Cisco phones and Linksys SRW224P   Tomislav Parčina
• [Asterisk-Users] Re: RE: Re: Cisco phones and Linksys SRW224P   Tomislav Parčina
• [Asterisk-Users] Re: Attended Transfer - transfer timeout, how to change?   Tomislav Parčina
• [Asterisk-Users] Re: Do Not Disturb?   Tomislav Parčina
• [Asterisk-Users] Re: Queues - calls going to agents lised as "In use"   Tomislav Parčina
• [Asterisk-Users] Re: OT: Unblocking bloced CID   Tomislav Parčina
• [Asterisk-Users] Pickupexten not working   Tomislav Parčina
• [Asterisk-Users] ZOMBIE on att transfer   Tomislav Parčina
• [Asterisk-Users] Re: embedded hardware for Asterisk?   Tomislav Parčina
• [Asterisk-Users] Re: Exchange 12 Unified Messaging   Tomislav Parčina
• [Asterisk-Users] Re: gsm picocells   Tomislav Parčina
• [Asterisk-Users] Re: Server freeze with meetme and sip GSM users   Tomislav Parčina
• [Asterisk-Users] Re: MeetMe - Causes * to crash :/   Tomislav Parčina
• [Asterisk-Users] Cisco 7970   Tomislav Parčina
• [Asterisk-Users] Re: SIP trunk problem   Tomislav Parčina
• [Asterisk-Users] Re: Pickupexten not working   Tomislav Parčina
• [Asterisk-Users] RE: Re: OT: Unblocking bloced CID   Tomislav Parčina
• [Asterisk-Users] Asterisk add-ons upgrade   Tomislav Parčina
• [Asterisk-Users] Re: TAC Case Cisco 7960 Proxy address showing up in callerID   Tomislav Parčina
• [Asterisk-Users] Re: Problem with Queue periodic announcemnets   Tomislav Parčina
• [Asterisk-Users] Re: Which g729 codec to download for a P4?   Tomislav Parčina
• [Asterisk-Users] Re: problems compiling zaptel on FC5   Tomislav Parčina
• [Asterisk-Users] Re: Best GUI for basic HostedPBX service   Tomislav Parčina
• [Asterisk-Users] Free g729   Tomislav Parčina
• [Asterisk-Users] Re: reload - restart   Tomislav Parčina
• [Asterisk-Users] Re: making ooh323 authenticate gateway just like sip does   Tomislav Parčina
• [Asterisk-Users] Re: Cisco 7960 - Have to press a menu button to dial   Tomislav Parčina
• [Asterisk-Users] Caller ID length   Tomislav Parčina
• [Asterisk-Users] RE: Re: Best GUI for basic HostedPBX service   Tomislav Parčina
• [Asterisk-Users] Transfer after group pick-up   Tomislav Parčina
• [Asterisk-Users] Re: Re: Cisco 7960 - Have to press a menu button to dial   Tomislav Parčina
• [Asterisk-Users] Re: Free g729   Tomislav Parčina
• [Asterisk-Users] Re: Cisco 7970   Tomislav Parčina
• [Asterisk-Users] Re: Re: Cisco 7970   Tomislav Parčina
• [Asterisk-Users] Addons 1.2.1 upgrade to 1.2.2   Tomislav Parčina
• [Asterisk-Users] h323 channel driver for production   Tomislav Parčina
• [Asterisk-Users] Re: Agent in multiple queues?   Tomislav Parčina
• [Asterisk-Users] Re: Re: Agent in multiple queues?   Tomislav Parčina
• [Asterisk-Users] Re: SJphone Do not send silence - option ? Should be disabled for Asterisk   Tomislav Parčina
• [Asterisk-Users] Re: H323 behind a Firewall   Tomislav Parčina
• [Asterisk-Users] How to make groups of extensions ???   Francisco Pérez Botella
• [Asterisk-Users] SIP/Video client for PocketPC that works with Asterisk?   Adrian A
• [Asterisk-Users] Asterisk with Vonage   Adrian A
• [Asterisk-Users] Vmail.cgi and #include   Adrian A
• [Asterisk-Users] How to route incoming calls to different contexts?   Zach A
• [Asterisk-Users] How to route incoming calls to different contexts?   Zach A
• [Asterisk-Users] MOH native files   Zach A
• [Asterisk-Users] Info about mp3 which are installed with Asterisk   Zach A
• [Asterisk-Users] spa3000 asterisk fxo gateway   Zach A
• [Asterisk-Users] How to receive faxes with SPA3000 and Asterisk setup   Zach A
• [Asterisk-Users] What port mpg123 uses for MoH?   Zach A
• [Asterisk-Users] Putting caller in queue and dialing an extension simultaneously   Zach A
• [Asterisk-Users] how to check if ztdummy is working properly?   Zach A
• [Asterisk-Users] how to check if ztdummy is working properly?   Zach A
• [Asterisk-Users] how to check if ztdummy is working properly?   Zach A
• [Asterisk-Users] Professional Recordings   Zach A
• [Asterisk-Users] my zap channel not ringing   ADEGOKE ARUNA
• [Asterisk-Users] my zap channel not ringing   ADEGOKE ARUNA
• [Asterisk-Users] my zap channel not ringing   ADEGOKE ARUNA
• [Asterisk-Users] my zap channel not ringing & source from internal to telco line   ADEGOKE ARUNA
• [Asterisk-Users] my zap channel not ringing   ADEGOKE ARUNA
• [Asterisk-Users] HW Echo Cancellers   ADEGOKE ARUNA
• [Asterisk-Users] I need to set NO CRC4 on zaptel.conf?   ADEGOKE ARUNA
• [Asterisk-Users] FW: I need to set NO CRC4 on zaptel.conf?   ADEGOKE ARUNA
• [Asterisk-Users] need to make my oh323 work with quintum no gatekeeper   ADEGOKE ARUNA
• [Asterisk-Users] need to make my oh323 work with quintum no gatekeeper   ADEGOKE ARUNA
• [Asterisk-Users] need to make my oh323 work with quintum no gatekeeper   ADEGOKE ARUNA
• [Asterisk-Users] Installing the none commercial intel g729 codecs into Asterisk@Home 2.2?   ADEGOKE ARUNA
• [Asterisk-Users] transcoding g723 or g729 on asterisk   ADEGOKE ARUNA
• [Asterisk-Users] Transcoding on asterisk   ADEGOKE ARUNA
• [Asterisk-Users] Re: How is Teliax ?   Lonnie Abelbeck
• [Asterisk-Users] Grabbing the billsec and duration after a hangup.   Mark Ackroyd
• [Asterisk-Users] Grabbing the billsec and duration after a hangup.   Mark Ackroyd
• [Asterisk-Users] Small noise every 3 seconds   Alejandro Acosta
• [Asterisk-Users] SIP Problem - Asterisk to Provider Gateway   Gavin Adams
• [Asterisk-Users] SIP Problem - Asterisk to Provider Gateway   Gavin Adams
• [Asterisk-Users] Location of MeetMe Recordings   Gavin Adams
• [Asterisk-Users] Native MOH - Convert mp3 to ulaw   Gavin Adams
• [Asterisk-Users] System Design   Jason Adams
• [Asterisk-Users] System Design   Jason Adams
• [Asterisk-Users] System Design   Jason Adams
• [Asterisk-Users] RFC Follow Me Find Me script   Jason Adams
• [Asterisk-Users] A room full of Cisco 7960s behind NAT   Rich Adamson
• [Asterisk-Users] A room full of Cisco 7960s behind NAT   Rich Adamson
• [Asterisk-Users] Asterisk with T1 card on laptop   Rich Adamson
• [Asterisk-Users] Polling Asterisk for Life   Rich Adamson
• [Asterisk-Users] Sipura RMA   Rich Adamson
• [Asterisk-Users] 20 seconds til voice transmission starts   Rich Adamson
• [Asterisk-Users] Program Buttons on Cisco 79xx Phones   Rich Adamson
• [Asterisk-Users] really need help with outgoing calls..PSTN errors   Rich Adamson
• [Asterisk-Users] Problem compiling ztdummy on centos 4, 2.6 kernel   Rich Adamson
• [Asterisk-Users] spa3000 asterisk fxo gateway   Rich Adamson
• [Asterisk-Users] fax receive using TDM400P   Rich Adamson
• [Asterisk-Users] Sangoma A200 error   Rich Adamson
• [Asterisk-Users] IAX2 + Sonicwall   Rich Adamson
• [Asterisk-Users] IAX2 + Sonicwall   Rich Adamson
• [Asterisk-Users] Clustering   Rich Adamson
• [Asterisk-Users] Hung IAX Channels   Rich Adamson
• [Asterisk-Users] stop monitor on transfer   Rich Adamson
• [Asterisk-Users] Multiple IAX clients behind a firewall   Rich Adamson
• [Asterisk-Users] G729A   Rich Adamson
• [Asterisk-Users] RE: Multiple IAX clients behind a firewall   Rich Adamson
• [Asterisk-Users] RE: Delay in ringing   Rich Adamson
• [Asterisk-Users] Failed installing zaptel   Rich Adamson
• [Asterisk-Users] Analog Desktop Phone   Rich Adamson
• [Asterisk-Users] OT call recording (was stop monitor on transfer)   Rich Adamson
• [Asterisk-Users] saydigits   Rich Adamson
• [Asterisk-Users] Can One FXO Support Multiple Phone Lines?   Rich Adamson
• [Asterisk-Users] echo problem + choppy sound   Rich Adamson
• [Asterisk-Users] Bad FXS Module?   Rich Adamson
• [Asterisk-Users] echo problem + choppy sound   Rich Adamson
• [Asterisk-Users] OSHA requirement to "reach a live human" ??   Rich Adamson
• [SPAM] Re: [Asterisk-Users] OT - force Cisco phones to reboot   Rich Adamson
• [Asterisk-Users] Sipura 3000 DMTF   Rich Adamson
• [Asterisk-Users] Cisco 7960 dual ethernet port - bandwidth impact   Rich Adamson
• [Asterisk-Users] GS BT102 dual ethernet port -bandwidth impact   Rich Adamson
• [Asterisk-Users] Polycom IP600 dual ethernet port - bandwidth impact   Rich Adamson
• [Asterisk-Users] Asterisk Users Mailing List Traffic   Rich Adamson
• [Asterisk-Users] Polycom IP600 - no ring?   Rich Adamson
• [Asterisk-Users] An FXO version of IAXy?   Rich Adamson
• [Asterisk-Users] Polycom IP600 - no ring?   Rich Adamson
• [Asterisk-Users] g729 and latency measures   Rich Adamson
• [Asterisk-Users] g729 and latency measures   Rich Adamson
• [Asterisk-Users] Call Pickup Woes   Rich Adamson
• [Asterisk-Users] GS BT102 dual ethernet port -bandwidth impact   Rich Adamson
• [Asterisk-Users] Asterisk Users Mailing List Traffic   Rich Adamson
• [Asterisk-Users] pickup problem   Rich Adamson
• [Asterisk-Users] pickup problem   Rich Adamson
• [Asterisk-Users] Problem with chan_iax.c implimentation causes bad audio?   Rich Adamson
• [Asterisk-Users] How to make groups of extensions ???   Rich Adamson
• [Asterisk-Users] Phones were working fine - Now there is noaudiowhen calling between extensions   Rich Adamson
• [Asterisk-Users] How to make groups of extensions ???   Rich Adamson
• [Asterisk-Users] How to make groups of extensions ???   Rich Adamson
• [Asterisk-Users] Realtime SIP Persistency   Rich Adamson
• [Asterisk-Users] Problem with chan_iax.cimplimentationcausesbadaudio?   Rich Adamson
• [Asterisk-Users] Dimensions of TDM 24 card   Rich Adamson
• [Asterisk-Users] Hardware Levels   Rich Adamson
• [Asterisk-Users] best MTU?   Rich Adamson
• [Asterisk-Users] "Not Found" in archive   Rich Adamson
• [Asterisk-Users] I'm FED UP with BroadVoice   Rich Adamson
• [Asterisk-Users] Stability of Asterisk with 2 x TDM400P cards (6 analogue lines)   Rich Adamson
• [Asterisk-Users] Plain Old Answering Machine   Rich Adamson
• [Asterisk-Users] getting your own phone number   Rich Adamson
• SV: [Asterisk-Users] re: Sound issues on SIP-SIP calls   Rich Adamson
• [Asterisk-Users] [1.2.5] DTMF not being set correctly (RESEND)   Rich Adamson
• [Asterisk-Users] apic vs xt-pic on fc3?   Rich Adamson
• [Asterisk-Users] FXO without answer supervision   Rich Adamson
• [Asterisk-Users] Dell 2850 w/TDM2400?   Rich Adamson
• [Asterisk-Users] Regulatory Ruling about Caller-ID   Rich Adamson
• [Asterisk-Users] Marketing Materials   Rich Adamson
• [Asterisk-Users] TDM04B sound volume   Rich Adamson
• [Asterisk-Users] Callid on T-1 trunk   Rich Adamson
• [Asterisk-Users] How is Teliax ?   Rich Adamson
• [Asterisk-Users] Re: How is Teliax ?   Rich Adamson
• [Asterisk-Users] Re: Asterisk in production as a fax server, anyone?   Rich Adamson
• [Asterisk-Users] Re: Asterisk in production as a fax server, anyone?   Rich Adamson
• [Asterisk-Users] Re: How is Teliax ?   Rich Adamson
• [Asterisk-Users] streaming recordings   Jon-o Addleman
• [Asterisk-Users] Voice volume using Monitor application   Jon-o Addleman
• [Asterisk-Users] Voice volume using Monitor application   Jonathan Addleman
• [Asterisk-Users] Failover Registration   Gabriel Afana
• [Asterisk-Users] Re: One Extension - Two Calls?   Gabriel Afana
• [Asterisk-Users] Calls between Asterisk servers using SIP? What about IAX (got it working w/ IAX but I have questions)   Gabriel Afana
• [Asterisk-Users] Calls between Asterisk servers using SIP?What about IAX (got it working w/ IAX but I have questions)   Gabriel Afana
• [Asterisk-Users] Some ignorance here, what exactly is a Session Control Border? (Verizon is asking me about this)   Gabriel Afana
• [Asterisk-Users] Extensions base policy   Gabriel Afana
• [Asterisk-Users] I think I found a glitch in LCDial()   Gabriel Afana
• [Asterisk-Users] transfering between asterisk servers - MOH drops out   Gabriel Afana
• [Asterisk-Users] Clustering   Gabriel Afana
• [Asterisk-Users] Clustering   Gabriel Afana
• [Asterisk-Users] Clustering   Gabriel Afana
• [Asterisk-Users] Can't figure out how to hangup a call and continue to the next priority   Gabriel Afana
• [Asterisk-Users] Clustering   Gabriel Afana
• [Asterisk-Users] Dialplan woes   Gabriel Afana
• [Asterisk-Users] Voice problem   Gabriel Afana
• [Asterisk-Users] Is it possible to reinvite twice?   Gabriel Afana
• [Asterisk-Users] Dialplan woes   Gabriel Afana
• [Asterisk-Users] Clustering   Gabriel Afana
• [Asterisk-Users] Voice problem   Gabriel Afana
• [Asterisk-Users] Clustering   Gabriel Afana
• [Asterisk-Users] Dialplan woes   Gabriel Afana
• [Asterisk-Users] priorityjumping=no   Gabriel Afana
• [Asterisk-Users] priorityjumping=no   Gabriel Afana
• [Asterisk-Users] Woooo, Polycom and * on crack - can't register!   Gabriel Afana
• [Asterisk-Users] Woooo, Polycom and * on crack - can't register!   Gabriel Afana
• [Asterisk-Users] Woooo, Polycom and * on crack - can'tregister!   Gabriel Afana
• [Asterisk-Users] Im at VON and should see Mark Spencer tomorrow....any requests??   Gabriel Afana
• [Asterisk-Users] how to show called name on calling polycom display   Gabriel Afana
• [Asterisk-Users] Feedback from VON expo! Info on * HA and Polycom phone!!   Gabriel Afana
• [Asterisk-Users] Re: how to show called name on callingpolycomdisplay   Gabriel Afana
• [Asterisk-Users] Feedback from VON expo! Info on * HA andPolycomphone!!   Gabriel Afana
• [Asterisk-Users] Feedback from VON expo! Info on*HAandPolycomphone!!   Gabriel Afana
• [Asterisk-Users] Re: regexten   Gabriel Afana
• [Asterisk-Users] Re: regexten   Gabriel Afana
• [Asterisk-Users] Re: regexten   Gabriel Afana
• [Asterisk-Users] Feedback from VON expo! Info on * HAandPolycomphone!!   Gabriel Afana
• [Asterisk-Users] Feedback from VON expo! Info on *HAandPolycomphone!!   Gabriel Afana
• [Asterisk-Users] Phones were working fine - Now there is no audio when calling between extensions   Gabriel Afana
• [Asterisk-Users] Phones were working fine - Now there is no audiowhen calling between extensions   Gabriel Afana
• [Asterisk-Users] Phones were working fine - Now there is noaudiowhen calling between extensions   Gabriel Afana
• [Asterisk-Users] Phones were working fine - Now there is noaudiowhen calling between extensions   Gabriel Afana
• [Asterisk-Users] Phones were working fine - Now there isnoaudiowhen calling between extensions   Gabriel Afana
• [Asterisk-Users] If a packet is dropped, is it possible to replay the previous packet in Asterisk?   Gabriel Afana
• [Asterisk-Users] Polycom 501's for sale   Gabriel Afana
• [Asterisk-Users] G729 codec problems   Gabriel Afana
• [Asterisk-Users] Any Polycom dealer willing to help?   Gabriel Afana
• [Asterisk-Users] Polycom IP 301 is slow   Gabriel Afana
• [Asterisk-Users] Polycom SIP 1.6.5 and Bootrom 3.1.3 - Download link   Gabriel Afana
• [Asterisk-Users] Asterisk and LCR   Gabriel Afana
• [Asterisk-Users] TE411P VPM   Imran Ahmed
• [Asterisk-Users] Speeding up the dial of DTMF's in SIP channel   Imran Ahmed
• [Asterisk-Users] Codec Issue   Aisling
• [Asterisk-Users] Codecs? - Asked to transmit frame type 256, while native formats is 8 (read/write = 8/8)   Aisling
• [Asterisk-Users] Asterisk Fax Question   Jorge Alayon
• [Asterisk-Users] help with asterisk installation   Alban
• [Asterisk-Users] Cisco POS 3-08-2   Nathan Alberti
• [Asterisk-Users] Cisco POS 3-08-2   Nathan Alberti
• [Asterisk-Users] G729 License questions   Nathan Alberti
• [Asterisk-Users] Multicast Music on Hold   Nathan Alberti
• [Asterisk-Users] DISA & SPA3000 issues   Alchaemist
• [Asterisk-Users] Question on compiling Zaptel   Larry Alkoff
• [Asterisk-Users] How to create [new_context] in extensions.conf?   Larry Alkoff
• [Asterisk-Users] How to create [new_context] in extensions.conf?   Larry Alkoff
• [Asterisk-Users] How to create [new_context] in extensions.conf?   Larry Alkoff
• [Asterisk-Users] How to create [new_context] in extensions.conf?   Larry Alkoff
• [Asterisk-Users] How to create [new_context] in extensions.conf?   Larry Alkoff
• [Asterisk-Users] How to create [new_context] in extensions.conf?   Larry Alkoff
• [Asterisk-Users] How to create [new_context] in extensions.conf?   Larry Alkoff
• [Asterisk-Users] How to create [new_context] in extensions.conf?   Larry Alkoff
• [Asterisk-Users] calls only for logging users   Pablo Allietti
• [Asterisk-Users] hangup on silence?   Pablo Allietti
• [Asterisk-Users] Can anybody get me setup with a hosted asterisk@home box or virtual server in the next 24 hours?   Carlos Alperin
• [Asterisk-Users] On site installtion Tech. wanted   Richard Amerman
• [Asterisk-Users] FreePBX & AAH   Richard Amerman
• [Asterisk-Users] Re: digium.com redesign   Benny Amorsen
• [Asterisk-Users] Re: Fw: anybody has SIP realtime working ?   Benny Amorsen
• [Asterisk-Users] Re: Fw: anybody has SIP realtime working ?   Benny Amorsen
• [Asterisk-Users] Re: Fw: anybody has SIP realtime working ?   Benny Amorsen
• [Asterisk-Users] Re: Fw: anybody has SIP realtime working ?   Benny Amorsen
• [Asterisk-Users] Re: How to nice agi scripts?   Benny Amorsen
• [Asterisk-Users] SIP caller id   Ryan Amos
• [Asterisk-Users] Chinaroby VOIP phones? SECOND TIME!   Andreas Anderson
• [Asterisk-Users] Capturing DIALSTATUS on a PARTICULAR channel if multiple-dialling?   Colin Anderson
• [Asterisk-Users] Snom 360 Hinting tricks   Colin Anderson
• [Asterisk-Users] Dial() cmd executing Macro - dropped audio   Colin Anderson
• [Asterisk-Users] Asterisk on MacOS?   Colin Anderson
• [Asterisk-Users] Asterisk download file locations   Colin Anderson
• [Asterisk-Users] Asterisk download file locations   Colin Anderson
• [Asterisk-Users] System Design   Colin Anderson
• [Asterisk-Users] OT: Snom 320, displaying text on the screen from *   Colin Anderson
• [Asterisk-Users] OT: Snom 320, displaying text on the scree n from *   Colin Anderson
• [Asterisk-Users] OT: Snom 320, displaying text on the scree n from *   Colin Anderson
• [Asterisk-Users] Adding entries on company directory   Colin Anderson
• [Asterisk-Users] OT: Using Sipsak to reboot a Snom phone   Colin Anderson
• [Asterisk-Users] OT: Using Sipsak to reboot a Snom phone < -a nswered my own question   Colin Anderson
• [Asterisk-Users] Echo canceller data-points   Colin Anderson
• [Asterisk-Users] New one on me: How to UN-transfer   Colin Anderson
• [Asterisk-Users] New one on me: How to UN-transfer   Colin Anderson
• [Asterisk-Users] New one on me: How to UN-transfer   Colin Anderson
• [Asterisk-Users] New one on me: How to UN-transfer   Colin Anderson
• [Asterisk-Users] Re: problems with emailing voicemail   Colin Anderson
• [Asterisk-Users] FOP flash panel: how to reload config files when running   Colin Anderson
• [Asterisk-Users] OT: HOWTO: Query channel state on an Ateus Voice Blue GSM gateway   Colin Anderson
• [Asterisk-Users] Asterisk with Vonage   Colin Anderson
• [Asterisk-Users] OT: FOP and reverse_transfer   Colin Anderson
• [Asterisk-Users] PRI CID signalling not working?   Erik Anderson
• [Asterisk-Users] interfacing w/ a legacy InterTel PBX   Erik Anderson
• [Asterisk-Users] PRI CID signalling not working?   Erik Anderson
• [Asterisk-Users] VERY IMPORTANT(TREAT WITH URGENCY)   Erik Anderson
• [Asterisk-Users] VERY IMPORTANT(TREAT WITH URGENCY)   Erik Anderson
• [Asterisk-Users] No sound after loading module wcte11xp   Robert Andersson
• [Asterisk-Users] Can this box handle 8 T1s (PSTN) with Asterisk?   Andres
• [Asterisk-Users] Can this box handle 8 T1s (PSTN) with Asterisk?   Andres
• [Asterisk-Users] PRI Behavior   Andres
• [Asterisk-Users] RTP frame size location?   Andres
• [Asterisk-Users] RTP frame size location?   Andres
• [Asterisk-Users] Zoom 5801 problems with * (Wellgate 3701a)   Cory Andrews
• [Asterisk-Users] Zoom 5801 problems with * (Wellgate 3701a)   Cory Andrews
• [Asterisk-Users] Problem compiling ztdummy on centos 4, 2.6 kernel   Cory Andrews
• [Asterisk-Users] Analog Desktop Phone   Cory Andrews
• [Asterisk-Users] Analog Desktop Phone   Cory Andrews
• [Asterisk-Users] Web-ex type solution for use with asterisk   Cory Andrews
• [Asterisk-Users] WiFi phones and WDS (Wireless Distribution System)   Cory Andrews
• [Asterisk-Users] An FXO version of IAXy?   Cory Andrews
• [Asterisk-Users] An FXO version of IAXy?   Cory Andrews
• [Asterisk-Users] Netgear FS116P and Cisco 79XX phones   Cory Andrews
• [Asterisk-Users] Call Monitoring / Call Takeover with Asterisk   Cory Andrews
• [Asterisk-Users] Asterisk Referral - Cleanup on Aisle 7   Cory Andrews
• [Asterisk-Users] beronet & bristuff   Francesco Angi
• [Asterisk-Users] Sipura SPA-3000 in Egypt   Samy Antoun
• [Asterisk-Users] Re: transfers/parked calls + polycom 501   Araba, Michael
• [Asterisk-Users] HDLC error   Lee Archer
• [Asterisk-Users] How to change Budgetone dialtone?   Lee Archer
• [Asterisk-Users] How to change Budgetone dialtone?   Lee Archer
• [Asterisk-Users] Double-ring tone   Lee Archer
• [Asterisk-Users] Double-ring tone   Lee Archer
• [Asterisk-Users] Multiple processes   Lee Archer
• [Asterisk-Users] Setting Max Calls on an IAX trunk   Marc Archer
• [Asterisk-Users] Receiving Multiple calls on asterisk at home   Marc Archer
• [Asterisk-Users] Asterisk Re-invites - how to tell ?   Marc Archer
• [Asterisk-Users] Building a small Office EPABX with VoIP GW with Asterisk   Sanjay Arora
• [Asterisk-Users] T1 card Ali   Ali Arshad
• FW: [Asterisk-Users] T1 card Ali   Ali Arshad
• [Asterisk-Users] Attended Transfer - transfer timeout, how to change?   Thomas Artner
• [Asterisk-Users] Re: Attended Transfer - transfer timeout, how to change?   Thomas Artner
• [Asterisk-Users] D-Link DVG-1402S   Stephen Arulraj
• [Asterisk-Users] D-Link DVG-1402S   Stephen Arulraj
• [Asterisk-Users] Quintum Tenor DX4060   Stephen Arulraj
• [Asterisk-Users] making ooh323 authenticate gateway just like sip does   Goke Aruna
• [Asterisk-Users] Asterisk transfer conflict   Bjorn Asmul
• [Asterisk-Users] Same CID on multiple users(friends9 in SIP.conf   Bjorn Asmul
• [Asterisk-Users] problems with DTMF   Bjorn Asmul
• [Asterisk-Users] Asterisk & Avaya Legend   Lacy Moore - Aspendora
• [Asterisk-Users] Asterisk Users Mailing List Traffic   Lacy Moore - Aspendora
• [Asterisk-Users] Asterisk Users   Lacy Moore - Aspendora
• [Asterisk-Users] Asterisk & Avaya Legend   Lacy Moore - Aspendora
• [Asterisk-Users] Asterisk Users   Lacy Moore - Aspendora
• [Asterisk-Users] Calling home while on the road, will it work?   Lacy Moore - Aspendora
• [Asterisk-Users] Regulatory Ruling about Caller-ID   Lacy Moore - Aspendora
• [Asterisk-Users] Call terminated after 60 seconds   Asterisk
• [Asterisk-Users] Call terminated after 60 seconds   Asterisk
• [Asterisk-Users] Dial Out IVR   Jonathan Attwood
• [Asterisk-Users] codec translation problem???   Atuc
• [Asterisk-Users] Polycom 501   Mark Aufflick
• [Asterisk-Users] Australian approved 4BRI PCI adapter preliminary testing results   Mark Aufflick
• [Asterisk-Users] Flash zap trunk from Softphone or IP Handsets...   Mark Aufflick
• [Asterisk-Users] Australian approved 4BRI PCI adapter preliminarytesting results   Mark Aufflick
• [Asterisk-Users] Question about meetme app   Jonathan Augenstine
• [Asterisk-Users] Question about meetme app   Jonathan Augenstine
• [Asterisk-Users] I'm FED UP with BroadVoice   Jonathan Augenstine
• [Asterisk-Users] WARNING[5171]: res_musiconhold.c:833 moh_register: Unable to open pseudo channel for timing   Jonathan Augenstine
• [Asterisk-Users] web meetme instructions   Dan Austin
• [Asterisk-Users] FXO without answer supervision   Dan Austin
• [Asterisk-Users] FXO without answer supervision   Dan Austin
• [Asterisk-Users] web meetme instructions   Dan Austin
• [Asterisk-Users] RTP frame size location?   Dan Austin
• [Asterisk-Users] Polycom 501   Azfhasterisk
• [Asterisk-Users] OT: Polycom BootRom 3.1.3 and vsftpd 2.0.3   Azfhasterisk
• [Asterisk-Users] Re: How is Teliax ?   Azfhasterisk
• [Asterisk-Users] Two PBX   Hafez Azzam
• [Asterisk-Users] Two PBX   Hafez Azzam
• [Asterisk-Users] I have my asterisk machine behind a Linux Nat ...   Anthony Azzopardi
• [Asterisk-Users] MWI, SER and asterisk   Christian B
• [Asterisk-Users] difference between records in CDR and real duration of call   Christian B
• [Asterisk-Users] Action after _caller_ has hungup(cmd Dial 'g'-option)   Christian B
• [Asterisk-Users] Action after _caller_ has hungup(cmd Dial 'g'-option)   Christian B
• [Asterisk-Users] Action after _caller_ has hungup(cmd Dial 'g'-option)   Christian B
• [Asterisk-Users] simple perl-agi - where's the error?   Christian B
• [Asterisk-Users] simple perl-agi - where's the error?   Christian B
• [Asterisk-Users] simple perl-agi - where's the error?   Christian B
• [Asterisk-Users] simple perl-agi - where's the error?   Christian B
• [Asterisk-Users] TDM400 DTMF Caller ID   Fernando BERRETTA
• [Asterisk-Users] TDM 2400 With 24 FXO   Fernando BERRETTA
• [Asterisk-Users] TDM 2400 With 24 FXO   Fernando BERRETTA
• [Asterisk-Users] test call quality   amaury BOSSE
• [Asterisk-Users] internals and ISDN calls fail when Internet is down   amaury BOSSE
• [Asterisk-Users] Same CID on multiple users(friends9 in SIP.conf   Michiel van Baak
• [Asterisk-Users] error messages on /var/log/asterisk/messages   Michiel van Baak
• [Asterisk-Users] error messages on /var/log/asterisk/messages   Michiel van Baak
• [Asterisk-Users] A room full of Cisco 7960s behind NAT   Michiel van Baak
• [Asterisk-Users] Program Buttons on Cisco 79xx Phones   Michiel van Baak
• [Asterisk-Users] Program Buttons on Cisco 79xx Phones   Michiel van Baak
• [Asterisk-Users] Preferred editor(s) dialplan coding?   Michiel van Baak
• [Asterisk-Users] seg fault when skinny phone answers   Michiel van Baak
• [Asterisk-Users] Program Buttons on Cisco 79xx Phones   Michiel van Baak
• [Asterisk-Users] Program Buttons on Cisco 79xx Phones   Michiel van Baak
• [Asterisk-Users] login/logout agents in a specific queue   Michiel van Baak
• [Asterisk-Users] NEWS: SIP Firmware Available for Cisco 7970   Michiel van Baak
• [Asterisk-Users] seg fault when skinny phone answers   Michiel van Baak
• [Asterisk-Users] Asterisk and CISCO 7970 color   Michiel van Baak
• [Asterisk-Users] monitor/statistic web interface for cdr   Michiel van Baak
• [Asterisk-Users] Preferred editor(s) dialplan coding?   Michiel van Baak
• [Asterisk-Users] How to assign a specific PSTN line to a specific extension ???   Michiel van Baak
• [Asterisk-Users] open source queue analyzer   Michiel van Baak
• [Asterisk-Users] MeetMe freezes machine with Junghanns QuadBRI cards   Michiel van Baak
• [Asterisk-Users] RE: IAX Incoming/Outgoing   Michiel van Baak
• [Asterisk-Users] time update (7905)   Michiel van Baak
• [Asterisk-Users] queue caveats   Michiel van Baak
• [Asterisk-Users] How to disable event_log?   Michiel van Baak
• [Asterisk-Users] OT: HOWTO: Query channel state on an Ateus Voice Blue GSM gateway   Michiel van Baak
• [Asterisk-Users] Asterisk and Hylafax, on the same box   Michiel van Baak
• [Asterisk-Users] Building Asterisk embedded device   Michiel van Baak
• [Asterisk-Users] IAX2 issue with calls transfered between systems.   Jason Bachman
• [Asterisk-Users] Remote dialtone   Jason Bachman
• [Asterisk-Users] Asterisk transfer conflict   Wes Baehr
• [Asterisk-Users] Queue Period-Announce   Wes Baehr
• [Asterisk-Users] Comments on Gafachi   Wes Baehr
• [Asterisk-Users] Problem compiling ztdummy on centos 4, 2.6 kernel   Sina Bahram
• [Asterisk-Users] Problem compiling ztdummy on centos 4, 2.6 kernel   Sina Bahram
• [Asterisk-Users] Problem compiling ztdummy on centos 4, 2.6 kernel   Sina Bahram
• [Asterisk-Users] Problem compiling ztdummy on centos 4, 2.6 kernel   Sina Bahram
• [Asterisk-Users] Problem compiling ztdummy on centos 4, 2.6 kernel   Sina Bahram
• [Asterisk-Users] Problem compiling ztdummy on centos 4, 2.6 kernel   Sina Bahram
• [Asterisk-Users] Problem compiling ztdummy on centos 4, 2.6 kernel   Sina Bahram
• [Asterisk-Users] Problem compiling ztdummy on centos 4, 2.6 kernel   Sina Bahram
• [Asterisk-Users] Reload astdb?   Boris Bakchiev
• [Asterisk-Users] Re: Asterisk in production as a fax server, anyone?   Boris Bakchiev
• [Asterisk-Users] Asterisk and Hylafax, on the same box   Boris Bakchiev
• [Asterisk-Users] Iaxmodem speed limit?   Boris Bakchiev
• [Asterisk-Users] ON DEMAND call Recording   Giridhar Bandi
• [Asterisk-Users] Re: ON DEMAND call Recording   Giridhar Bandi
• [Asterisk-Users] most common VOIP echo simulaton for research purposes ?   Giridhar Bandi
• [Asterisk-Users] pap2 Dial plan   Giridhar Bandi
• [Asterisk-Users] Re: ON DEMAND call Recording   Giridhar Bandi
• [Asterisk-Users] Using softphone from a remote location to get into *   Giridhar Bandi
• [Asterisk-Users] paly sound when we Start and stop recording   Giridhar Bandi
• [Asterisk-Users] pap2 Dial plan   Giridhar Bandi
• RES: [Asterisk-Users] pap2 Dial plan   Giridhar Bandi
• [Asterisk-Users] Voice Mail woe   Giridhar Bandi
• [Asterisk-Users] How is Teliax ?   Giridhar Reddy Bandi
• [Asterisk-Users] How is Teliax ?   Giridhar Reddy Bandi
• [Asterisk-Users] Auto dial feature   Time Bandit
• [Asterisk-Users] Reverse group in zapata.conf   Time Bandit
• [Asterisk-Users] IVR dial by extension option..   Time Bandit
• [Asterisk-Users] Dialplan woes   Time Bandit
• [Asterisk-Users] Latest Dell SC430 Compatibility With Wildcard   Time Bandit
• [Asterisk-Users] cards   Time Bandit
• [Asterisk-Users] Incoming calls   Time Bandit
• [Asterisk-Users] send text to a device   Time Bandit
• [Asterisk-Users] Question about advanced IVR   Time Bandit
• [Asterisk-Users] More Voicemail prompts   Time Bandit
• [Asterisk-Users] pickup a call in queue   Time Bandit
• [Asterisk-Users] Extension a?   Time Bandit
• [Asterisk-Users] restart problem   Avneet Bansal
• [Asterisk-Users] asterisk and iptables   Andres Baravalle
• [Asterisk-Users] Calls between Asterisk servers using SIP? What about IAX (got it working w/ IAX but I have questions)   Umair Bari
• [Asterisk-Users] GSM/DECT handsets (was gsm picocells)   Iain Barker
• [Asterisk-Users] Asterisk transfer conflict   Pete Barnwell
• [Asterisk-Users] Preferred editor(s) dialplan coding?   Pete Barnwell
• [Asterisk-Users] help with asterisk installation   Pete Barnwell
• [Asterisk-Users] problems with emailing voicemail   Pete Barnwell
• [Asterisk-Users] Aastra Questions   Pete Barnwell
• [Asterisk-Users] g729 and latency measures   Pete Barnwell
• [Asterisk-Users] Re: problems with emailing voicemail   Pete Barnwell
• [Asterisk-Users] Howto cut the first digit   Pete Barnwell
• [Asterisk-Users] How to forward inbound sipgate calls to different users in my entreprise, (   Francois-Xavier Bas
• [Asterisk-Users] Inbound sipgate number forwarding to differnet users   Francois-Xavier Bas
• [Asterisk-Users] iax limit question   Dan Batrams
• [Asterisk-Users] iax limit question   Dan Batrams
• [Asterisk-Users] chan_h323 problem   Balgansuren Batsukh
• [Asterisk-Users] chan_h323 problem   Balgansuren Batsukh
• [Asterisk-Users] how to run asterisk?   Bayrouni
• [Asterisk-Users] How often do YOU register?   Arek Bekiersz
• [Asterisk-Users] Action after _caller_ has hungup(cmd Dial 'g'-option)   Benchev
• [Asterisk-Users] Action after _caller_ has hungup(cmd Dial 'g'-option)   Benchev
• [Asterisk-Users] Asterisk RealTime Question, Please help   Benchev
• [Asterisk-Users] Asterisk RealTime Question, Please help   Benchev
• [Asterisk-Users] CDR question   Benchev
• [Asterisk-Users] CDR question   Benchev
• [Asterisk-Users] CDR question   Benchev
• [Asterisk-Users] Asterisk RealTime Question, Please help   Benchev
• [Asterisk-Users] Asterisk RealTime Question, Please help   Benchev
• [Asterisk-Users] Anyone using the GSMgateway from CyberTelecom ?   Benchev
• [Asterisk-Users] Anyone using the GSMgateway from CyberTelecom ?   Benchev
• [Asterisk-Users] iax limit question   Benchev
• [Asterisk-Users] Prepaid / postpaid solution   Dovid Bender
• [Asterisk-Users] Problem compiling ztdummy on centos 4, 2.6 kernel   Dovid Bender
• [Asterisk-Users] 160 analogue phones..   Dovid Bender
• [Asterisk-Users] Problem compiling ztdummy on centos 4, 2.6 kernel   Dovid Bender
• [Asterisk-Users] Preferred editor(s) dialplan coding?   Dovid Bender
• [Asterisk-Users] Info about mp3 which are installed with Asterisk   Dovid Bender
• [Asterisk-Users] Problem compiling ztdummy on centos 4, 2.6 kernel   Dovid Bender
• [Asterisk-Users] Problem compiling ztdummy on centos 4, 2.6 kernel   Dovid Bender
• [Asterisk-Users] Using softphone from a remote location to get into *   Dovid Bender
• [Asterisk-Users] Setting Vaaibles   Dovid Bender
• [Asterisk-Users] No DTMF   Dovid Bender
• [Asterisk-Users] Setting Vaaibles   Dovid Bender
• [Asterisk-Users] List Problems   Dovid Bender
• [Asterisk-Users] Asterisk download file locations   Dovid Bender
• [Asterisk-Users] Asterisk Prepaid Card   Dovid Bender
• [Asterisk-Users] No DTMF   Dovid Bender
• [Asterisk-Users] Preferred editor(s) dialplan coding?   Dovid Bender
• [Asterisk-Users] how to check if ztdummy is working properly?   Dovid Bender
• [Asterisk-Users] Call and then play IVR   Dovid Bender
• [Asterisk-Users] stop monitor on transfer   Dovid Bender
• [Asterisk-Users] Stuck. Extenions.conf? Realtime? MySQL? Grrrrr!   Dovid Bender
• [Asterisk-Users] Stuck. Extenions.conf? Realtime? MySQL? Grrrrr!   Dovid Bender
• [Asterisk-Users] OT: Unblocking bloced CID   Dovid Bender
• [Asterisk-Users] saydigits   Dovid Bender
• [Asterisk-Users] OT: Unblocking bloced CID   Dovid Bender
• [Asterisk-Users] Outbound paging dialplan example?   Dovid Bender
• [Asterisk-Users] Question about advanced IVR   Dovid Bender
• [Asterisk-Users] All calls in queue go to agent that is down??   Dovid Bender
• [Asterisk-Users] Annoying Asterisk Realtime Limitation   Dovid Bender
• [Asterisk-Users] ADT/Brinks alarm dialing through Asterisk   Dovid Bender
• [Asterisk-Users] Asterisk Users Mailing List Traffic   Dovid Bender
• [Asterisk-Users] VERY IMPORTANT(TREAT WITH URGENCY)   Dovid Bender
• [Asterisk-Users] Dialling Problem   Dovid Bender
• [Asterisk-Users] How to create [new_context] in extensions.conf?   Dovid Bender
• [Asterisk-Users] I'm FED UP with BroadVoice   Dovid Bender
• [Asterisk-Users] I'm FED UP with BroadVoice   Dovid Bender
• [Asterisk-Users] Best GUI for basic HostedPBX service   Dovid Bender
• [Asterisk-Users] problems with DTMF   Dovid Bender
• [Asterisk-Users] compiling Zaptel-1.2.4 on CentOS 4.3   Dovid Bender
• [Asterisk-Users] Voicemail to Email   Dovid Bender
• [Asterisk-Users] Re: Best GUI for basic HostedPBX service   Dovid Bender
• [Asterisk-Users] registration with different username   Dovid Bender
• [Asterisk-Users] after-queues   Dovid Bender
• [Asterisk-Users] FreePBX & AAH   Dovid Bender
• [Asterisk-Users] Outgoing SIP Failover   Dovid Bender
• [Asterisk-Users] Please Help Test Quad PRI Using NFAS   Dovid Bender
• [Asterisk-Users] Realtime Users/Peers/Friends - Ick   Dovid Bender
• [Asterisk-Users] Connecting a Grandstream Handytone 486 to Asterisk   Dovid Bender
• [Asterisk-Users] ISDN and Analog DIAL UP Connection Through Asterisk and Digium TE405P   Dovid Bender
• [Asterisk-Users] Marketing Materials   Dovid Bender
• [Asterisk-Users] Calling home while on the road, will it work?   Dovid Bender
• [Asterisk-Users] multiple auto attendants   Dovid Bender
• [Asterisk-Users] cdr records on transfer   Christian Benke
• [Asterisk-Users] cdr records on transfer   Christian Benke
• [Asterisk-Users] Re: Which Mac OSX softphone with IAX2 support?   Aldo Bergamini
• [Asterisk-Users] Can One FXO Support Multiple Phone Lines?   Andrew Berman
• [Asterisk-Users] Can One FXO Support Multiple Phone Lines?   Andrew Berman
• [Asterisk-Users] Call Path Optimization?   Vipul Bhatt
• [Asterisk-Users] queues & tranfers   Dov Bigio
• [Asterisk-Users] error messages on /var/log/asterisk/messages   Dov Bigio
• [Asterisk-Users] Fax behind ATA   Dov Bigio
• [Asterisk-Users] cdr data   Dov Bigio
• [Asterisk-Users] after-queues   Dov Bigio
• [Asterisk-Users] statechange_queue   Dov Bigio
• [Asterisk-Users] Help with Gizmo from outside firewall   Bill
• [Asterisk-Users] Re: Help with Gizmo from outside firewall   Bill
• [Asterisk-Users] Help with Gizmo from outside firewall <- update   Bill
• [Asterisk-Users] Any Polycom dealer willing to help?   Eric Bishop
• [Asterisk-Users] Meetme   John Bittner
• SV: [Asterisk-Users] IAX - only one way traffic   Bjorn
• SV: [Asterisk-Users] IAX - only one way traffic   Bjorn
• [Asterisk-Users] setting callerid not working if no callerid on incoming number   Gareth Blades
• [Asterisk-Users] setting callerid not working if no callerid on incoming number   Gareth Blades
• [Asterisk-Users] Best budget IP phone at the moment?   Gareth Blades
• [Asterisk-Users] OT - force Cisco phones to reboot   Steve Blair
• [Asterisk-Users] Cisco 7960 SIP - Displaying Time   Ben Blakely
• [Asterisk-Users] new beta Grandstream firmware HT488_496_386   Phil Blundell
• [Asterisk-Users] Unknown signalling method 'pri_cpe'   Kevin Bockman
• [Asterisk-Users] Attended Transfer - transfer timeout, how to change?   Kevin Bockman
• [Asterisk-Users] Menu in queue   William Boehlke
• [Asterisk-Users] Stability and motherboard questions with TE406Pand TE410P   William Boehlke
• [Asterisk-Users] Asterisk Failover without SER   William Boehlke
• [Asterisk-Users] Asterisk and Hylafax, on the same box   William Boehlke
• [Asterisk-Users] G.726 codec - can we select bandwidth?   Leo Ann Boon
• [Asterisk-Users] A room full of Cisco 7960s behind NAT   Leo Ann Boon
• [Asterisk-Users] G.726 codec - can we select bandwidth?   Leo Ann Boon
• [Asterisk-Users] PAP2 won't make two g729 calls at the same time   Leo Ann Boon
• [Asterisk-Users] Latest Dell SC430 Compatibility With Wildcard   Leo Ann Boon
• [Asterisk-Users] gsm picocells   Leo Ann Boon
• [Asterisk-Users] gsm picocells   Leo Ann Boon
• [Asterisk-Users] 计划生育的无耻宣传该结束了   Leo Ann Boon
• [Asterisk-Users] 计划生育的无耻宣传该结束了   Leo Ann Boon
• [Asterisk-Users] Re: gsm picocells   Leo Ann Boon
• [Asterisk-Users] best MTU?   Leo Ann Boon
• [Asterisk-Users] How to create [new_context] in extensions.conf?   Leo Ann Boon
• [Asterisk-Users] GSM/DECT handsets (was gsm picocells)   Leo Ann Boon
• [Asterisk-Users] Psgw   Leo Ann Boon
• [Asterisk-Users] Cisco 7960 SIP - Displaying Time   Nathan Bowyer
• [Asterisk-Users] Cisco 7960 SIP - Displaying Time   Nathan Bowyer
• [Asterisk-Users] how to show called name on calling polycom display   Nathan Bowyer
• [Asterisk-Users] how to show called name on calling polycomdisplay   Nathan Bowyer
• [Asterisk-Users] Feedback from VON expo! Info on * HA and Polycom phone!!   Nathan Bowyer
• [Asterisk-Users] MusicOnHold with mpg123   Nathan Bowyer
• [Asterisk-Users] cdr_odbc appears to have fields missing   Nathan Bowyer
• [Asterisk-Users] Advice on configuration   Peter Bowyer
• [Asterisk-Users] Clustering   Peter Bowyer
• [Asterisk-Users] Clustering   Peter Bowyer
• [Asterisk-Users] Clustering "NEW THREAD", Almost Working   Peter Bowyer
• [Asterisk-Users] regexten   Peter Bowyer
• [Asterisk-Users] Feedback from VON expo! Info on * HA and Polycomphone!!   Peter Bowyer
• [Asterisk-Users] Feedback from VON expo! Info on * HA andPolycomphone!!   Peter Bowyer
• [Asterisk-Users] AMP and ABE   Peter Bowyer
• [Asterisk-Users] GSM/DECT handsets (was gsm picocells)   Peter Bowyer
• [Asterisk-Users] Asterisk with Vonage   Peter Bowyer
• [Asterisk-Users] Regulatory Ruling about Caller-ID   Peter Bowyer
• [Asterisk-Users] Building Asterisk embedded device   Peter Bowyer
• [Asterisk-Users] Cisco 7960 8.2 callerID lists proxy?   Peter Braidwood
• [Asterisk-Users] Re: FAX over PRI   Adolfo R. Brandes
• [Asterisk-Users] Asterisk in production as a fax server, anyone?   Adolfo R. Brandes
• [Asterisk-Users] Re: Asterisk in production as a fax server, anyone?   Adolfo R. Brandes
• [Asterisk-Users] Re: Asterisk in production as a fax server, anyone?   Adolfo R. Brandes
• [Asterisk-Users] Re: One Extension - Two Calls?   Bromont
• [Asterisk-Users] Mitel SIP firmware   Bromont
• [Asterisk-Users] Sipura 3000 DMTF   John Brookes
• [Asterisk-Users] Asterisk to a Huawei softX3000   Glen Browley
• [Asterisk-Users] Cisco 7970 SIP Image   Paul Brown
• [Asterisk-Users] Advice on configuration   Paul A Brown
• [Asterisk-Users] Advice on configuration   Paul A Brown
• [Asterisk-Users] Newbie Questions - Any help appreciated   Paul A Brown
• [Asterisk-Users] Cisco 7960 - Have to press a menu button to dial   Paul A Brown
• [Asterisk-Users] Cisco 7960 - Have to press a menu button to dial   Paul A Brown
• [Asterisk-Users] Same CID on multiple users(friends9 in SIP.conf   Bruce
• [Asterisk-Users] Asterisk Users Mailing List Traffic   Matt Bruce
• [Asterisk-Users] hang up when pickup analog phone   Paco Brufal
• [Asterisk-Users] Polycom 501 and single call only using AAH 2.2   Rolf Brusletto
• [Asterisk-Users] Upgrading AAH   Rolf Brusletto
• [Asterisk-Users] Receiving Multiple calls on asterisk at home   Rolf Brusletto
• [Asterisk-Users] ZapATA channels up, but calls cannot be made   Mark Buckaway
• [Asterisk-Users] No ring when doing blind transfer.   Chuck Bunn
• [Asterisk-Users] Music on hold volume too high - using built in music on hold.   Chuck Bunn
• [Asterisk-Users] Zaptel not compiling on lastest Centos 4.2 kernel.   Chuck Bunn
• [Asterisk-Users] Failed installing zaptel   Chuck Bunn
• [Asterisk-Users] Failed installing zaptel   Chuck Bunn
• Spam? Re: [Asterisk-Users] Failed installing zaptel   Chuck Bunn
• [Asterisk-Users] Failed installing zaptel   Chuck Bunn
• [Asterisk-Users] All calls in queue go to agent that is down??   Chuck Bunn
• [Asterisk-Users] How to set priority for SJPhone.   Chuck Bunn
• [Asterisk-Users] Is it possible to turn off password for transfers on FOP   Chuck Bunn
• [Asterisk-Users] Voice mail not working with Asteriks 1.2.5   Chuck Bunn
• [Asterisk-Users] How to make groups of extensions ???   Chuck Bunn
• [Asterisk-Users] WiFi phones and WDS (Wireless Distribution System)   Chuck Bunn
• [Asterisk-Users] How to make groups of extensions ???   Chuck Bunn
• [Asterisk-Users] How to make groups of extensions ???   Chuck Bunn
• [Asterisk-Users] Bluetooth headset with SJPhone   Chuck Bunn
• [Asterisk-Users] Bluetooth headset in handsfree mode with SJPhone or X-lite   Chuck Bunn
• [Asterisk-Users] Bluetooth headset in handsfree mode with SJPhoneor X-lite   Chuck Bunn
• [Asterisk-Users] Bluetooth headset in handsfree modewith SJPhoneor X-lite   Chuck Bunn
• [Asterisk-Users] Bluetooth headset in handsfree modewith SJPhoneorX-lite   Chuck Bunn
• [Asterisk-Users] Bluetooth stack for cordless telephone   Chuck Bunn
• [Asterisk-Users] Wrong extension indicated when logging in as agent   Chuck Bunn
• [Asterisk-Users] Sjphone looses registry in my Asterisk Server then i need to restart Sjphone PC, why?   Chuck Bunn
• [Asterisk-Users] Sjphone looses registry in my Asterisk Server then i need to restart Sjphone PC, why?   Chuck Bunn
• [Asterisk-Users] BUG: FOP reports incorrect (duplicate) IP address until restarted   Chuck Bunn
• [Asterisk-Users] I have debug off why are the logs show debug info   Chuck Bunn
• [Asterisk-Users] I have debug off why are the logs show debug info   Chuck Bunn
• [Asterisk-Users] Asterisk Realtime voicemail question   Leo Burd
• [Asterisk-Users] Using Voicemail with MP3 files...   Leo Burd
• [Asterisk-Users] VERY IMPORTANT(TREAT WITH URGENCY)   Alexander Burke
• [Asterisk-Users] 7970 SIP Firmware; SIP 8.2 for 7940/7960   Alexander Burke
• [Asterisk-Users] Two FXOs getting bridged?   Warren Burstein
• [Asterisk-Users] PAP2 won't make two g729 calls at the same time   Warren Burstein
• [Asterisk-Users] Using softphone from a remote location to get into *   Leonard Burton
• [Asterisk-Users] T1 > GSM or DECT ? Searching for a complete microcell solution   Claude C
• [Asterisk-Users] GXP-2000 Volume Issue   Paul C
• [Asterisk-Users] Setting Max Calls on an IAX trunk   Paul C
• [Asterisk-Users] Australian E1 from Optus still not working   Paul C
• [Asterisk-Users] incoming calls dropout on PRI over TE110p   Paul C
• [Asterisk-Users] digium.com redesign   Ross C
• [Asterisk-Users] ENUM lookup issues with e164.org   Scott Call
• [Asterisk-Users] ENUM lookup issues with e164.org   Scott Call
• [Asterisk-Users] PRI DMS100 -> Nortel Meridian Option 81   Greg Camp
• [Asterisk-Users] PRI DMS100 -> Nortel Meridian Option 81   Greg Camp
• [Asterisk-Users] PRI DMS100 -> Nortel Meridian Option 81   Greg Camp
• [Asterisk-Users] Asterisk as Voicemail Server for Option 61c?   Greg Camp
• [Asterisk-Users] indications & SIP   Can2002
• [Asterisk-Users] indications & SIP   Can2002
• [Asterisk-Users] indications & SIP   Can2002
• [Asterisk-Users] DTMF digits lost on TDM400 card   Can2002
• [Asterisk-Users] Asterisk on hosted server   Can2002
• [Asterisk-Users] Zaptel compilation problem on SUSE   Can2002
• [Asterisk-Users] Oh this is bad.... bindaddr and rtp traffic   Brian Capouch
• [Asterisk-Users] Stuck. Extenions.conf? Realtime? MySQL? Grrrrr!   Brian Capouch
• [Asterisk-Users] How to make groups of extensions ???   Brian Capouch
• [Asterisk-Users] problems with DTMF   Brian Capouch
• [Asterisk-Users] Realtime Query   Brian Capouch
• [Asterisk-Users] Fw: anybody has SIP realtime working ?   Brian Capouch
• [Asterisk-Users] RE: IAX Incoming/Outgoing   Brian Capouch
• [Asterisk-Users] RE: IAX Incoming/Outgoing   Brian Capouch
• [Asterisk-Users] Preferred editor(s) dialplan coding?   JP Carballo
• [Asterisk-Users] Preferred editor(s) dialplan coding?   JP Carballo
• [Asterisk-Users] Info about mp3 which are installed with Asterisk   JP Carballo
• [Asterisk-Users] Call and then play IVR   JP Carballo
• [Asterisk-Users] PSTN to Asterisk VOIP in Manila   JP Carballo
• [Asterisk-Users] AstCC   JP Carballo
• [Asterisk-Users] AstCC   JP Carballo
• [Asterisk-Users] AstCC   JP Carballo
• [Asterisk-Users] AstCC   JP Carballo
• [Asterisk-Users] FW: I need to set NO CRC4 on zaptel.conf?   Oscar Carriles
• [Asterisk-Users] login/logout agents in a specific queue   Adrian Carter
• [Asterisk-Users] login/logout agents in a specific queue   Adrian Carter
• [Asterisk-Users] Problem compiling zaptel on latest RHEL kernel (2.6.9-34.EL)   Adrian Carter
• [Asterisk-Users] Multiple IAX clients behind a firewall   Adrian Carter
• [Asterisk-Users] stop monitor on transfer   Adrian Carter
• [Asterisk-Users] stop monitor on transfer   Adrian Carter
• [Asterisk-Users] stop monitor on transfer   Adrian Carter
• [Asterisk-Users] OT call recording   Adrian Carter
• [Asterisk-Users] RE: Multiple IAX clients behind a firewall   Adrian Carter
• [Asterisk-Users] RE: Multiple IAX clients behind a firewall   Adrian Carter
• [Asterisk-Users] invalid wav gsm frame size: 1 bytes ??   Adrian Carter
• [Asterisk-Users] Re: digium.com redesign   Adrian Carter
• [Asterisk-Users] Echo/Milliwatt Test Numbers in Oz ?   Adrian Carter
• [Asterisk-Users] Re: Milliwatt Analyzer available   Juan Carlos Castro y Castro
• [Asterisk-Users] Toshiba DK424 / Asterisk / DTMF problems   Anthony Cennami
• [Asterisk-Users] PRI channel hangs after BUSY   Anthony Cennami
• [Asterisk-Users] setmusiconhold doesn't work between 2 SIP phones   Albert Chaffman
• [Asterisk-Users] Sipura SPA-3000 and PSTN dtmf   Arsen Chaloyan
• [Asterisk-Users] Re: Sipura SPA-3000 and PSTN dtmf   Arsen Chaloyan
• [Asterisk-Users] Asterisk to receive fax   Gidean Chan
• [Asterisk-Users] Asterisk to receive fax   Gidean Chan
• [Asterisk-Users] external modem   Gidean Chan
• [Asterisk-Users] Asterisk to receive fax   Gidean Chan
• [Asterisk-Users] Unable to start Asterisk 1.2.5 with Asterisk-Addons 1.2.1   Sharath Chandra
• [Asterisk-Users] Unable to start Asterisk 1.2.5 with Asterisk-Addons 1.2.1   Sharath Chandra
• [Asterisk-Users] Dial Out IVR   Sharath Chandra
• [Asterisk-Users] On ParkAndAnnounce and parking lot   Sharath Chandra
• [Asterisk-Users] WARNINGS For SIP call   Sharath Chandra
• [Asterisk-Users] Oneway Audio   Sharath Chandra
• [Asterisk-Users] Asterisk out of Media Path - Call Park   Sharath Chandra
• [Asterisk-Users] Does an entry in AstDB stay after reboot?   Min Hwan Chang
• [Asterisk-Users] Faxing with MFC/r2   Carlos Chavez
• [Asterisk-Users] Unicall, Fax and Echo cancellation   Carlos Chavez
• [Asterisk-Users] Unicall and Fax detection   Carlos Chavez
• [Asterisk-Users] Diff between X100M and X100P?   Carlos Chavez
• [Asterisk-Users] Asterisk in the E1/R2 digital compelled enviroment   Carlos Chavez
• [Asterisk-Users] Iaxmodem speed limit?   Carlos Chavez
• [Asterisk-Users] Big Traffic anyway?   Alexander Chemeris
• [Asterisk-Users] 计划生育的无耻宣传该结束了   Jeffery Chen
• [Asterisk-Users] T38 fax pass thru to Cisco as53xx   Raymond Chen
• [Asterisk-Users] Bug Help or Suggestion - Grandstream GXP2000(firmware 1.0.2.8) - BLF, Hints, call-limit   Richard Cheung
• [Asterisk-Users] Wanted: IAX ATA w/ FXO   James Ching
• [Asterisk-Users] iax2_poke_noanswer on IP change. Sometimes permanent.   Ryan Chirches
• [Asterisk-Users] Asterisk on MacOS?   Christian
• [Asterisk-Users] parking slot lights - testers wanted   Dr. Michael J. Chudobiak
• [Asterisk-Users] IAX2 + Sonicwall   Dr. Michael J. Chudobiak
• [Asterisk-Users] IAX2 + Sonicwall   Dr. Michael J. Chudobiak
• [Asterisk-Users] pstn to asterisk, DVG-3004S, MP104?   Dr. Michael J. Chudobiak
• [Asterisk-Users] IAX2 + Sonicwall   Dr. Michael J. Chudobiak
• [Asterisk-Users] can't get TDM400P to answer   Dr. Michael J. Chudobiak
• [Asterisk-Users] Analog POTS line -> Rhino FXO Channel Bank -> No Hangup   Dr. Michael J. Chudobiak
• [Asterisk-Users] TDM400 FXO module not answering or dialing out.   Dr. Michael J. Chudobiak
• [Asterisk-Users] An FXO version of IAXy?   Dr. Michael J. Chudobiak
• [Asterisk-Users] TDM400 FXO module not answering or dialing out.   Dr. Michael J. Chudobiak
• [Asterisk-Users] best MTU?   Dr. Michael J. Chudobiak
• [Asterisk-Users] isdn out of band signalling   Piotr Chytla
• [Asterisk-Users] RE: [on-asterisk] containers, virtualization, and high availability   John Cianfarani
• [Asterisk-Users] RE: [on-asterisk] Brainstorming dual-core and Asterisk   John Cianfarani
• [Asterisk-Users] Unicall Question   Jorge Cisneros
• [Asterisk-Users] is there a variable for the calling IP ?   Simone Cittadini
• [Asterisk-Users] can't call some numbers/providers , ani code missing in sip header (found the problem, not the solution)   Simone Cittadini
• [Asterisk-Users] (unexplicable) peaks of machine load   Simone Cittadini
• [Asterisk-Users] (unexplicable) peaks of machine load   Simone Cittadini
• [Asterisk-Users] (unexplicable) peaks of machine load   Simone Cittadini
• [Asterisk-Users] Problem with chan_iax.c implimentationcausesbadaudio?   Simone Cittadini
• [Asterisk-Users] Double Call Progress tones   Simone Cittadini
• [Asterisk-Users] Polling Asterisk for Life   Mike Clark
• [Asterisk-Users] Polling Asterisk for Life   Mike Clark
• [Asterisk-Users] web meetme instructions   Mike Clark
• [Asterisk-Users] low call volume   Mike Clark
• [Asterisk-Users] HW Echo Cancellers   Mike Clark
• [Asterisk-Users] Location of MeetMe Recordings   Mike Clark
• [Asterisk-Users] Sangoma A200 error   Mike Clark
• [Asterisk-Users] Hardware timing source for MeetMe   Mike Clark
• [Asterisk-Users] Zaptel will not build   Mike Clark
• [Asterisk-Users] Can realtime extensions be used within AEL contexts?   Mike Clark
• [Asterisk-Users] asterisk management interface   Tzafrir Cohen
• [Asterisk-Users] Cannot boot machine up after working on zapt el....   Tzafrir Cohen
• [Asterisk-Users] A room full of Cisco 7960s behind NAT   Tzafrir Cohen
• [Asterisk-Users] How hard to create Asterisk for Compact Flash?   Tzafrir Cohen
• [Asterisk-Users] asterisk management interface   Tzafrir Cohen
• [Asterisk-Users] Child PID's   Tzafrir Cohen
• [Asterisk-Users] Preferred editor(s) dialplan coding?   Tzafrir Cohen
• [Asterisk-Users] fax receive using TDM400P   Tzafrir Cohen
• [Asterisk-Users] Preferred editor(s) dialplan coding?   Tzafrir Cohen
• [Asterisk-Users] Two asterisks on one machine   Tzafrir Cohen
• [Asterisk-Users] asterisk hang   Tzafrir Cohen
• [Asterisk-Users] Asterisk Native Sounds - in case you missed it...   Tzafrir Cohen
• [Asterisk-Users] Zaptel will not build   Tzafrir Cohen
• [Asterisk-Users] beronet & bristuff   Tzafrir Cohen
• [Asterisk-Users] Asterisk snapshots?   Tzafrir Cohen
• [Asterisk-Users] restart problem   Tzafrir Cohen
• [Asterisk-Users] Dial Out IVR   Dean Collins
• [Asterisk-Users] ring (hunt?) group   Dean Collins
• [Asterisk-Users] OT: Flash/web site developer in Boca Raton FL required   Dean Collins
• [Asterisk-Users] Re: Asterisk Question   Michael Collins
• [Asterisk-Users] Re: Asterisk Question   Michael Collins
• [Asterisk-Users] Re: Asterisk Question   Michael Collins
• [Asterisk-Users] Reverse group in zapata.conf   Michael Collins
• [Asterisk-Users] Oh this is bad.... bindaddr and rtp traffic   Michael Collins
• [Asterisk-Users] Dumb question (hang up detection/Zapata.conf)   Michael Collins
• [Asterisk-Users] Clustering   Michael Collins
• [Asterisk-Users] asterisk perl commands   Michael Collins
• [Asterisk-Users] Asterisk integration with office PBX   Michael Collins
• [Asterisk-Users] Programming the Manager API   Michael Collins
• [Asterisk-Users] simple perl-agi - where's the error?   Michael Collins
• [Asterisk-Users] RE: IAX Incoming/Outgoing   Joshua Colp
• [Asterisk-Users] RE: IAX Incoming/Outgoing   Joshua Colp
• [Asterisk-Users] Reload astdb?   Joshua Colp
• [Asterisk-Users] Reload astdb?   Joshua Colp
• [Asterisk-Users] Asterisk spanDSP / Faxing problem   Juan Jose Comellas
• [Asterisk-Users] re: Sixtel Services   VIC IP Communications
• [Asterisk-Users] re: Billing Package for Asterisk   VIC IP Communications
• [Asterisk-Users] Zap channel not hanging up   John Congdon
• [Asterisk-Users] Polycom 501 power over ethernet   William M Conlon
• [Asterisk-Users] Polycom 501 power over ethernet   William M Conlon
• [Asterisk-Users] Polycom 501 power over ethernet   William M Conlon
• [Asterisk-Users] Polycom 501 power over ethernet   William M Conlon
• [Asterisk-Users] OT: Polycom BootRom 3.1.3 and vsftpd 2.0.3 WARNING!!!   William M Conlon
• [Asterisk-Users] Asterisk Disconnecting after 30sec when someone leaving VM   William M Conlon
• [Asterisk-Users] SIP - Problem with audio clipping   William M Conlon
• [Asterisk-Users] Polycom 601 Message Center   William M Conlon
• [Asterisk-Users] Polycom 601 Message Center   William M Conlon
• [Asterisk-Users] Polycom 501 power over ethernet   The VoIP Connection
• [Asterisk-Users] An FXO version of IAXy?   The VoIP Connection
• [Asterisk-Users] Any Polycom dealer willing to help?   The VoIP Connection
• [Asterisk-Users] Asterisk with Vonage   The VoIP Connection
• [Asterisk-Users] PRI not answering call after asterisk upgrade   Derek Conniffe
• [Asterisk-Users] Polycom 4000 results?   Tim Connolly
• [Asterisk-Users] Cisco 7960 8.2 callerID lists proxy?   Tim Connolly
• [Asterisk-Users] Cisco 7960 8.2 callerID lists proxy?   Tim Connolly
• [Asterisk-Users] Cisco 7960 8.2 callerID lists proxy?   Tim Connolly
• [Asterisk-Users] Dialplan : Forwarding call to voicemail after onering iif extension is busy   Tim Connolly
• [Asterisk-Users] Cisco 7960 8.2 callerID lists proxy?   Tim Connolly
• [Asterisk-Users] TAC Case Cisco 7960 Proxy address showing up in callerID   Tim Connolly
• [Asterisk-Users] Asterisk to MySQL Data Lookup Warning Message?   Tim Connolly
• [Asterisk-Users] Dial from php   Andre Courchesne - Consultant
• [Asterisk-Users] Play wav while in connection with a caller   Andre Courchesne - Consultant
• [Asterisk-Users] Polling Asterisk for Life   David Cook
• [Asterisk-Users] Zoom 5801 problems with *   Sean Cook
• [Asterisk-Users] web meetme instructions   Sean Cook
• [Asterisk-Users] Sip Realtime Configs Samples with MySQL   Sean Cook
• [Asterisk-Users] Echo Cancelation on TE110P   Sean Cook
• [Asterisk-Users] Receiving Multiple calls on asterisk at home   Sean Cook
• [Asterisk-Users] Real Time Asterisk   Sean Cook
• [Asterisk-Users] Real Time Asterisk   Sean Cook
• [Asterisk-Users] Merlin Magix Integration   Sean Cook
• [Asterisk-Users] IVR woes   Sean Cook
• [Asterisk-Users] Voicetronix OpenSwitch / Sangoma Analog Card   Sean Cook
• [Asterisk-Users] Re: transfers/parked calls + polycom 501   Sean Cook
• [Asterisk-Users] Re: transfers/parked calls + polycom 501   Sean Cook
• [Asterisk-Users] Disappearing voicemail   Sean Cook
• [Asterisk-Users] High Density Analog   Sean Cook
• [Asterisk-Users] I'm FED UP with BroadVoice   Sean Cook
• [Asterisk-Users] RealTime Extensions   Sean Cook
• [Asterisk-Users] Asterisk & Avaya Legend   Sean Cook
• [Asterisk-Users] [OT] Polycom provisioning   Sean Cook
• [Asterisk-Users] Stability of Asterisk with 2 x TDM400P cards (6 analogue lines)   Sean Cook
• [Asterisk-Users] Problem with MeetMe Conference!!!   Sean Cook
• [Asterisk-Users] Asterisk & Avaya Legend   Sean Cook
• [Asterisk-Users] Hints in Realtime   Sean Cook
• [Asterisk-Users] definity g3 voicemail   Sean Cook
• [Asterisk-Users] definity g3 voicemail   Sean Cook
• [Asterisk-Users] definity g3 voicemail   Sean Cook
• [Asterisk-Users] MixMonitor Problems -- sssshh, don't be too loud   Johnathan Corgan
• [Asterisk-Users] MixMonitor Problems -- sssshh, don't be too loud   Johnathan Corgan
• [Asterisk-Users] Calls forwarding to numbers only in user's context   Johnathan Corgan
• [Asterisk-Users] SPA3000 and callerID   Dave Cotton
• [Asterisk-Users] Voip-Info   Dave Cotton
• [Asterisk-Users] Zaptel compile errors on x86_64   Dave Cotton
• [Asterisk-Users] Aastra Questions   Dave Cotton
• [Asterisk-Users] ISDN Protocol Unknom Error with Junghanns OctoBRI   Dave Cotton
• [Asterisk-Users] ISDN Protocol Unknom Error with Junghanns OctoBRI   Dave Cotton
• [Asterisk-Users] Mandrake zaptel module not found after compiling   Dave Cotton
• [Asterisk-Users] Error in starting * with latest trunk   Dave Cotton
• [Asterisk-Users] Error in starting * with latest trunk   Dave Cotton
• [Asterisk-Users] indications.conf.sample   Dave Cotton
• [Asterisk-Users] Clustering   David Coulson
• [Asterisk-Users] FreePBX 2.0.1 released!   Ryan Courtnage
• [Asterisk-Users] Hung IAX Channels   Jonathan k. Creasy
• [Asterisk-Users] OT: Unblocking bloced CID   Jonathan k. Creasy
• [Asterisk-Users] OT: Unblocking bloced CID   Jonathan k. Creasy
• [Asterisk-Users] Problem with intermittent one-way audio   Jonathan k. Creasy
• [Asterisk-Users] FAX over PRI   Jonathan k. Creasy
• [Asterisk-Users] Multiple commands per priority   Jonathan k. Creasy
• [Asterisk-Users] Re: OT: Unblocking bloced CID   Jonathan k. Creasy
• [Asterisk-Users] Polycom IP 301 is slow   Jonathan k. Creasy
• [Asterisk-Users] Dumb question - reaching the PSTN   Jonathan k. Creasy
• [Asterisk-Users] Avoiding initial deadlock on iax?   Jonathan k. Creasy
• [Asterisk-Users] registration with different username   Jonathan k. Creasy
• [Asterisk-Users] Quintum Tenor DX4060   Jonathan k. Creasy
• [Asterisk-Users] Asterisk and Hylafax, on the same box   Jonathan k. Creasy
• [Asterisk-Users] Clustering   Matthew Crocker
• [Asterisk-Users] problem with tdm22b   Diego Quintana Cruz
• [Asterisk-Users] Status of another channel from AGI   Alistair Cunningham
• [Asterisk-Users] Asterisk download file locations   Alistair Cunningham
• [Asterisk-Users] Asterisk download file locations   Alistair Cunningham
• [Asterisk-Users] No audio on PRI.   Ken D'Ambrosio
• [Asterisk-Users] Initiate and monitor multiple calls?   Ken D'Ambrosio
• [Asterisk-Users] Polycom 501 power over ethernet   Ken D'Ambrosio
• [Asterisk-Users] OT: Polycom BootRom 3.1.3 and vsftpd 2.0.3   Ken D'Ambrosio
• [Asterisk-Users] OT: Polycom BootRom 3.1.3 and vsftpd 2.0.3   Ken D'Ambrosio
• [Asterisk-Users] ip telephony project   JOSE MANUEL CORTES DAVID
• [Asterisk-Users] problem configuring a digium quad E1 card   JOSE MANUEL CORTES DAVID
• [Asterisk-Users] cards   JOSE MANUEL CORTES DAVID
• [Asterisk-Users] definity prologix   JOSE MANUEL CORTES DAVID
• [Asterisk-Users] (no subject)   JOSE MANUEL CORTES DAVID
• [Asterisk-Users] How to transmit Video   JOSE MANUEL CORTES DAVID
• [Asterisk-Users] TE 205P/A102 fit in hp dc7600?   JOSE MANUEL CORTES DAVID
• [Asterisk-Users] Re: res_features pickupexten   DRi at b-w-computer.de
• [Asterisk-Users] misdn <--> zap problem   DRi at b-w-computer.de
• [Asterisk-Users] misdn problem   DRi at b-w-computer.de
• [Asterisk-Users] misdn problem   DRi at b-w-computer.de
• [Asterisk-Users] hfc-pci cards on ppc   DRi at b-w-computer.de
• [Asterisk-Users] Problem compiling ztdummy on centos 4, 2.6 kernel   Bart van Daal
• ***SPAM*** Re: [Asterisk-Users] D-Link DVG-1402S   Gerald Dachs
• [Asterisk-Users] get no connection, very often, but not allways, why?   Gerald Dachs
• [Asterisk-Users] Re: sipura 841 mass provisioning   Josh Dady
• [Asterisk-Users] Call Pickup Woes   Adam Dale
• [Asterisk-Users] Call Pickup Woes   Adam Dale
• [Asterisk-Users] Call Pickup Woes   Adam Dale
• [Asterisk-Users] Call Pickup Woes   Adam Dale
• [Asterisk-Users] Call Pickup Woes   Adam Dale
• [Asterisk-Users] Call Pickup Woes   Adam Dale
• [Asterisk-Users] Call Pickup Woes   Adam Dale
• [Asterisk-Users] realtime and queues and persistantmembers in 1.2.5   Glenn Dalgliesh
• [Asterisk-Users] Panasonic KX-TDA1000 with asterisk server   Daniel
• [Asterisk-Users] the best configuration for DTMF detection on SPA 2000   Daniel
• [Asterisk-Users] automatic callback when busy   Daniel
• [Asterisk-Users] blacklisting   Daniel
• [Asterisk-Users] Asterisk, QSIG and Tenovis PBX?   Daniel
• [Asterisk-Users] TE411P VPM   Aaron Daniel
• [Asterisk-Users] TE411P VPM   Aaron Daniel
• [Asterisk-Users] TE411P VPM   Aaron Daniel
• [Asterisk-Users] NEWS: SIP Firmware Available for Cisco 7970   Aaron Daniel
• [Asterisk-Users] More 7940 Questions   Aaron Daniel
• [Asterisk-Users] 7940/60 SIP 8.2   Aaron Daniel
• [Asterisk-Users] Cisco 7960 SIP - Displaying Time   Aaron Daniel
• [Asterisk-Users] 7940/60 SIP 8.2   Aaron Daniel
• [Asterisk-Users] 7940/60 SIP 8.2   Aaron Daniel
• [Asterisk-Users] Dial plans and forwarded phones   Aaron Daniel
• [Asterisk-Users] Dial plans and forwarded phones   Aaron Daniel
• [Asterisk-Users] 7970 Configs   Aaron Daniel
• [Asterisk-Users] 7970 Configs   Aaron Daniel
• [Asterisk-Users] Cisco 7960 8.2 callerID lists proxy?   Aaron Daniel
• [Asterisk-Users] Phase locked mode   Aaron Daniel
• [Asterisk-Users] Cisco 7960 8.2 callerID lists proxy?   Aaron Daniel
• [Asterisk-Users] Realtime SIP   Aaron Daniel
• [Asterisk-Users] OT - force Cisco phones to reboot   Aaron Daniel
• [Asterisk-Users] Realtime Extensions   Aaron Daniel
• [Asterisk-Users] Realtime Extensions   Aaron Daniel
• [SPAM] Re: [Asterisk-Users] OT - force Cisco phones to reboot   Aaron Daniel
• [Asterisk-Users] Feedback from VON expo! Info on * HA and Polycomphone!!   Aaron Daniel
• [Asterisk-Users] Asterisk Users Mailing List Traffic   Aaron Daniel
• [Asterisk-Users] Aterisk with Realtime   Aaron Daniel
• [Asterisk-Users] Aterisk with Realtime   Aaron Daniel
• [Asterisk-Users] Feedback from VON expo! Info on * HAandPolycomphone!!   Aaron Daniel
• [Asterisk-Users] Feedback from VON expo! Info on *HAandPolycomphone!!   Aaron Daniel
• [Asterisk-Users] Asterisk Users Mailing List Traffic   Aaron Daniel
• [Asterisk-Users] Aterisk with Realtime   Aaron Daniel
• [Asterisk-Users] How to make groups of extensions ???   Aaron Daniel
• [Asterisk-Users] Aterisk with Realtime   Aaron Daniel
• [Asterisk-Users] Fw: anybody has SIP realtime working ?   Aaron Daniel
• [Asterisk-Users] Cisco 7960 - Have to press a menu button to dial   Aaron Daniel
• [Asterisk-Users] [OT] Cisco 7970 SCCP Image   Aaron Daniel
• [Asterisk-Users] Modem Bank   Aaron Daniel
• [Asterisk-Users] Fw: anybody has SIP realtime working ?   Aaron Daniel
• [Asterisk-Users] 'Click to Dial'   Aaron Daniel
• [Asterisk-Users] Cisco 7970 SIP Image   Aaron Daniel
• [Asterisk-Users] Fw: anybody has SIP realtime working ?   Aaron Daniel
• [Asterisk-Users] Asterisk Users Mailing List Traffic   Aaron Daniel
• [Asterisk-Users] Fw: anybody has SIP realtime working ?   Aaron Daniel
• [Asterisk-Users] Realtime / SIP Peers etc   Aaron Daniel
• [Asterisk-Users] Realtime Query   Aaron Daniel
• [Asterisk-Users] Big Traffic anyway?   Aaron Daniel
• [Asterisk-Users] RE: Asterisk Users   Aaron Daniel
• [Asterisk-Users] 7970 SIP Firmware; SIP 8.2 for 7940/7960   Aaron Daniel
• [Asterisk-Users] 7970 SIP Firmware; SIP 8.2 for 7940/7960   Aaron Daniel
• [Asterisk-Users] Re: Fw: anybody has SIP realtime working ?   Aaron Daniel
• [Asterisk-Users] Asterisk Users   Aaron Daniel
• [Asterisk-Users] Ok... what is 'sip show peers' really used for?   Aaron Daniel
• [Asterisk-Users] Realtime Query   Aaron Daniel
• [Asterisk-Users] Ok... what is 'sip show peers' really used for?   Aaron Daniel
• [Asterisk-Users] How to create [new_context] in extensions.conf?   Aaron Daniel
• [Asterisk-Users] Fw: anybody has SIP realtime working ?   Aaron Daniel
• [Asterisk-Users] TAC Case Cisco 7960 Proxy address showing up in callerID   Aaron Daniel
• [Asterisk-Users] [OT] Polycom provisioning   Aaron Daniel
• [Asterisk-Users] How to create [new_context] in extensions.conf?   Aaron Daniel
• [Asterisk-Users] [OT] Polycom provisioning   Aaron Daniel
• [Asterisk-Users] Fw: anybody has SIP realtime working ?   Aaron Daniel
• [Asterisk-Users] [OT] Polycom provisioning   Aaron Daniel
• [Asterisk-Users] Fw: anybody has SIP realtime working ?   Aaron Daniel
• [Asterisk-Users] Fw: anybody has SIP realtime working ?   Aaron Daniel
• [Asterisk-Users] Fw: anybody has SIP realtime working ?   Aaron Daniel
• [Asterisk-Users] [OT] Polycom provisioning   Aaron Daniel
• [Asterisk-Users] Asterisk Users   Aaron Daniel
• FW: [Asterisk-Users] Asterisk Users   Aaron Daniel
• FW: [Asterisk-Users] Asterisk Users   Aaron Daniel
• [Asterisk-Users] Asterisk Failover without SER   Aaron Daniel
• [Asterisk-Users] Asterisk Failover without SER   Aaron Daniel
• [Asterisk-Users] Re: [OT] Polycom provisioning   Aaron Daniel
• [Asterisk-Users] Transferring a call with IAX   Aaron Daniel
• [Asterisk-Users] Asterisk Failover without SER   Aaron Daniel
• [Asterisk-Users] Transferring a call with IAX   Aaron Daniel
• [Asterisk-Users] Asterisk Failover without SER   Aaron Daniel
• [Asterisk-Users] Transferring a call with IAX   Aaron Daniel
• [Asterisk-Users] Transferring a call with IAX   Aaron Daniel
• [Asterisk-Users] Transferring a call with IAX   Aaron Daniel
• [Asterisk-Users] Transferring a call with IAX   Aaron Daniel
• [Asterisk-Users] Transferring a call with IAX   Aaron Daniel
• [Asterisk-Users] Transferring a call with IAX   Aaron Daniel
• [Asterisk-Users] Transferring a call with IAX   Aaron Daniel
• [Asterisk-Users] Transferring a call with IAX   Aaron Daniel
• [Asterisk-Users] Polycom 601 Message Center   Aaron Daniel
• [Asterisk-Users] Re: Cisco 7960 - Have to press a menu button to dial   Aaron Daniel
• [Asterisk-Users] Polycoms and hints   Aaron Daniel
• [Asterisk-Users] How to send announcement after called has picked up the phone?   Aaron Daniel
• [Asterisk-Users] asterisk doesn't wait for whole extension   Aaron Daniel
• [Asterisk-Users] 'sip show users' shows NAT RFC3581   Aaron Daniel
• [Asterisk-Users] Problem compiling ztdummy on centos 4, 2.6 kernel   John Daragon
• [Asterisk-Users] Operator consoles for large systems   John Daragon
• [Asterisk-Users] Cannot load wcfxo -- Please help!   John Daragon
• [Asterisk-Users] Diff between X100M and X100P?   John Daragon
• [Asterisk-Users] 10minutes to restart Asterisk@home 2.7   John Daragon
• [Asterisk-Users] AVM C2 chan_capi-cm-0.6.3 Error on Dial   John Daragon
• [Asterisk-Users] AVM C2 chan_capi-cm-0.6.4 Error on Dial   John Daragon
• [Asterisk-Users] AVM C2 chan_capi-cm-0.6.4 Error on Dial   John Daragon
• [Asterisk-Users] Idiot's guide to Q.932?   John Daragon
• [Asterisk-Users] ISDN BRI and UK Premium Rate Numbers   John Daragon
• [Asterisk-Users] MixMonitor and transferred calls   John Daragon
• [Asterisk-Users] answer delay   John Daragon
• [Asterisk-Users] answer delay   John Daragon
• [Asterisk-Users] stop monitor on transfer   John Daragon
• [Asterisk-Users] stop monitor on transfer   John Daragon
• [Asterisk-Users] MFCR2   Darlon
• [Asterisk-Users] Asterisk Disconnecting after 30sec when someone leaving VM   Dave
• [Asterisk-Users] Re: Echo Cancelation on TE110P   Paul Davidson
• [Asterisk-Users] Re: System Design   Paul Davidson
• [Asterisk-Users] Dell 2850 w/TDM2400?   Mark Davies
• [Asterisk-Users] HW Echo Cancellers   Steve Davies
• [Asterisk-Users] Single E1 with HW Echo Can?   Steve Davies
• [Asterisk-Users] IVR weirdness   Steve Davies
• [Asterisk-Users] Echo canceller data-points   Steve Davies
• [Asterisk-Users] Echo canceller data-points   Steve Davies
• [Asterisk-Users] Echo canceller data-points   Steve Davies
• [Asterisk-Users] Echo canceller data-points   Steve Davies
• [Asterisk-Users] Snom 360 Hinting tricks   Steve Davies
• [Asterisk-Users] callerid= in zapata.conf   Steve Davies
• [Asterisk-Users] Zap<-->IAX codec?   Steve Davies
• [Asterisk-Users] Snom 360 Hinting tricks   Steve Davies
• [Asterisk-Users] Snom 360 Hinting tricks   Steve Davies
• [Asterisk-Users] Echo cancellation   Steve Davies
• R: [Asterisk-Users] Echo cancellation   Steve Davies
• [Asterisk-Users] Sipura RMA   Karl Davis
• [Asterisk-Users] Snom 360 Hinting tricks   Jared Davison
• [Asterisk-Users] Re: Subscription state after reload (New subject)   Jared Davison
• [Asterisk-Users] Stability of Asterisk with 2 x TDM400P cards (6 analogue lines)   Jared Davison
• [Asterisk-Users] Why would asterisk presume a loop (482 "Loop Detected")?   Peter J Dean
• [Asterisk-Users] Festival tts   Rusty Dekema
• [Asterisk-Users] Codec Issue   Rusty Dekema
• [Asterisk-Users] OSHA requirement to "reach a live human" ??   Rusty Dekema
• [Asterisk-Users] Asterisk 1.2.6 on Solaris 8?   Rusty Dekema
• [Asterisk-Users] Re: Asterisk 1.2.6 on Solaris 8?   Rusty Dekema
• [Asterisk-Users] GUI Web interface   Joe Dennick
• [Asterisk-Users] Tearing my hair out with Queues   Joe Dennick
• [Asterisk-Users] Reporting?   Joe Dennick
• [Asterisk-Users] Config TE110P and TDM400 with 2 FXS modules   Dennis
• [Asterisk-Users] Config TE110P and TDM400 with 2 FXS modules   Dennis
• [Asterisk-Users] TDM400 FXO module not answering or dialing out.   Mike Dent
• [Asterisk-Users] Which Mac OSX softphone with IAX2 support?   Mike Dent
• [Asterisk-Users] VERY IMPORTANT(TREAT WITH URGENCY)   Austin Denyer
• [Asterisk-Users] Problem with two cards Digium   Saul Diaz
• [Asterisk-Users] RE: Predictive Dialer   Saul Diaz
• [Asterisk-Users] IAX2 + Sonicwall   Saul Diaz
• [Asterisk-Users] ISDN Protocol Unknom Error with Junghanns OctoBRI   Henk Dick
• [Asterisk-Users] Asterisk and Quintum AXT800 Gateway   Dofear
• [Asterisk-Users] OT: Snom 320, displaying text on the screen from *   Dofear
• [Asterisk-Users] Development news :: T38 passthrough support   Ray Van Dolson
• [Asterisk-Users] SCCP problem with ATA188, Asterisk@home and chan_sccp   Dontigny, Jean-Francois
• [Asterisk-Users] Speakeasy VOIP + Asterisk?   Simon Dorfman
• [Asterisk-Users] Anybody know about Cisco VOIP routers?   Doug
• [Asterisk-Users] Error in starting * with latest trunk   Paradise Dove
• [Asterisk-Users] Ringduration problem when calling out via Sip   Philipp Dreimann
• AW: [Asterisk-Users] Asterisk to receive fax   Philipp Dreimann
• [Asterisk-Users] help on mfc/r2   Krzysztof Drewicz
• [Asterisk-Users] Asterisk billing from CDR database   Krzysztof Drewicz
• [Asterisk-Users] help on mfc/r2   Krzysztof Drewicz
• [Asterisk-Users] help on mfc/r2   Krzysztof Drewicz
• [Asterisk-Users] help on mfc/r2   Krzysztof Drewicz
• [Asterisk-Users] help on mfc/r2   Krzysztof Drewicz
• [Asterisk-Users] help on mfc/r2   Krzysztof Drewicz
• [Asterisk-Users] help on mfc/r2   Krzysztof Drewicz
• [Asterisk-Users] Stability of Asterisk with 2 x TDM400P cards (6analogue lines)   Krzysztof Drewicz
• [Asterisk-Users] need to find an asterisk user from Costa Rica.   Dualcall.com
• [Asterisk-Users] CHINA DID   Steve Ducat
• [Asterisk-Users] Asterisk to a Huawei softX3000   Steve Ducat
• [Asterisk-Users] Outgoing SIP Failover   Steve Ducat
• [Asterisk-Users] TIMESTAMP, DATETIME not working   Corporate IT Solutions - Michael Dunne
• [Asterisk-Users] TIMESTAMP, DATETIME not working   Corporate IT Solutions - Michael Dunne
• [Asterisk-Users] Bug Help or Suggestion - Grandstream GXP2000 (firmware 1.0.2.8) - BLF, Hints, call-limit   Corporate IT Solutions - Michael Dunne
• [Asterisk-Users] Countries supporting SMS on PSTN (ISDN)   Corporate IT Solutions - Michael Dunne
• [Asterisk-Users] Jittery meetme conference using Linksys 942 phones   Rana Dutt
• [Asterisk-Users] Jittery Linksys/Sipura meetme conference fixed   Rana Dutt
• [Asterisk-Users] Re: Junghanns and Digium TDM400?   Chris Earle
• [Asterisk-Users] Re: Junghanns and Digium TDM400?   Chris Earle
• [Asterisk-Users] Re: BRI cards, HFC, and bristuff - a general questionto clear up my understanding.   Chris Earle
• [Asterisk-Users] Re: BRI cards, HFC, and bristuff - a general questionto clear up my understanding.   Chris Earle
• [Asterisk-Users] Toshiba Strata DK-280 support?   Philip Edelbrock
• [Asterisk-Users] Asterisk with Vonage   Philip Edelbrock
• [Asterisk-Users] Call Recording?   Kris Edwards
• [Asterisk-Users] Setting Max Calls on an IAX trunk   Mark Edwards
• [Asterisk-Users] No DTMF   Mark Edwards
• [Asterisk-Users] Outgoing calls via Sipgate   Christoph Eicke
• [Asterisk-Users] Squished faxes with txfax   Scott Eisert
• [Asterisk-Users] Squished faxes with txfax   Scott Eisert
• [Asterisk-Users] Two FXOs getting bridged?   Dan Elder
• [Asterisk-Users] Re: Two FXOs getting bridged?   Dan Elder
• [Asterisk-Users] Disable flash transfers?   Dan Elder
• [Asterisk-Users] Disable flash transfers?   Dan Elder
• [Asterisk-Users] 3 way calls & transfers   Dan Elder
• [Asterisk-Users] caller unable to transfer   Dan Elder
• [Asterisk-Users] Transfer problems revisited   Dan Elder
• [Asterisk-Users] Call transfer problems, SOLVED   Dan Elder
• [Asterisk-Users] AAH Mailing list   Dan Elder
• [Asterisk-Users] Merlin Magix Integration   Darren Ellis
• [Asterisk-Users] Re: Subscription state after reload (New subject)   My Other Email
• [Asterisk-Users] Agents, queues and Pentalties   Erik
• [Asterisk-Users] SMS in Spain (it seems Protocol 2)   Carles Pina i Estany
• [Asterisk-Users] SMS in Spain (it seems Protocol 2)   Carles Pina i Estany
• [Asterisk-Users] SMS in Spain (it seems Protocol 2)   Carles Pina i Estany
• [Asterisk-Users] PRI CID signalling not working?   Damon Estep
• [Asterisk-Users] impact of qualify=yes   Damon Estep
• [Asterisk-Users] incoming limit, call_limit, or call-limit?   Damon Estep
• [Asterisk-Users] CDR Bug?   Damon Estep
• [Asterisk-Users] CDR Bug?   Damon Estep
• [Asterisk-Users] dropping voice frame ulaw - slin?   Damon Estep
• [Asterisk-Users] ODBC voicemail storage   Damon Estep
• [Asterisk-Users] asterisk management interface   Dumpolid Exeplish
• [Asterisk-Users] what version s this??   Dumpolid Exeplish
• [Asterisk-Users] Memory Problems   Dumpolid Exeplish
• [Asterisk-Users] Memory Problems   Dumpolid Exeplish
• [Asterisk-Users] Question from a newbie on finding digium hosts   Gene Expression
• [Asterisk-Users] about operator   C F
• [Asterisk-Users] Re: Polycom Default Ring Volume [OT]   C F
• [Asterisk-Users] Re: sipura 841 mass provisioning   C F
• [Asterisk-Users] Re: Asterisk at large   C F
• [Asterisk-Users] Two FXOs getting bridged?   C F
• [Asterisk-Users] Re: sipura 841 mass provisioning   C F
• [Asterisk-Users] Re: Asterisk at large   C F
• [Asterisk-Users] Changing caller id on transfer   C F
• [Asterisk-Users] Preferred editor(s) dialplan coding?   C F
• [Asterisk-Users] *** Yet another boring weekend? Test new Asterisk features in development!   C F
• [Asterisk-Users] Polycom 501 power over ethernet   C F
• [Asterisk-Users] Variable   C F
• [Asterisk-Users] Oh this is bad.... bindaddr and rtp traffic   C F
• [Asterisk-Users] OT: Polycom Registration Weirdness   C F
• [Asterisk-Users] sending text to display of sip phones   C F
• [Asterisk-Users] OT: Polycom BootRom 3.1.3 and vsftpd 2.0.3   C F
• [Asterisk-Users] HW Echo Cancellers   C F
• [Asterisk-Users] 7940/60 SIP 8.2   C F
• [Asterisk-Users] 7940/60 SIP 8.2   C F
• [Asterisk-Users] Merlin Magix Integration   C F
• [Asterisk-Users] 7940/60 SIP 8.2   C F
• [Asterisk-Users] Extracting info from the $EXTEN variable   C F
• [Asterisk-Users] Disable flash transfers?   C F
• [Asterisk-Users] Asterisk Re-invites - how to tell ?   C F
• [Asterisk-Users] Disable flash transfers?   C F
• [Asterisk-Users] Development news :: T38 passthrough support   C F
• [Asterisk-Users] Limiting the number of concurrent calls for a group of SIP devices   C F
• [Asterisk-Users] Analog Desktop Phone   C F
• [Asterisk-Users] dipura 2002 auto dial or intercom   C F
• [Asterisk-Users] Menu in queue   C F
• [Asterisk-Users] Callerid on transfer   C F
• [Asterisk-Users] Asterisk large scale, help needed   C F
• [Asterisk-Users] Re: how to show called name on calling polycom display   C F
• [Asterisk-Users] how to show called name on calling polycomdisplay   C F
• [SPAM] Re: [Asterisk-Users] OT - force Cisco phones to reboot   C F
• [Asterisk-Users] Flash Operator Panel   C F
• [Asterisk-Users] New one on me: How to UN-transfer   C F
• [Asterisk-Users] OT: Unblocking bloced CID   C F
• [Asterisk-Users] OT: Unblocking bloced CID   C F
• [Asterisk-Users] OT: Unblocking bloced CID   C F
• [Asterisk-Users] OT: Unblocking bloced CID   C F
• [Asterisk-Users] Best budget IP phone at the moment?   C F
• [Asterisk-Users] Best budget IP phone at the moment?   C F
• [Asterisk-Users] how to get separate CDR for inbound and outbound legs of a call   C F
• [Asterisk-Users] Extra Debugging without console   C F
• [Asterisk-Users] Call Pickup Woes   C F
• [Asterisk-Users] Call Pickup Woes   C F
• [Asterisk-Users] Call Pickup Woes   C F
• [Asterisk-Users] Grabbing the billsec and duration after a hangup.   C F
• [Asterisk-Users] Caller ID forwarding with Pickup() application?   C F
• [Asterisk-Users] How to make groups of extensions ???   C F
• [Asterisk-Users] Phones were working fine - Now there is noaudiowhen calling between extensions   C F
• [Asterisk-Users] How to make groups of extensions ???   C F
• [Asterisk-Users] Phones were working fine - Now there is noaudiowhen calling between extensions   C F
• [Asterisk-Users] How to make groups of extensions ???   C F
• [Asterisk-Users] How to make groups of extensions ???   C F
• [Asterisk-Users] Cannot leave voicemail, Asterisk/Zaptel/libpi v1.0.9   C F
• [Asterisk-Users] Re: OT: Unblocking bloced CID   C F
• [Asterisk-Users] Multiple commands per priority   C F
• [Asterisk-Users] How to make groups of extensions ???   C F
• [Asterisk-Users] Ok... what is 'sip show peers' really used for?   C F
• [Asterisk-Users] connecting Avaya Partnet with asterisk , TE205P   C F
• [Asterisk-Users] High Density Analog   C F
• [Asterisk-Users] Ok... what is 'sip show peers' really used for?   C F
• [Asterisk-Users] How to create [new_context] in extensions.conf?   C F
• [Asterisk-Users] [OT] Polycom provisioning   C F
• [Asterisk-Users] transfer incoming call to VM without answering call   C F
• [Asterisk-Users] FXS channel banks   C F
• [Asterisk-Users] RE: Re: OT: Unblocking bloced CID   C F
• [Asterisk-Users] RE: Re: OT: Unblocking bloced CID   C F
• [Asterisk-Users] Call Waiting Issues   C F
• [Asterisk-Users] Caller ID length   C F
• [Asterisk-Users] definity g3 voicemail   C F
• [Asterisk-Users] definity g3 voicemail   C F
• [Asterisk-Users] Unable to authenticate password - VM   C F
• [Asterisk-Users] How to send announcement after called has picked up the phone?   C F
• [Asterisk-Users] Redirect problem/bug/feature   C F
• [Asterisk-Users] Regulatory Ruling about Caller-ID   C F
• [Asterisk-Users] Disable polycom call waiting?   C F
• [Asterisk-Users] Attended transfer returns invalid extension   Alexis FECOURT
• [Asterisk-Users] uniqueid   FaberK
• [Asterisk-Users] Re: uniqueid   FaberK
• [Asterisk-Users] Problem with 827-4v and asterisk as a pstn GW   FaberK
• [Asterisk-Users] answer delay   FaberK
• [Asterisk-Users] answer delay   FaberK
• [Asterisk-Users] answer delay   FaberK
• [Asterisk-Users] compiling chan_h323   FaberK
• [Asterisk-Users] Problems with wcte11xp module   Jon Farmer
• [Asterisk-Users] time update (7905)   Jon Farmer
• [Asterisk-Users] Problems with wcte11xp module   Jon Farmer
• [Asterisk-Users] Problems with wcte11xp module   Jon Farmer
• [Asterisk-Users] Configs for Gradwell and inWeb   Simon Faulkner
• [Asterisk-Users] Transferring a call with IAX   Jonathan Feally
• [Asterisk-Users] Stability of Asterisk with 2 x TDM400P cards (6 analogue lines)   Mike Fedyk
• [Asterisk-Users] Professional Recordings   Martinez Felix
• [Asterisk-Users] Professional Recordings   Martinez Felix
• [Asterisk-Users] Festival tts   Robert La Ferla
• [Asterisk-Users] Getting to the last "old" voicemail message   Robert La Ferla
• [Asterisk-Users] Re: Getting to the last "old" voicemail message   Robert La Ferla
• [Asterisk-Users] Looking for docs on adjusting txgain/rxgain   Robert La Ferla
• [Asterisk-Users] Asterisk Users Mailing List Traffic   Robert La Ferla
• [Asterisk-Users] Asterisk download file locations   Peter Fern
• [Asterisk-Users] Problem with chan_iax.c implimentation causes bad audio?   Peter Fern
• [Asterisk-Users] Problem with intermittent one-way audio   Peter Fern
• [Asterisk-Users] Problem with chan_iax.c implimentation causes bad audio?   Peter Fern
• [Asterisk-Users] CHINA DID   Peter Fern
• [Asterisk-Users] Dial out .call files File permissions??   Peter Fern
• [Asterisk-Users] Dropped calls   Peter Fern
• [Asterisk-Users] Re: TDM400P digium card   Alan Ferrency
• [Asterisk-Users] Re: sipura 841 mass provisioning   Alan Ferrency
• [Asterisk-Users] SendDTMF in connected call?   Alan Ferrency
• [Asterisk-Users] Re: TDM400P digium card   Alan Ferrency
• [Asterisk-Users] SIP: INFO before answer causes disconnect   Alan Ferrency
• [Asterisk-Users] Sending ANI to TDM40B FXS?   Bryan Field-Elliot
• [Asterisk-Users] Problem with two cards Digium   Bart Fisher
• [Asterisk-Users] OT: ADIT 600 Manual needed   Bart Fisher
• [Asterisk-Users] On site installtion Tech. wanted   Bart Fisher
• [Asterisk-Users] Re: a2billing without IVR   Barry Flanagan
• [Asterisk-Users] Problem with uac_replace and corrupted From   Barry Flanagan
• [Asterisk-Users] Attended Transfer - transfer timeout, how to change?   Barry Flanagan
• [Asterisk-Users] Attended transfers timing out after 3 rings   Barry Flanagan
• [Asterisk-Users] Attended Transfer - transfer timeout, how to change?   Barry Flanagan
• [Asterisk-Users] Problem with intermittent one-way audio   Barry Flanagan
• [Asterisk-Users] Problem with intermittent one-way audio   Barry Flanagan
• [Asterisk-Users] VoIP prepaid billing   Barry Flanagan
• [Asterisk-Users] Problem with intermittent one-way audio   Barry Flanagan
• [Asterisk-Users] Realtime SIP Persistency   Barry Flanagan
• [Asterisk-Users] VoIP prepaid billing   Barry Flanagan
• [Asterisk-Users] How to hide CallerID - SetCallerPres(prohib) not working   Barry Flanagan
• [Asterisk-Users] How to hide CallerID - SetCallerPres(prohib) not working   Barry Flanagan
• [Asterisk-Users] Zaptel will not build   Assaf Flatto
• [Asterisk-Users] TE411P VPM   Kevin P. Fleming
• [Asterisk-Users] Child PID's   Kevin P. Fleming
• [Asterisk-Users] PLEASE respond: how to get Asterisk to change coders on RTP handoff??   Kevin P. Fleming
• [Asterisk-Users] Polycom 501 power over ethernet   Kevin P. Fleming
• [Asterisk-Users] Asterisk download file locations   Kevin P. Fleming
• [Asterisk-Users] Is everyone getting mails except me?   Kevin P. Fleming
• [Asterisk-Users] [Slightly OT] Does TE110P (a 32-bit PCI) fit into PCIe x8 slot?   Kevin P. Fleming
• [Asterisk-Users] digium certification for Europe   Kevin P. Fleming
• [Asterisk-Users] Clustering   Kevin P. Fleming
• [Asterisk-Users] Clustering   Kevin P. Fleming
• [Asterisk-Users] H.264   Kevin P. Fleming
• [Asterisk-Users] Understanding queue timeouts + possible bug found   Kevin P. Fleming
• [Asterisk-Users] Realtime Extensions   Kevin P. Fleming
• [Asterisk-Users] priorityjumping=no   Kevin P. Fleming
• [Asterisk-Users] Failed to read gains: Invalid argument   Kevin P. Fleming
• [Asterisk-Users] Re: Asterisk-Users Digest, Vol 20, Issue 153   Kevin P. Fleming
• SV: [Asterisk-Users] re: Sound issues on SIP-SIP calls   Kevin P. Fleming
• [Asterisk-Users] Polycom IP 301 is slow   Kevin P. Fleming
• [Asterisk-Users] Asterisk and "Commercial Unix"   Kevin P. Fleming
• [Asterisk-Users] Dell 2850 w/TDM2400?   Kevin P. Fleming
• R: [Asterisk-Users] Echo cancellation   Kevin P. Fleming
• [Asterisk-Users] Softphone accepting URL   Kevin P. Fleming
• [Asterisk-Users] Problems with wcte11xp module   Kevin P. Fleming
• [Asterisk-Users] Span monitoring   Kevin P. Fleming
• [Asterisk-Users] Reload astdb?   Kevin P. Fleming
• [Asterisk-Users] Multicast Music on Hold   Kevin P. Fleming
• [Asterisk-Users] IAXY codec support and questions..   Kevin P. Fleming
• [Asterisk-Users] statechange_queue   Kevin P. Fleming
• [Asterisk-Users] transcoding g723 or g729 on asterisk   Kevin P. Fleming
• [Asterisk-Users] IAXY codec support and questions..   Kevin P. Fleming
• [Asterisk-Users] Openline4 and asterisk@home   Chuck Fletcher
• [Asterisk-Users] voicetronix and asterisk@home   Chuck Fletcher
• [Asterisk-Users] Can anybody get me setup with a hosted asterisk@home box or virtual server in the next 24 hours?   Chuck Fletcher
• [Asterisk-Users] interop problem: "Missing handling for mandatory IE 24 (cs0, Channel Identification)"   Deti Fliegl
• [Asterisk-Users] HW Echo Cancellers   Matt Florell
• [Asterisk-Users] RE: Predictive Dialer   Matt Florell
• [Asterisk-Users] Need help implementing call center features ofAsterisk   Matt Florell
• [Asterisk-Users] Need help implementing call center featuresofAsterisk   Matt Florell
• [Asterisk-Users] RE: Predictive Dialer   Matt Florell
• [Asterisk-Users] (unexplicable) peaks of machine load   Matt Florell
• [Asterisk-Users] (unexplicable) peaks of machine load   Matt Florell
• [Asterisk-Users] D4 AMI - No Caller ID   Matt Florell
• [Asterisk-Users] D4 AMI - No Caller ID   Matt Florell
• [Asterisk-Users] IAX Phone?   Matt Florell
• [Asterisk-Users] New astGUIclient VICIDIAL Released: 1.1.10   Matt Florell
• [Asterisk-Users] Local Channel   Matt Florell
• [Asterisk-Users] A general deployment question (OT)   Matt Florell
• [Asterisk-Users] Caller ID length   Matt Florell
• [Asterisk-Users] Call Monitoring / Call Takeover with Asterisk   Matt Florell
• [Asterisk-Users] Sangoma A200 error   El Flynn
• [Asterisk-Users] Sangoma A200 error   El Flynn
• [Asterisk-Users] saydigits   El Flynn
• [Asterisk-Users] New ncurses Asterisk Manager Interface   El Flynn
• [Asterisk-Users] New ncurses Asterisk Manager Interface   El Flynn
• [Asterisk-Users] New ncurses Asterisk Manager Interface   El Flynn
• [Asterisk-Users] queue_log   El Flynn
• [Asterisk-Users] Native MOH - Convert mp3 to ulaw   El Flynn
• [Asterisk-Users] Cannot leave voicemail, Asterisk/Zaptel/libpi v1.0.9   El Flynn
• [Asterisk-Users] Programming the Manager API   El Flynn
• [Asterisk-Users] Programming the Manager API   El Flynn
• [Asterisk-Users] Stability and motherboard questions with TE406P and TE410P   El Flynn
• [Asterisk-Users] beronet & bristuff   Alessio Focardi
• [Asterisk-Users] IMACS800   Patrick Forbes
• [Asterisk-Users] SMS in Spain (it seems Protocol 2)   Fran
• [Asterisk-Users] SMS in Spain (it seems Protocol 2)   Fran
• [Asterisk-Users] SMS in Spain (it seems Protocol 2)   Fran
• [Asterisk-Users] Call Path Optimization?   Matthew Fredrickson
• [Asterisk-Users] Yet again: chan_zap.c: Unable to specify channel 4: No such device   Phil Freed
• [Asterisk-Users] Yet again: chan_zap.c: Unable to specify channel 4: No such device   Phil Freed
• [Asterisk-Users] Cannot load wcfxo -- Please help!   Phil Freed
• [Asterisk-Users] Diff between X100M and X100P?   Phil Freed
• [Asterisk-Users] Cannot load wcfxo -- Please help!   Phil Freed
• [Asterisk-Users] Disappearing voicemail   Phil Freed
• [Asterisk-Users] RE: Disappearing voicemail   Phil Freed
• [Asterisk-Users] Outbound paging dialplan example?   Patrick Friedel
• [Asterisk-Users] Flash call transfer problem   Andrea Frigo
• [Asterisk-Users] Called number not recognised   Jason Frisch
• [Asterisk-Users] REGISTER headers changed   Jason Frisch
• [Asterisk-Users] REGISTER headers changed   Jason Frisch
• [Asterisk-Users] REGISTER headers changed   Jason Frisch
• [Asterisk-Users] Multiple commands per priority   Jason Frisch
• [Asterisk-Users] DNS lookup not working   Jason Frisch
• [Asterisk-Users] Problems with installing a TE110P on a Dell Poweredge 850 running Fedora Core 4   John Fulton
• [Asterisk-Users] RE: Problems with installing a TE110P on a Dell Poweredge 850   John Fulton
• [Asterisk-Users] Problem with processing channels   John Fulton
• [Asterisk-Users] Mitel 3300 PRI problems   John Fulton
• [Asterisk-Users] Re: problems with emailing voicemail   Andrew Furey
• [Asterisk-Users] IAX choppy sound   Stojan Sljivic - GDS
• [Asterisk-Users] IAX choppy sound   Stojan Sljivic - GDS
• [Asterisk-Users] IAX choppy sound   Stojan Sljivic - GDS
• [Asterisk-Users] IAX choppy sound   Stojan Sljivic - GDS
• [Asterisk-Users] IAX choppy sound   Stojan Sljivic - GDS
• [Asterisk-Users] Asterisk in the E1/R2 digital compelled enviroment   Ganbaa
• [Asterisk-Users] Re: Asterisk-Users Digest, Vol 20, Issue 211   Abhishek Gangal
• [Asterisk-Users] [SPAM] [asterisk-dev] CALL FOR COMMENTS - Dialplan   James Gardiner
• [Asterisk-Users] Echo Cancelation on TE110P   Kerry Garrison
• [Asterisk-Users] Analog Desktop Phone   Kerry Garrison
• [Asterisk-Users] Latest Dell SC430 Compatibility With Wildcard   Kerry Garrison
• [Asterisk-Users] Asterisk Users Group Meeting March 16, Irvine, Ca   Kerry Garrison
• [Asterisk-Users] Attended call transfer with GXP-2000   Kerry Garrison
• [Asterisk-Users] Asterisk Users Group Tonight, Irvine, Ca   Kerry Garrison
• [Asterisk-Users] simple question on asterisk   Kerry Garrison
• [Asterisk-Users] 3Com Phones   Kerry Garrison
• [Asterisk-Users] Re: Best GUI for basic HostedPBX service   Kerry Garrison
• [Asterisk-Users] FreePBX & AAH   Kerry Garrison
• [Asterisk-Users] Dell 2850 w/TDM2400?   Kerry Garrison
• [Asterisk-Users] AAH lost my IVR phrases   Kerry Garrison
• [Asterisk-Users] Dumb question - reaching the PSTN   Kerry Garrison
• [Asterisk-Users] Marketing Materials   Kerry Garrison
• [Asterisk-Users] How is Teliax ?   Kerry Garrison
• [Asterisk-Users] Re: Asterisk at large   Douglas Garstang
• [Asterisk-Users] Re: Asterisk at large   Douglas Garstang
• [Asterisk-Users] Re: Asterisk at large   Douglas Garstang
• [Asterisk-Users] Re: Asterisk at large   Douglas Garstang
• [Asterisk-Users] Call Transfer - "Both legs must reside on Asterisk box to transfer at this time"   Douglas Garstang
• [Asterisk-Users] Meetme Participant Announcement   Douglas Garstang
• [Asterisk-Users] Meetme Timing Interface   Douglas Garstang
• [Asterisk-Users] Does an entry in AstDB stay after reboot?   Douglas Garstang
• [Asterisk-Users] Background() App From AGI   Douglas Garstang
• [Asterisk-Users] Polycom 501 power over ethernet   Douglas Garstang
• [Asterisk-Users] Polycom 501 power over ethernet   Douglas Garstang
• [Asterisk-Users] Background() App From AGI   Douglas Garstang
• [Asterisk-Users] Call Transfer - "Both legs must reside on Asterisk box to transfer at this time"   Douglas Garstang
• [Asterisk-Users] Meetme Participant Announcement   Douglas Garstang
• [Asterisk-Users] Bad Meetme() Bug   Douglas Garstang
• [Asterisk-Users] Buddy watch?   Douglas Garstang
• [Asterisk-Users] Meetme Participant Announcement   Douglas Garstang
• [Asterisk-Users] Oh this is bad.... bindaddr and rtp traffic   Douglas Garstang
• [Asterisk-Users] Oh this is bad.... bindaddr and rtp traffic   Douglas Garstang
• [Asterisk-Users] res_mysql.conf & DNS SRV lookup   Douglas Garstang
• [Asterisk-Users] res_mysql.conf & DNS SRV lookup   Douglas Garstang
• [Asterisk-Users] res_mysql.conf & DNS SRV lookup   Douglas Garstang
• [Asterisk-Users] OT: Polycom BootRom 3.1.3 and vsftpd 2.0.3   Douglas Garstang
• [Asterisk-Users] OT: Polycom Registration Weirdness   Douglas Garstang
• [Asterisk-Users] res_mysql.conf & DNS SRV lookup   Douglas Garstang
• [Asterisk-Users] Oh this is bad.... bindaddr and rtp traffic   Douglas Garstang
• [Asterisk-Users] Oh this is bad.... bindaddr and rtp traffic   Douglas Garstang
• [Asterisk-Users] Oh this is bad.... bindaddr and rtp traffic   Douglas Garstang
• [Asterisk-Users] res_mysql.conf & DNS SRV lookup   Douglas Garstang
• [Asterisk-Users] Clustering   Douglas Garstang
• [Asterisk-Users] Clustering   Douglas Garstang
• [Asterisk-Users] Clustering   Douglas Garstang
• [Asterisk-Users] Clustering   Douglas Garstang
• [Asterisk-Users] Clustering   Douglas Garstang
• [Asterisk-Users] Clustering   Douglas Garstang
• [Asterisk-Users] Clustering / Dundi   Douglas Garstang
• [Asterisk-Users] Clustering   Douglas Garstang
• [Asterisk-Users] Clustering   Douglas Garstang
• [Asterisk-Users] Clustering   Douglas Garstang
• [Asterisk-Users] Clustering   Douglas Garstang
• [Asterisk-Users] Clustering   Douglas Garstang
• [Asterisk-Users] Clustering   Douglas Garstang
• [Asterisk-Users] Clustering   Douglas Garstang
• [Asterisk-Users] Clustering   Douglas Garstang
• [Asterisk-Users] Clustering "NEW THREAD", Almost Working   Douglas Garstang
• [Asterisk-Users] Clustering "NEW THREAD", Almost Working   Douglas Garstang
• [Asterisk-Users] Realtime SIP   Douglas Garstang
• [Asterisk-Users] Voip-Info   Douglas Garstang
• [Asterisk-Users] Voip-Info   Douglas Garstang
• [Asterisk-Users] Realtime Extensions   Douglas Garstang
• [Asterisk-Users] Realtime Extensions   Douglas Garstang
• [Asterisk-Users] Realtime Extensions   Douglas Garstang
• [Asterisk-Users] Stuck. Extenions.conf? Realtime? MySQL? Grrrrr!   Douglas Garstang
• [Asterisk-Users] Asterisk Native Sounds - in case you missed it...   Douglas Garstang
• [Asterisk-Users] Double-ring tone   Douglas Garstang
• [Asterisk-Users] how to show called name on calling polycomdisplay   Douglas Garstang
• [Asterisk-Users] how to show called name on calling polycomdisplay   Douglas Garstang
• [Asterisk-Users] Asterisk Native Sounds - in case you missed it...   Douglas Garstang
• [Asterisk-Users] Feedback from VON expo! Info on * HA andPolycomphone!!   Douglas Garstang
• [Asterisk-Users] Feedback from VON expo! Info on *HAandPolycomphone!!   Douglas Garstang
• [Asterisk-Users] Feedback from VON expo! Info on * HA and Polycomphone!!   Douglas Garstang
• [Asterisk-Users] voip-info.... again   Douglas Garstang
• [Asterisk-Users] regexten   Douglas Garstang
• [Asterisk-Users] regexten   Douglas Garstang
• [Asterisk-Users] regexten   Douglas Garstang
• [Asterisk-Users] DUNDi .... Halfway   Douglas Garstang
• [Asterisk-Users] Feedback from VON expo! Info on * HA and Polycomphone!!   Douglas Garstang
• [Asterisk-Users] regexten   Douglas Garstang
• [Asterisk-Users] Feedback from VON expo! Info on * HA and Polycomphone!!   Douglas Garstang
• [Asterisk-Users] Feedback from VON expo! Infoon*HAandPolycomphone!!   Douglas Garstang
• [Asterisk-Users] Feedback from VON expo!Infoon*HAandPolycomphone!!   Douglas Garstang
• [Asterisk-Users] Feedback from VON expo! Info on * HA andPolycomphone!!   Douglas Garstang
• [Asterisk-Users] regexten   Douglas Garstang
• [Asterisk-Users] regexten   Douglas Garstang
• [Asterisk-Users] Sticky Problem SER/Asterisk   Douglas Garstang
• [Asterisk-Users] Extra Debugging without console   Douglas Garstang
• [Asterisk-Users] Re: regexten   Douglas Garstang
• [Asterisk-Users] Re: regexten   Douglas Garstang
• [Asterisk-Users] Re: regexten   Douglas Garstang
• [Asterisk-Users] SIP Realtime Users   Douglas Garstang
• [Asterisk-Users] SIP Realtime Users   Douglas Garstang
• [Asterisk-Users] Realtime SIP users/peers   Douglas Garstang
• [Asterisk-Users] Realtime SIP users/peers - Screwed?   Douglas Garstang
• [Asterisk-Users] Feedback from VON expo! Info on * HA andPolycomphone!!   Douglas Garstang
• [Asterisk-Users] Annoying Asterisk Realtime Limitation   Douglas Garstang
• [Asterisk-Users] Voicemail Bug?   Douglas Garstang
• [Asterisk-Users] Native MOH - Convert mp3 to ulaw   Douglas Garstang
• [Asterisk-Users] Realtime / SIP Peers etc   Douglas Garstang
• [Asterisk-Users] Realtime SIP Persistency   Douglas Garstang
• [Asterisk-Users] Realtime SIP Persistency   Douglas Garstang
• [Asterisk-Users] Realtime SIP Persistency   Douglas Garstang
• [Asterisk-Users] Native MOH - Convert mp3 to ulaw   Douglas Garstang
• [Asterisk-Users] Native MOH - Convert mp3 to ulaw   Douglas Garstang
• [Asterisk-Users] Native MOH - Convert mp3 to ulaw   Douglas Garstang
• [Asterisk-Users] VoiceMailMain(@context) Problem with Option 5(Advanced)   Douglas Garstang
• [Asterisk-Users] Fw: anybody has SIP realtime working ?   Douglas Garstang
• [Asterisk-Users] 'Click to Dial'   Douglas Garstang
• [Asterisk-Users] Fw: anybody has SIP realtime working ?   Douglas Garstang
• [Asterisk-Users] RE: VoiceMailMain(@context) Problem with Option 5(Advanced)   Douglas Garstang
• [Asterisk-Users] Programming the Manager API   Douglas Garstang
• [Asterisk-Users] Programming the Manager API   Douglas Garstang
• [Asterisk-Users] Programming the Manager API   Douglas Garstang
• [Asterisk-Users] Realtime / SIP Peers etc   Douglas Garstang
• [Asterisk-Users] Fw: anybody has SIP realtime working ?   Douglas Garstang
• [Asterisk-Users] Realtime Query   Douglas Garstang
• [Asterisk-Users] polycom queue bug   Douglas Garstang
• [Asterisk-Users] Realtime Query   Douglas Garstang
• [Asterisk-Users] Ok... what is 'sip show peers' really used for?   Douglas Garstang
• [Asterisk-Users] Ok... what is 'sip show peers' really used for?   Douglas Garstang
• [Asterisk-Users] Ok... what is 'sip show peers' really used for?   Douglas Garstang
• [Asterisk-Users] Fw: anybody has SIP realtime working ?   Douglas Garstang
• [Asterisk-Users] Re: Fw: anybody has SIP realtime working ?   Douglas Garstang
• [Asterisk-Users] Ok... what is 'sip show peers' really used for?   Douglas Garstang
• [Asterisk-Users] Ok... what is 'sip show peers' really used for?   Douglas Garstang
• [Asterisk-Users] Fw: anybody has SIP realtime working ?   Douglas Garstang
• [Asterisk-Users] Re: Subscription state after reload (New subject)   Douglas Garstang
• [Asterisk-Users] Re: Subscription state after reload (New subject)   Douglas Garstang
• [Asterisk-Users] Fw: anybody has SIP realtime working ?   Douglas Garstang
• [Asterisk-Users] Tearing my hair out with Queues   Douglas Garstang
• [Asterisk-Users] Re: Fw: anybody has SIP realtime working ?   Douglas Garstang
• [Asterisk-Users] Tearing my hair out with Queues   Douglas Garstang
• [Asterisk-Users] What do the Queue timeouts really mean?   Douglas Garstang
• [Asterisk-Users] Tearing my hair out with Queues   Douglas Garstang
• [Asterisk-Users] Tearing my hair out with Queues   Douglas Garstang
• [Asterisk-Users] Fw: anybody has SIP realtime working ?   Douglas Garstang
• [Asterisk-Users] Fw: anybody has SIP realtime working ?   Douglas Garstang
• [Asterisk-Users] Tearing my hair out with Queues   Douglas Garstang
• [Asterisk-Users] Tearing my hair out with Queues   Douglas Garstang
• [Asterisk-Users] Tearing my hair out with Queues   Douglas Garstang
• [Asterisk-Users] Maximum Queue Name Length   Douglas Garstang
• [Asterisk-Users] Realtime Agents   Douglas Garstang
• [Asterisk-Users] Asterisk Failover without SER   Douglas Garstang
• [Asterisk-Users] Transferring a call with IAX   Douglas Garstang
• [Asterisk-Users] Asterisk Failover without SER   Douglas Garstang
• [Asterisk-Users] Transferring a call with IAX   Douglas Garstang
• [Asterisk-Users] Asterisk Failover without SER   Douglas Garstang
• [Asterisk-Users] Transferring a call with IAX   Douglas Garstang
• [Asterisk-Users] Transferring a call with IAX   Douglas Garstang
• [Asterisk-Users] Transferring a call with IAX   Douglas Garstang
• [Asterisk-Users] Transferring a call with IAX   Douglas Garstang
• [Asterisk-Users] Transferring a call with IAX   Douglas Garstang
• [Asterisk-Users] Transferring a call with IAX   Douglas Garstang
• [Asterisk-Users] Transferring a call with IAX   Douglas Garstang
• [Asterisk-Users] Transferring a call with IAX   Douglas Garstang
• [Asterisk-Users] Transferring a call with IAX   Douglas Garstang
• [Asterisk-Users] IAX Incoming/Outgoing   Douglas Garstang
• [Asterisk-Users] RE: IAX Incoming/Outgoing   Douglas Garstang
• [Asterisk-Users] RE: IAX Incoming/Outgoing   Douglas Garstang
• [Asterisk-Users] RE: IAX Incoming/Outgoing   Douglas Garstang
• [Asterisk-Users] RE: IAX Incoming/Outgoing   Douglas Garstang
• [Asterisk-Users] RE: IAX Incoming/Outgoing   Douglas Garstang
• [Asterisk-Users] RE: IAX Incoming/Outgoing   Douglas Garstang
• [Asterisk-Users] RE: IAX Incoming/Outgoing   Douglas Garstang
• [Asterisk-Users] RE: IAX Incoming/Outgoing   Douglas Garstang
• [Asterisk-Users] Copying SIP Subscriptions   Douglas Garstang
• [Asterisk-Users] Copying SIP Subscriptions   Douglas Garstang
• [Asterisk-Users] Polycom IP 301 is slow   Douglas Garstang
• [Asterisk-Users] Config File Management   Douglas Garstang
• [Asterisk-Users] Background() App From AGI   Douglas Garstang
• [Asterisk-Users] Config File Management   Douglas Garstang
• [Asterisk-Users] Transfer Calls - REFER   Douglas Garstang
• [Asterisk-Users] BUG 0003710 - RE: Transfer Calls - REFER   Douglas Garstang
• [Asterisk-Users] NATted phones transferring calls - BUG0003710   Douglas Garstang
• [Asterisk-Users] Agents on DND still receiving calls...   Douglas Garstang
• [Asterisk-Users] NATted phones transferring calls - BUG0003710   Douglas Garstang
• [Asterisk-Users] Transferring calls - BUG0003710   Douglas Garstang
• [Asterisk-Users] Transferring calls - BUG0003710   Douglas Garstang
• [Asterisk-Users] Receptionist Phones   Douglas Garstang
• [Asterisk-Users] Realtime Users/Peers/Friends - Ick   Douglas Garstang
• [Asterisk-Users] Realtime Users/Peers/Friends - Ick   Douglas Garstang
• [Asterisk-Users] 'sip show users' shows NAT RFC3581   Douglas Garstang
• [Asterisk-Users] Reload astdb?   Douglas Garstang
• [Asterisk-Users] Reload astdb?   Douglas Garstang
• [Asterisk-Users] Reload astdb?   Douglas Garstang
• [Asterisk-Users] Reload astdb?   Douglas Garstang
• [Asterisk-Users] Question about meetme app   Michael Gaudette
• [Asterisk-Users] Question about meetme app   Michael Gaudette
• [Asterisk-Users] Question about meetme app   Michael Gaudette
• [Asterisk-Users] FAX over PRI   Michael Gaudette
• [Asterisk-Users] FAX over PRI   Michael Gaudette
• [Asterisk-Users] FAX over PRI   Michael Gaudette
• [Asterisk-Users] Best GUI for basic HostedPBX service   Michael Gaudette
• [Asterisk-Users] Problem with System() command.   Nello Gaudino
• [Asterisk-Users] (no subject)   Savvas Gavriel
• [Asterisk-Users] asterisk management interface   Stefan-Michael. Guenther (in-put GbR)
• [Asterisk-Users] asterisk management interface   Stefan-Michael. Guenther (in-put GbR)
• [Asterisk-Users] asterisk management interface   Stefan-Michael. Guenther (in-put GbR)
• [Asterisk-Users] Benchmarking an Asterisk Server with 14k users   Stefan-Michael. Guenther (in-put GbR)
• [Asterisk-Users] asterisk turn key solution   Doug Geary
• [Asterisk-Users] call manager integration   Jerry Geis
• [Asterisk-Users] call manager integration   Jerry Geis
• [Asterisk-Users] call manager integration   Jerry Geis
• [Asterisk-Users] AMD64 x2 and asterisk 1.2.4 not hearing demo-congrats   Jerry Geis
• [Asterisk-Users] saydigits   Jerry Geis
• [Asterisk-Users] saydigits   Jerry Geis
• [Asterisk-Users] cisco 7912 not taking config   Jerry Geis
• [Asterisk-Users] TFTP question   Jerry Geis
• [Asterisk-Users] wellgate 38XX FX & FXS voip gateways with outgoing call files   Jerry Geis
• [Asterisk-Users] SIP contexts being confused   Michael George
• [Asterisk-Users] SIP contexts being confused   Michael George
• [Asterisk-Users] digium.com redesign   Michael George
• [Asterisk-Users] Sound dies   Arnar Gestsson
• [Asterisk-Users] Early media and custom SIP return codes   Federico Giannici
• [Asterisk-Users] A room full of Cisco 7960s behind NAT   Bill Gibbs
• [Asterisk-Users] Preferred editor(s) dialplan coding?   Bill Gibbs
• [Asterisk-Users] Program Buttons on Cisco 79xx Phones   Bill Gibbs
• [Asterisk-Users] Program Buttons on Cisco 79xx Phones   Bill Gibbs
• [Asterisk-Users] Program Buttons on Cisco 79xx Phones   Bill Gibbs
• [Asterisk-Users] Polycom - directory dial   Bill Gibbs
• [Asterisk-Users] Set caller ID for outgoing PRI calls   Bill Gibbs
• [Asterisk-Users] Flash Operator Panel   Bill Gibbs
• [Asterisk-Users] Anybody know about Cisco VOIP routers?   Bill Gibbs
• [Asterisk-Users] ISDN and Analog DIAL UP Connection Through Asterisk and Digium TE405P   Nico Giefing
• [Asterisk-Users] ISDN and Analog DIAL UP Connection Through Asterisk and Digium TE405P   Nico Giefing
• [Asterisk-Users] A general deployment question (OT)   Rob Gillan
• [Asterisk-Users] Softphone accepting URL   Jean-Denis Girard
• [Asterisk-Users] Calling home while on the road, will it work?   Kiffin Gish
• [Asterisk-Users] Flash Operator Panel   Giuseppe
• [Asterisk-Users] queues - monitor-join   Giuseppe
• [Asterisk-Users] Call transfer - (Call failed)   Giuseppe
• [Asterisk-Users] Call transfer - (Call failed)   Giuseppe
• [Asterisk-Users] Echo still present with Eicon Diva Server 4 Bri   Giuseppe
• [Asterisk-Users] Echo cancellation problem   Giuseppe
• [Asterisk-Users] Which g729 codec to download for a P4?   Steve Gladden
• [Asterisk-Users] Which g729 codec to download for a P4?   Steve Gladden
• [Asterisk-Users] problems with international dialing   Will Glass-Husain
• [Asterisk-Users] problems with DTMF   Will Glass-Husain
• [Asterisk-Users] Asterisk with Vonage   Gleim, Jason
• [Asterisk-Users] pseudo Direct Outward Dial   Brad Glonka
• [Asterisk-Users] Plain Old Answering Machine   Brad Glonka
• [Asterisk-Users] Plain Old Answering Machine   Brad Glonka
• [Asterisk-Users] Call Waiting Issues   Brad Glonka
• [Asterisk-Users] Realtime Content on LCD Display   Max Glucksmann
• [Asterisk-Users] Cisco 7970 SIP Image - hint lines   René Enskat [Teamware GmbH]
• [Asterisk-Users] Config File Management   David Gomillion
• [Asterisk-Users] TDM11B desperate Help wanted   Sergio Gonzalez
• [Asterisk-Users] NEWS: SIP Firmware Available for Cisco 7970   Julien Goodwin
• [Asterisk-Users] Unable to authenticate password - VM   Ben Gore
• [Asterisk-Users] Re: Unable to authenticate password - VM   Ben Gore
• [Asterisk-Users] [Slightly OT] Does TE110P (a 32-bit PCI) fit into PCIe x8 slot?   Josip Gracin
• [Asterisk-Users] [Slightly OT] Does TE110P (a 32-bit PCI) fit into PCIe x8 slot?   Josip Gracin
• [Asterisk-Users] [Slightly OT] Does TE110P (a 32-bit PCI) fit into PCIe x8 slot?   Josip Gracin
• [Asterisk-Users] VoIP QoS monitoring and failover re-routing   Tristram Graham
• [Asterisk-Users] Capturing DTMF during a call   Giordano Grandis
• R: [Asterisk-Users] Capturing DTMF during a call   Giordano Grandis
• [Asterisk-Users] QuesCom 400 IP/GSM   Giordano Grandis
• R: [Asterisk-Users] QuesCom 400 IP/GSM   Giordano Grandis
• [Asterisk-Users] Echo cancellation   Giordano Grandis
• R: [Asterisk-Users] Echo cancellation   Giordano Grandis
• [Asterisk-Users] Psgw   Giordano Grandis
• R: R: [Asterisk-Users] Echo cancellation   Giordano Grandis
• R: RE : [Asterisk-Users] Echo cancellation   Giordano Grandis
• [Asterisk-Users] Can One FXO Support Multiple Phone Lines?   Michael Graves
• [Asterisk-Users] A room full of Cisco 7960s behind NAT   Ed Greenberg
• [Asterisk-Users] A room full of Cisco 7960s behind NAT   Ed Greenberg
• [Asterisk-Users] SIP video voicemail problem   John Gruenenfelder
• [Asterisk-Users] SIP video voicemail problem   John Gruenenfelder
• [Asterisk-Users] ignore a DID?   Jesse Guardiani
• [Asterisk-Users] cidname via IAX2?   Jesse Guardiani
• [Asterisk-Users] Problem ChanSpy   David Guarnido
• [Asterisk-Users] Problem ChanSpy   David Guarnido
• [Asterisk-Users] Zaptel not compiling on lastest Centos 4.2 kernel.   Gabriel Gunderson
• [Asterisk-Users] Part-Time work available   Sahil Gupta
• [Asterisk-Users] MultiBRI in Australia - found one - maybe   Craig Guy
• [Asterisk-Users] MultiBRI in Australia - found one - maybe   Craig Guy
• [Asterisk-Users] MultiBRI in Australia - found one - maybe   Craig Guy
• [Asterisk-Users] MultiBRI in Australia - found one - maybe   Craig Guy
• [Asterisk-Users] MultiBRI in Australia - found one - maybe   Craig Guy
• [Asterisk-Users] asterisk as a fax server   Craig Guy
• [Asterisk-Users] Asterisk in production as a fax server, anyone?   Craig Guy
• [Asterisk-Users] Re: Asterisk in production as a fax server, anyone?   Craig Guy
• [Asterisk-Users] Cisco Call Manager SIP trunk + Asterisk   Chris HARIGA
• [Asterisk-Users] How to transmit Video   RAHEEL HASSAN
• [Asterisk-Users] Asterish Guru needed in Phoenix ASAP   Kyle Hagan
• [Asterisk-Users] digium certification for Europe   David Hajek
• [Asterisk-Users] Asterisk perms in manager.conf   David Hajek
• [Asterisk-Users] Re: Asterisk perms in manager.conf   David Hajek
• [Asterisk-Users] bristuff does not work with TDM400P   David Hajek
• [Asterisk-Users] Re: BRI cards, HFC, and bristuff - a general questionto clear up my understanding.   David Hajek
• [Asterisk-Users] Re: BRI cards, HFC, and bristuff - a generalquestionto clear up my understanding.   David Hajek
• [Asterisk-Users] canreinvite=yes   Paul Hales
• [Asterisk-Users] Polycom 501   Paul Hales
• [Asterisk-Users] Re: Asterisk Question   Paul Hales
• [Asterisk-Users] Polycom 501 power over ethernet   Paul Hales
• [Asterisk-Users] Polycom 501 power over ethernet   Paul Hales
• [Asterisk-Users] Variable   Paul Hales
• [Asterisk-Users] Failed installing zaptel   Hall, Eric M.
• Spam? Re: [Asterisk-Users] Failed installing zaptel   Hall, Eric M.
• [Asterisk-Users] Unknown signalling method 'pri_cpe'   Hall, Eric M.
• Spam? Re: [Asterisk-Users] Unknown signalling method 'pri_cpe'   Hall, Eric M.
• [Asterisk-Users] Web based voicemail client   Hall, Eric M.
• [Asterisk-Users] Web based voicemail client   Hall, Eric M.
• [Asterisk-Users] RE: Asterisk Users   Justin Hamade
• [Asterisk-Users] Aastra 9331i phones   Justin Hamade
• [Asterisk-Users] Best GUI for basic HostedPBX service   Justin Hamade
• [Asterisk-Users] trunking questions   Mike Hammett
• [Asterisk-Users] Trunking Questions   Mike Hammett
• [Asterisk-Users] Dial plan question - exclamtion mark   Mike Hammett
• [Asterisk-Users] AAH lost my IVR phrases   Jim Hanlon
• [Asterisk-Users] Re: AAH lost my IVR phrases   Jim Hanlon
• [Asterisk-Users] interop problem: "Missing handling for mandatory IE 24 (cs0, Channel Identification)"   Johann Hanne
• [Asterisk-Users] Asterisk, QSIG and Tenovis PBX?   Johann Hanne
• [Asterisk-Users] Managed Switches QoS to deal with network bottleneck   Dana Harding
• [Asterisk-Users] pickup a call in queue   Kristof Hardy
• [Asterisk-Users] pickup a call in queue   Kristof Hardy
• [Asterisk-Users] MultiBRI in Australia - found one - maybe   James Harper
• [Asterisk-Users] MultiBRI in Australia - found one - maybe   James Harper
• [Asterisk-Users] MultiBRI in Australia - found one - maybe   James Harper
• [Asterisk-Users] MultiBRI in Australia - found one - maybe   James Harper
• [Asterisk-Users] MultiBRI in Australia - found one - maybe   James Harper
• [Asterisk-Users] MultiBRI in Australia - found one - maybe   James Harper
• [Asterisk-Users] MultiBRI in Australia - found one - maybe   James Harper
• [Asterisk-Users] Australian approved 4BRI PCI adapter preliminary testing results   James Harper
• [Asterisk-Users] MultiBRI in Australia - found one - maybe   James Harper
• [Asterisk-Users] Australian approved 4BRI PCI adapter preliminarytesting results   James Harper
• [Asterisk-Users] Can One FXO Support Multiple Phone Lines?   James Harper
• [Asterisk-Users] MultiBRI in Australia - found one - maybe   James Harper
• [Asterisk-Users] isdn out of band signalling   James Harper
• [Asterisk-Users] Re: Cisco phones and Linksys SRW224P   James Harper
• [Asterisk-Users] Unable to forward frame   James Harper
• [Asterisk-Users] Countries supporting SMS on PSTN (ISDN)   James Harper
• [Asterisk-Users] gsm picocells   James Harper
• [Asterisk-Users] Countries supporting SMS on PSTN (ISDN)   James Harper
• [Asterisk-Users] gsm picocells   James Harper
• [Asterisk-Users] gsm picocells   James Harper
• [Asterisk-Users] gsm picocells   James Harper
• [Asterisk-Users] gsm picocells   James Harper
• [Asterisk-Users] ISDN NT Mode & CAPI   James Harper
• [Asterisk-Users] Asterisk Users Mailing List Traffic   James Harper
• [Asterisk-Users] HFC USB (was MultiBRI in Australia - found one - maybe)   James Harper
• [Asterisk-Users] HFC USB (was MultiBRI in Australia - found one-maybe)   James Harper
• [Asterisk-Users] Asterisk Users Mailing List Traffic   James Harper
• [Asterisk-Users] best MTU?   James Harper
• [Asterisk-Users] Re: gsm picocells   James Harper
• [Asterisk-Users] GSM/DECT handsets (was gsm picocells)   James Harper
• [Asterisk-Users] GSM/DECT handsets (was gsm picocells)   James Harper
• [Asterisk-Users] Re: BRI cards, HFC, and bristuff - a general questionto clear up my understanding.   James Harper
• [Asterisk-Users] Re: Subscription state after reload (New subject)   William Harrison
• FW: [Asterisk-Users] Re: Fw: anybody has SIP realtime working ?   William Harrison
• [Asterisk-Users] Asterisk Fax Question   Darrick Hartman
• [Asterisk-Users] Asterisk Fax Question   Darrick Hartman
• [Asterisk-Users] Is everyone getting mails except me?   Darrick Hartman
• [Asterisk-Users] embedded hardware for Asterisk?   Darrick Hartman
• [Asterisk-Users] Polycom IP 301 is slow   Darrick Hartman
• [Asterisk-Users] Marketing Materials   Darrick Hartman
• [Asterisk-Users] How is Teliax ?   Darrick Hartman
• [Asterisk-Users] SER inside of Asterisk is SCALABLE ?   Hamid Hashemi
• [Asterisk-Users] txfax problem   James Hawks
• [Asterisk-Users] txfax problem   James Hawks
• [Asterisk-Users] Development news :: T38 passthrough support   Paul Hayes
• [Asterisk-Users] dipura 2002 auto dial or intercom   Paul Hayes
• [Asterisk-Users] Creating a voip network... use asterisk?   Mark Hayward
• [Asterisk-Users] simple question on asterisk   Mark Hayward
• [Asterisk-Users] 3Com Phones   Daniel Hazelbaker
• [Asterisk-Users] 3Com Phones   Daniel Hazelbaker
• [Asterisk-Users] Receptionist Phones (was 3Com Phones)   Daniel Hazelbaker
• [Asterisk-Users] Receptionist Phones (was 3Com Phones)   Daniel Hazelbaker
• [Asterisk-Users] Receptionist Phones   Daniel Hazelbaker
• [Asterisk-Users] Receptionist Phones   Daniel Hazelbaker
• [Asterisk-Users] Receptionist Phones   Daniel Hazelbaker
• [Asterisk-Users] Receptionist Phones   Daniel Hazelbaker
• [Asterisk-Users] Receptionist Phones   Daniel Hazelbaker
• [Asterisk-Users] Asterisk eating CPU   Script Head
• [Asterisk-Users] Snom 360 problems   Guido Hecken
• [Asterisk-Users] OT - Cisco IP Phone and PC in diferent VLANs(with 802.1x)   Joash Herbrink
• [Asterisk-Users] Receiving Multiple calls on asterisk at home   Jeff Herring
• [Asterisk-Users] Receiving Multiple calls on asterisk at home   Jeff Herring
• [Asterisk-Users] Polycom 501 and single call only using AAH 2.2   Jeff Herring
• [Asterisk-Users] Re: asterisk-u] best CENTOS to use for latest asterisk   R P Herrold
• [Asterisk-Users] Lowering Server Load   Henri Herscher
• [Asterisk-Users] Child PID's   Paul Hewlett
• [Asterisk-Users] IPv6   Chris Hills
• [Asterisk-Users] MultiBRI in Australia - found one - maybe   David Hindmarsh
• [Asterisk-Users] QuesCom 400 IP/GSM   Jean-Michel Hiver
• R: [Asterisk-Users] QuesCom 400 IP/GSM   Jean-Michel Hiver
• [Asterisk-Users] Fake Ring Tone/Compile Addon   Kenige Ho
• [Asterisk-Users] ooh323 Gatekeeper Bug   Kenige Ho
• [Asterisk-Users] Fake Ring Tone/Compile Addon   Kenige Ho
• [Asterisk-Users] Working Asterisk with Austrian ISDN p2p   Marcus Hofbauer
• [Asterisk-Users] Working Asterisk with Austrian ISDN p2p   Marcus Hofbauer
• [Asterisk-Users] Working Asterisk with Austrian ISDN p2p   Marcus Hofbauer
• [Asterisk-Users] Working Asterisk with Austrian ISDN p2p   Marcus Hofbauer
• [Asterisk-Users] Working Asterisk with Austrian ISDN p2p   Marcus Hofbauer
• [Asterisk-Users] Working Asterisk with Austrian ISDN p2p   Marcus Hofbauer
• [Asterisk-Users] Which 2 Port ISDN Card for P2P (Austria)   Marcus Hofbauer
• [Asterisk-Users] asterisk management interface   Nick Hoffman
• [Asterisk-Users] asterisk management interface   Nick Hoffman
• [Asterisk-Users] Redirecting to another service/server   Nick Hoffman
• [Asterisk-Users] Is everyone getting mails except me?   Nick Hoffman
• [Asterisk-Users] PAP2 won't make two g729 calls at the same time   Nick Hoffman
• [Asterisk-Users] MWI, SER and asterisk   Nick Hoffman
• [Asterisk-Users] Clustering   Nick Hoffman
• [Asterisk-Users] Asterisk Native Sounds - in case you missed it...   Nick Hoffman
• [Asterisk-Users] Creating Asterisk Bounties   Nick Hoffman
• [Asterisk-Users] Polycom IP 301 is slow   Nick Hoffman
• [Asterisk-Users] Polycom IP 301 is slow   Nick Hoffman
• [Asterisk-Users] Dell 2850 w/TDM2400?   Nick Hoffman
• [Asterisk-Users] MeetMe freezes machine with Junghanns QuadBRI cards   Henning Holtschneider
• [Asterisk-Users] MeetMe freezes machine with Junghanns QuadBRI cards   Henning Holtschneider
• [Asterisk-Users] RE: MeetMe freezes machine with Junghanns   Henning Holtschneider
• [Asterisk-Users] * Meetme Freeze patch found   Henning Holtschneider
• [Asterisk-Users] OT: Snom 320, displaying text on the scree n from *   Harald Holzer
• [Asterisk-Users] IAX Phone?   Joe Hood
• [Asterisk-Users] Dialplan woes   Dave Hope
• [Asterisk-Users] Dialplan woes   Dave Hope
• [Asterisk-Users] Outgoing calls via Sipgate   Dave Hope
• [Asterisk-Users] Outgoing calls via Sipgate   Dave Hope
• [Asterisk-Users] Line connections   Jeff Hoppe
• [Asterisk-Users] Voice volume using Monitor application   Jeff Hoppe
• [Asterisk-Users] Meetme monitoring only bug   Jeff Hoppe
• [Asterisk-Users] Voice volume using Monitor application   Jeff Hoppe
• [Asterisk-Users] Feedback from VON expo! Info on * HA and Polycomphone!!   Jim Houser
• [Asterisk-Users] problems with emailing voicemail   Jim Houser
• [Asterisk-Users] FreePBX & AAH   Jim Houser
• [Asterisk-Users] FreePBX & AAH   Jim Houser
• [Asterisk-Users] Asterisk Tools for OSX   Jim Houser
• [Asterisk-Users] Marketing Materials   Jim Houser
• [Asterisk-Users] FreePBX & AAH   Jim Houser
• [Asterisk-Users] Building Asterisk embedded device   Jim Houser
• [Asterisk-Users] Asterisk, QSIG and Tenovis PBX?   Jim Houser
• [Asterisk-Users] Asterisk Fax Question   Lee Howard
• [Asterisk-Users] Asterisk in production as a fax server, anyone?   Lee Howard
• [Asterisk-Users] Re: Asterisk in production as a fax server, anyone?   Lee Howard
• [Asterisk-Users] Re: Asterisk in production as a fax server, anyone?   Lee Howard
• [Asterisk-Users] Asterisk and Hylafax, on the same box   Lee Howard
• [Asterisk-Users] Asterisk in production as a fax server, anyone?   Lee Howard
• [Asterisk-Users] integration with Toshiba PBX system   Charles Huang
• [Asterisk-Users] Re: Stuck. Extenions.conf? Realtime? MySQL?   Nic Hughes
• [Asterisk-Users] TIMESTAMP, DATETIME not working   Mark Hulber
• [Asterisk-Users] really need help with outgoing calls..PSTN errors   Mark Hulber
• [Asterisk-Users] There is lacking behind in recorded calls via sox   Mazhar Hussain
• [Asterisk-Users] there is lack behind in recoded calls via sox   Mazhar Hussain
• [Asterisk-Users] Voice mail not working with Asteriks 1.2.5   Mazhar Hussain
• [Asterisk-Users] How to assign a specific PSTN line to a specific extension ???   Faisal Inam
• [Asterisk-Users] How to configure PSTN lines permissions to different extensions ???   Faisal Inam
• [Asterisk-Users] How to configure PSTN lines permissions to different extensions ???   Faisal Inam
• [Asterisk-Users] PSTN lines permission settings to different extensions   Faisal Inam
• [Asterisk-Users] How to make caller groups ???   Faisal Inam
• [Asterisk-Users] How to make extension groups ???   Faisal Inam
• [Asterisk-Users] How to make groups of extensions ???   Faisal Inam
• [Asterisk-Users] Callerid error: receiving DNID instead of callerID   Giorgio Incantalupo
• [Asterisk-Users] grandstream handytone 286 sometimes dials out wrong number   Giorgio Incantalupo
• [Asterisk-Users] grandstream handytone 286 sometimes dials out wrong number   Giorgio Incantalupo
• [Asterisk-Users] how to show called name on calling polycom display   Giorgio Incantalupo
• [Asterisk-Users] send text to a device   Giorgio Incantalupo
• [Asterisk-Users] FOP flash panel: how to reload config files when running   Giorgio Incantalupo
• [Asterisk-Users] TFTP problems on FC4   Ioan Indreias
• [Asterisk-Users] DID billing   Bernard Cresencia - CrossNet International
• [Asterisk-Users] error messages on /var/log/asterisk/messages   Ira
• [Asterisk-Users] Asterisk Fax Question   Ira
• [Asterisk-Users] really need help with outgoing calls..PSTN errors   Ira
• [Asterisk-Users] really need help with outgoing calls..PSTN errors   Ira
• [Asterisk-Users] Dialplan - strip IDD prefix and insert another   Ira
• [Asterisk-Users] Extracting info from the $EXTEN variable   Ira
• [Asterisk-Users] RE: Delay in ringing   Ira
• [Asterisk-Users] Asterisk Native Sounds - in case you missed it...   Ira
• [Asterisk-Users] Asterisk Native Sounds - in case you missed it...   Ira
• [Asterisk-Users] asterisk@home V's Asterisk   Ira
• [Asterisk-Users] asterisk@home V's Asterisk   Ira
• [Asterisk-Users] asterisk@home V's Asterisk   Ira
• [Asterisk-Users] Asterisk Users Mailing List Traffic   Ira
• [Asterisk-Users] Asterisk Users Mailing List Traffic   Ira
• [Asterisk-Users] How to make extension groups ???   Ira
• [Asterisk-Users] I'm FED UP with BroadVoice   Ira
• [Asterisk-Users] GnuGk and Asterisk IVR   Angel Ivanov
• [Asterisk-Users] How to change Budgetone dialtone?   Dmitry Ivanov
• [Asterisk-Users] How to change Budgetone dialtone?   Dmitry Ivanov
• [Asterisk-Users] Set caller ID for outgoing PRI calls   Dmitry Ivanov
• [Asterisk-Users] Set caller ID for outgoing PRI calls   Dmitry Ivanov
• [Asterisk-Users] Set caller ID for outgoing PRI calls   Dmitry Ivanov
• [Asterisk-Users] IAX to SIP conversion: SIP From header issue   JS
• [Asterisk-Users] IAX to SIP conversion: SIP From header issue   JS
• [Asterisk-Users] Can we replace existing SIP call with new one?   JS
• [Asterisk-Users] channel manipulation   JS
• [Asterisk-Users] channel bridging   JS
• [Asterisk-Users] Streaming voice using IAX   JS
• [Asterisk-Users] ignore a DID?   Nabeel Jafferali
• [Asterisk-Users] wake up calls   Nabeel Jafferali
• [Asterisk-Users] snom 320 MWI light   Nabeel Jafferali
• [Asterisk-Users] Sipura RMA   Nabeel Jafferali
• [Asterisk-Users] snom 320 MWI light   Nabeel Jafferali
• [Asterisk-Users] snom 320 MWI light   Nabeel Jafferali
• [Asterisk-Users] NEWS: SIP Firmware Available for Cisco 7970   Nabeel Jafferali
• [Asterisk-Users] 7940/60 SIP 8.2   Nabeel Jafferali
• [Asterisk-Users] Cisco 7960 SIP - Displaying Time   Nabeel Jafferali
• [Asterisk-Users] Is extension.conf documentation wrong?   Nabeel Jafferali
• [Asterisk-Users] 7940/60 SIP 8.2   Nabeel Jafferali
• [Asterisk-Users] dipura 2002 auto dial or intercom   Nabeel Jafferali
• [Asterisk-Users] Cisco 7960 8.2 callerID lists proxy?   Nabeel Jafferali
• [Asterisk-Users] Aastra 480i CT - multiple lines?   Nabeel Jafferali
• [Asterisk-Users] answer delay   Nabeel Jafferali
• [Asterisk-Users] callerid= in zapata.conf   Nabeel Jafferali
• [Asterisk-Users] callerid= in zapata.conf   Nabeel Jafferali
• [Asterisk-Users] callerid= in zapata.conf   Nabeel Jafferali
• [Asterisk-Users] pseudo Direct Outward Dial   Nabeel Jafferali
• [Asterisk-Users] Booklet (mini hacking guide) for Asterisk + Asterisk-Java + NISTSIP + SIPp   Cokorda Raka Angga Jananuraga
• [Asterisk-Users] Booklet (mini hacking guide) for Asterisk + Asterisk-Java + NISTSIP + SIPp   Cokorda Raka Angga Jananuraga
• [Asterisk-Users] Periodic-announce in queues   CC Jay
• [Asterisk-Users] answer delay   CC Jay
• [Asterisk-Users] Tearing my hair out with Queues   CC Jay
• [Asterisk-Users] Tearing my hair out with Queues   CC Jay
• [Asterisk-Users] Tearing my hair out with Queues   CC Jay
• [Asterisk-Users] Tearing my hair out with Queues   CC Jay
• [Asterisk-Users] Tearing my hair out with Queues   CC Jay
• [Asterisk-Users] SIP Realtime 1.2.5 and Username/auth name mismatch ?   Frederic Jean
• [Asterisk-Users] Fw: anybody has SIP realtime working ?   Frederic Jean
• [Asterisk-Users] Fw: anybody has SIP realtime working ?   Frederic Jean
• [Asterisk-Users] CallerID chopped by half ? :-)   Frederic Jean
• [Asterisk-Users] Mandrake zaptel module not found after compiling   Frederic Jean
• [Asterisk-Users] SIP realtime: how to authenticate without "name" field ?   Frederic Jean
• [Asterisk-Users] IAX problems - please help me   Frederic Jean
• [Asterisk-Users] How we tell who is using VAD ?   Frederic Jean
• [Asterisk-Users] capiHOLD missing in BRIstuff 0.3.0   Jens.Kammann at dlr.de
• [Asterisk-Users] Asterisk transfer conflict   Fredrik Jensen
• [Asterisk-Users] Periodic-announce in queues   Fredrik Emil Jensen
• [Asterisk-Users] Periodic-announce in queues   Fredrik Emil Jensen
• [Asterisk-Users] Sipura SPA-3000 vs Linksys SPA3000   John Jensen
• [Asterisk-Users] Upgrading to 1.2.5?   John Jensen
• [Asterisk-Users] Size'ing/performance   John Jensen
• [Asterisk-Users] Asterisk hosted solution   Thorben Jensen
• [Asterisk-Users] Authenticate CDR Logging   Jeremy
• [Asterisk-Users] Authenticate CDR Logging   Jeremy
• [Asterisk-Users] redirect output   Jeremy
• [Asterisk-Users] New one on me: How to UN-transfer   Jeremy
• [Asterisk-Users] (no subject)   Jeremy
• [Asterisk-Users] DISA alternative   Jeremy
• [Asterisk-Users] asterisk billing   Jeremy
• [Asterisk-Users] VoIP prepaid billing   Jeremy
• [Asterisk-Users] CLI Echo   Jeremy
• [Asterisk-Users] Master.csv Shell Script   Jeremy
• [Asterisk-Users] Master.csv Shell Script   Jeremy
• [Asterisk-Users] CID passthrough   Jeremy
• [Asterisk-Users] AstCC   Jeremy
• [Asterisk-Users] Blacklist out bound numbers from file   Jeremy
• [Asterisk-Users] Blacklist out bound numbers from file   Jeremy
• [Asterisk-Users] AstCC   Jeremy
• [Asterisk-Users] Authenticate   Jeremy
• [Asterisk-Users] Authenticate   Jeremy
• [Asterisk-Users] AstCC   Jeremy
• [Asterisk-Users] AstCC   Jeremy
• [Asterisk-Users] G729, G729 annex A or G729 annex B?   Pavel Jezek
• [Asterisk-Users] RE: Re: Cisco phones and Linksys SRW224P   Pavel Jezek
• [Asterisk-Users] Re: RE: Re: Cisco phones and Linksys SRW224P   Pavel Jezek
• [Asterisk-Users] Re: RE: Re: Cisco phones and Linksys SRW224P   Pavel Jezek
• [Asterisk-Users] Prodding channel h323   Pavel Jezek
• [Asterisk-Users] Netgear FS116P and Cisco 79XX phones   Pavel Jezek
• [Asterisk-Users] IAX: Auto-congesting call due to slow response   Pavel Jezek
• [Asterisk-Users] Bad FXS Module?   Jimmy
• [Asterisk-Users] New ncurses Asterisk Manager Interface   Jimmy
• [Asterisk-Users] SIP/Video client for PocketPC that works with Asterisk?   Joe
• [Asterisk-Users] SIP/Video client for PocketPC that works with Asterisk?   Joe
• [Asterisk-Users] Scrolling messages   Joe
• [Asterisk-Users] TFTP problems on FC4   Joe
• [Asterisk-Users] Redirect problem/bug/feature   Joe
• [Asterisk-Users] Cisco POS 3-08-2   Ron Joffe
• [Asterisk-Users] Cisco POS 3-08-2   Ron Joffe
• [Asterisk-Users] Cisco POS 3-08-2   Ron Joffe
• [Asterisk-Users] Agents and Chanspy   Johann
• [Asterisk-Users] login/logout agents in a specific queue   Johann
• [Asterisk-Users] RFC Follow Me Find Me script   Johann
• [Asterisk-Users] Span monitoring   Johann
• [Asterisk-Users] Same CID on multiple users(friends9 in SIP.conf   Arne Morten Johansen
• [Asterisk-Users] *** Yet another boring weekend? Test new Asterisk features in development!   Olle E Johansson
• [Asterisk-Users] ENUM lookup issues with e164.org   Olle E Johansson
• [Asterisk-Users] Re: [asterisk-dev] Confusion about construction of RURIs from contact headers for BYEs generated by *   Olle E Johansson
• [Asterisk-Users] Re: [asterisk-dev] Confusion about construction of RURIs from contactheaders for BYEs generated by *   Olle E Johansson
• [Asterisk-Users] indications & SIP   Olle E Johansson
• [Asterisk-Users] res_mysql.conf & DNS SRV lookup   Olle E Johansson
• [Asterisk-Users] res_mysql.conf & DNS SRV lookup   Olle E Johansson
• [Asterisk-Users] Multiple Subscriptions to SIP accounts at Same Domain   Olle E Johansson
• [Asterisk-Users] sending text to display of sip phones   Olle E Johansson
• [Asterisk-Users] parking slot lights - testers wanted   Olle E Johansson
• [Asterisk-Users] Real Time Asterisk   Olle E Johansson
• [Asterisk-Users] Any way to change dns timeout value? Asterisk hangs if internet unreachable   Olle E Johansson
• [Asterisk-Users] Jitter buffer for SIP channels (OT?)   Olle E Johansson
• [Asterisk-Users] Development news :: T38 passthrough support   Olle E Johansson
• [Asterisk-Users] Clustering / Dundi   Olle E Johansson
• [Asterisk-Users] Development news :: T38 passthrough support   Olle E Johansson
• [Asterisk-Users] SIP Jitter Buffer for 1.2.5   Olle E Johansson
• [Asterisk-Users] incoming limit, call_limit, or call-limit?   Olle E Johansson
• [Asterisk-Users] priorityjumping=no   Olle E Johansson
• [Asterisk-Users] SIP Jitter Buffer for 1.2.5   Olle E Johansson
• [Asterisk-Users] SIP Jitter Buffer for 1.2.5   Olle E Johansson
• [Asterisk-Users] SIP Jitter Buffer for 1.2.5   Olle E Johansson
• [Asterisk-Users] Realtime / SIP Peers etc   Olle E Johansson
• [Asterisk-Users] Realtime / SIP Peers etc   Olle E Johansson
• [Asterisk-Users] what are these and can they be fixed?   Olle E Johansson
• [Asterisk-Users] Re: Fw: anybody has SIP realtime working ?   Olle E Johansson
• [Asterisk-Users] Ok... what is 'sip show peers' really used for?   Olle E Johansson
• [Asterisk-Users] Ok... what is 'sip show peers' really used for?   Olle E Johansson
• [Asterisk-Users] Re: Subscription state after reload (New subject)   Olle E Johansson
• [Asterisk-Users] Transferring calls - BUG0003710   Olle E Johansson
• [Asterisk-Users] MeetAsterisk Europe:: Get an Asterisk one-day introduction!   Olle E Johansson
• [Asterisk-Users] Sipura RMA   Hugh L. Johnson
• [Asterisk-Users] E911 from Remote Office via PRI   Hugh L. Johnson
• [Asterisk-Users] E911 from Remote Office via PRI   Hugh L. Johnson
• [Asterisk-Users] Polycom IP600 - no ring?   Peter Johnson
• [Asterisk-Users] Considering Asterisk   Thomas Johnson
• [Asterisk-Users] Same CID on multiple users(friends9 in SIP.conf   Jerry Jones
• [Asterisk-Users] Asterisk Fax Question   Jerry Jones
• RES: [Asterisk-Users] DTFM or FSK   Jerry Jones
• [Asterisk-Users] Re: how to show called name on calling polycom display   Jerry Jones
• [Asterisk-Users] type of incoming lines   Jerry Jones
• [Asterisk-Users] How to create [new_context] in extensions.conf?   Jerry Jones
• [Asterisk-Users] Receptionist Phones   Jerry Jones
• [Asterisk-Users] Receptionist Phones   Jerry Jones
• [Asterisk-Users] How to check if a phone / line is used?   Jerry Jones
• [Asterisk-Users] Display Name   Jerry Jones
• [Asterisk-Users] new beta Grandstream firmware HT488_496_386   Steve Jones
• [Asterisk-Users] new beta Grandstream firmware HT488_496_386   Steve Jones
• [Asterisk-Users] OSHA requirement to "reach a live human" ??   Steve Jones
• [Asterisk-Users] OSHA requirement to "reach a live human" ??   Steve Jones
• [Asterisk-Users] asterisk@home V's Asterisk   Steve Jones
• [Asterisk-Users] Asterisk select outbound trunk based on minutes used per month??   Steve Jones
• [Asterisk-Users] Re: problems with emailing voicemail   Steve Jones
• [Asterisk-Users] Re: problems with emailing voicemail   Steve Jones
• [Asterisk-Users] Re: problems with emailing voicemail   Steve Jones
• [Asterisk-Users] I'm FED UP with BroadVoice   Steve Jones
• [Asterisk-Users] Asterisk with Vonage   Steve Jones
• [Asterisk-Users] Two X100p clones. One not available for outbound?   Steve Jones
• [Asterisk-Users] Asterisk with Vonage   Steve Jones
• [Asterisk-Users] asterisk management interface   Senad Jordanovic
• [Asterisk-Users] asterisk management interface   Senad Jordanovic
• [Asterisk-Users] Cannot log into mailbox , guidance requested   John Joseph
• [Asterisk-Users] asterisk management interface   John Joseph
• [Asterisk-Users] Can log into the mailbox from Soft-phone , but not from Hardware Phone   John Joseph
• [Asterisk-Users] Can log into the mailbox from Soft-phone , but not from Hardware Phone   John Joseph
• [Asterisk-Users] Problem compiling ztdummy on centos 4, 2.6 kernel   John Joseph
• [Asterisk-Users] Can I avoid configuring FXS part in zaptel.conf and zapata.conf   John Joseph
• [Asterisk-Users] ZT_CHANCONFIG failed on channel 2: , Guidance requested   John Joseph
• [Asterisk-Users] ZT_CHANCONFIG failed on channel 2: , Guidance requested   John Joseph
• [Asterisk-Users] Calls from PSTN , answering, When transfered get Hungup 'Zap/1-1'   John Joseph
• [Asterisk-Users] How to setup Proxy info to * box , [* box behind a squid proxy and firewall ]   John Joseph
• [Asterisk-Users] RE: IAX Incoming/Outgoing   John Joseph
• [Asterisk-Users] A room full of Cisco 7960s behind NAT   Martin Joseph
• [Asterisk-Users] Zoom 5801 problems with * (Wellgate 3701a)   Martin Joseph
• [Asterisk-Users] Newbie config help? Wellgate 3701a   Martin Joseph
• [Asterisk-Users] Sipura SPA-3000 and PSTN dtmf   Martin Joseph
• [Asterisk-Users] Re: G729 and Meetme   Martin Joseph
• [Asterisk-Users] asterisk management interface   Martin Joseph
• [Asterisk-Users] Polling Asterisk for Life   Martin Joseph
• [Asterisk-Users] dtmf tones problem with unicall and E1   Martin Joseph
• [Asterisk-Users] Echo Cancelation on TE110P   Martin Joseph
• [Asterisk-Users] Hardware Requirements for 1M minutes   Martin Joseph
• [Asterisk-Users] IAX2 register problem   Martin Joseph
• [Asterisk-Users] Sipura RMA   Martin Joseph
• [Asterisk-Users] new beta Grandstream firmware HT488_496_386   Martin Joseph
• [Asterisk-Users] Sipura RMA   Martin Joseph
• [Asterisk-Users] Call Waiting? Should this just work?   Martin Joseph
• [Asterisk-Users] Asterisk 1.2.5 Released   Martin Joseph
• [Asterisk-Users] Upgrading to 1.2.5?   Martin Joseph
• [Asterisk-Users] Asterisk on MacOS?   Martin Joseph
• [Asterisk-Users] Two asterisks on one machine   Martin Joseph
• [Asterisk-Users] grandstream handytone 286 sometimes dials out wrong number   Martin Joseph
• [Asterisk-Users] new beta Grandstream firmware HT488_496_386   Martin Joseph
• [Asterisk-Users] What is asterisk   Martin Joseph
• [Asterisk-Users] Re: ON DEMAND call Recording   Martin Joseph
• [Asterisk-Users] Re: ON DEMAND call Recording   Martin Joseph
• [Asterisk-Users] PBX-VPN-SIP-Asterisk trouble   Martin Joseph
• [Asterisk-Users] Asterisk Re-invites - how to tell ?   Martin Joseph
• [Asterisk-Users] ADPCM - vs - G.726   Martin Joseph
• [Asterisk-Users] Analog Desktop Phone   Martin Joseph
• [Asterisk-Users] Speakeasy VOIP + Asterisk?   Martin Joseph
• [Asterisk-Users] stop monitor on transfer   Martin Joseph
• [Asterisk-Users] news-reading question   Martin Joseph
• [Asterisk-Users] OT call recording   Martin Joseph
• [Asterisk-Users] Analog Desktop Phone   Martin Joseph
• [Asterisk-Users] Multiple IAX clients behind a firewall   Martin Joseph
• [Asterisk-Users] Considering Asterisk   Martin Joseph
• [Asterisk-Users] Dumb question (hang up detection/Zapata.conf)   Martin Joseph
• [Asterisk-Users] OT call recording (was stop monitor on transfer)   Martin Joseph
• [Asterisk-Users] Echo Cancellation   Martin Joseph
• [Asterisk-Users] Asterisk Native Sounds - in case you missed it...   Martin Joseph
• [Asterisk-Users] asterisk crash too much?   Martin Joseph
• [Asterisk-Users] IAX choppy sound   Martin Joseph
• [Asterisk-Users] Codecs? - Asked to transmit frame type 256, while native formats is 8 (read/write = 8/8)   Martin Joseph
• [Asterisk-Users] RFC 2833 and SIP? DTMF? What am I not getting?   Martin Joseph
• [Asterisk-Users] Dumb question (hang up detection/Zapata.conf)   Martin Joseph
• [Asterisk-Users] IAX Phone?   Martin Joseph
• [Asterisk-Users] Exchange 12 Unified Messaging   Martin Joseph
• [Asterisk-Users] One-Way SIP Audio with SVN Codebase (CANCEL)   Martin Joseph
• [Asterisk-Users] RFC 2833 and SIP? DTMF? What am I not getting?   Martin Joseph
• [Asterisk-Users] Asterisk Users Mailing List Traffic   Martin Joseph
• [Asterisk-Users] An FXO version of IAXy?   Martin Joseph
• [Asterisk-Users] Grandstream unit HT-488   Martin Joseph
• [Asterisk-Users] Problem with chan_iax.c implimentation causes bad audio?   Martin Joseph
• [Asterisk-Users] Phones were working fine - Now there is no audiowhen calling between extensions   Martin Joseph
• [Asterisk-Users] 'Click to Dial'   Martin Joseph
• [Asterisk-Users] Native MOH - Convert mp3 to ulaw   Martin Joseph
• [Asterisk-Users] An FXO version of IAXy?   Martin Joseph
• [Asterisk-Users] Sound issues on SIP-SIP calls   Martin Joseph
• [Asterisk-Users] problems with DTMF   Martin Joseph
• [Asterisk-Users] G729 License questions   Martin Joseph
• [Asterisk-Users] SIP - Problem with audio clipping   Martin Joseph
• [Asterisk-Users] Re: problems with emailing voicemail   Martin Joseph
• [Asterisk-Users] welltech Wellgate 3804 in SIP mode   Martin Joseph
• [Asterisk-Users] I'm FED UP with BroadVoice   Martin Joseph
• [Asterisk-Users] Polycom 501's for sale   Martin Joseph
• [Asterisk-Users] Which Mac OSX softphone with IAX2 support?   Martin Joseph
• [Asterisk-Users] Extension a?   Martin Joseph
• [Asterisk-Users] SIP trunk problem   Martin Joseph
• [Asterisk-Users] SIP trunk problem   Martin Joseph
• [Asterisk-Users] RE: IAX Incoming/Outgoing   Martin Joseph
• [Asterisk-Users] RE: IAX Incoming/Outgoing   Martin Joseph
• [Asterisk-Users] Asterisk and "Commercial Unix"   Martin Joseph
• [Asterisk-Users] Asterisk and "Commercial Unix"   Martin Joseph
• [Asterisk-Users] welltech Wellgate 3804 in SIP mode   Martin Joseph
• [Asterisk-Users] Incoming calls   Josh
• [Asterisk-Users] SIP Plugin or Component   Dan Journo
• [Asterisk-Users] PSTN to Asterisk VOIP in Manila   Lawrence Jovellanos
• [Asterisk-Users] IAX2 register problem   Bartosz Jozwiak
• [Asterisk-Users] PLEASE HELP , a2billing problem with call duration   Areski K
• [Asterisk-Users] ANNOUNCEMENT : A2Billing (Asterisk2Billing) - release v1.1   Areski K
• [Asterisk-Users] Experiences with ATA model Octtel SP200SO   Norbert Kamenicky
• [Asterisk-Users] Polling Asterisk for Life   Geoff Karl
• [Asterisk-Users] Remote dialtone   Karlos
• [Asterisk-Users] Remote dialtone   Karlos
• [Asterisk-Users] Remote dialtone   Karlos
• [Asterisk-Users] OT - Cisco IP Phone and PC in diferent VLANs (with 802.1x)   Nicholas Kathmann
• [Asterisk-Users] Real Time Asterisk   Nicholas Kathmann
• [Asterisk-Users] Polycom 601 Message Center   Nicholas Kathmann
• [Asterisk-Users] Zaptel not compiling on lastest Centos 4.2 kernel.   Alejandro Kauffmann
• [Asterisk-Users] Blind Transfer / Ring Groups Problem   Martin Keding
• [Asterisk-Users] UK Caller ID - Asterisk 1.2.5 - TDM4 Card   Magnus Kelly
• [Asterisk-Users] Snom 360 problems   Brian Kennedy
• [Asterisk-Users] Snom 360 problems   Brian Kennedy
• [Asterisk-Users] Snom 360 problems   Brian Kennedy
• [Asterisk-Users] Snom 360 problems   Brian Kennedy
• [Asterisk-Users] Reverse group in zapata.conf   Sean Kennedy
• [Asterisk-Users] Reverse group in zapata.conf   Sean Kennedy
• [Asterisk-Users] OT: Snom 320, displaying text on the screen from *   Sean Kennedy
• [Asterisk-Users] OT: Snom 320, displaying text on the screen from *   Sean Kennedy
• [Asterisk-Users] Menu in queue   Sean Kennedy
• [Asterisk-Users] priorityjumping=no   Steve Kennedy
• [Asterisk-Users] priorityjumping=no   Steve Kennedy
• [Asterisk-Users] priorityjumping=no   Steve Kennedy
• [Asterisk-Users] EICON Diva 4BRI   Steve Kennedy
• [Asterisk-Users] UK Caller ID - Asterisk 1.2.5 - TDM4 Card   Steve Kennedy
• [Asterisk-Users] gsm picocells   Steve Kennedy
• [Asterisk-Users] gsm picocells   Steve Kennedy
• [Asterisk-Users] Zap<-->IAX codec?   Steve Kennedy
• [Asterisk-Users] Re: gsm picocells   Steve Kennedy
• [Asterisk-Users] UK EI   Steve Kennedy
• [Asterisk-Users] sipura spa2 + asterisk bug ?   Steve Kennedy
• [Asterisk-Users] MCC v.1.3 Released   Mindaugas Kezys
• [Asterisk-Users] ruby-agi-1.1.2 released   Mohammad Khan
• [Asterisk-Users] G729 and Meetme   Kristian Kielhofner
• [Asterisk-Users] G.726 codec - can we select bandwidth?   Kristian Kielhofner
• [Asterisk-Users] IAX to SIP conversion: SIP From header issue   Kristian Kielhofner
• [Asterisk-Users] Upgrading to 1.2.5?   Kristian Kielhofner
• [Asterisk-Users] Capturing DTMF during a call   Kristian Kielhofner
• [Asterisk-Users] OT: Polycom BootRom 3.1.3 and vsftpd 2.0.3 WARNING!!!   Kristian Kielhofner
• [Asterisk-Users] OT: Polycom BootRom 3.1.3 and vsftpd 2.0.3   Kristian Kielhofner
• [Asterisk-Users] OT: Polycom BootRom 3.1.3 and vsftpd 2.0.3   Kristian Kielhofner
• [Asterisk-Users] System Design   Kristian Kielhofner
• [Asterisk-Users] PAP2 won't make two g729 calls at the same time   Kristian Kielhofner
• [Asterisk-Users] Development news :: T38 passthrough support   Kristian Kielhofner
• [Asterisk-Users] Asterisk Native Sounds - in case you missed it...   Kristian Kielhofner
• [Asterisk-Users] Asterisk Native Sounds - in case you missed it...   Kristian Kielhofner
• [Asterisk-Users] re: asterisk and iptables   Kristian Kielhofner
• [Asterisk-Users] voip-info.... again   Kristian Kielhofner
• [Asterisk-Users] Best budget IP phone at the moment?   Kristian Kielhofner
• [Asterisk-Users] Asterisk snapshots?   Kristian Kielhofner
• [Asterisk-Users] Asterisk snapshots?   Kristian Kielhofner
• [Asterisk-Users] Asterisk and "Commercial Unix"   Kristian Kielhofner
• [Asterisk-Users] Asterisk and "Commercial Unix"   Kristian Kielhofner
• [Asterisk-Users] CLI Echo   Kristian Kielhofner
• [Asterisk-Users] Sipura SPA-3000 vs Linksys SPA3000   Frank A. Kingswood
• [Asterisk-Users] Comments on Gafachi   John Kington
• [Asterisk-Users] chan_zap ast_pickup_call issue redux   Andrew Kirch
• [Asterisk-Users] RFC Follow Me Find Me script   Andrew Kirch
• [Asterisk-Users] Dial from php   Andrew Kirch
• [Asterisk-Users] asterisk turn key solution   Andrew Kirch
• [Asterisk-Users] Unable to start Asterisk 1.2.5 with Asterisk-Addons 1.2.1   Andrew D Kirch
• [Asterisk-Users] RFC Follow Me Find Me script   Andrew D Kirch
• [Asterisk-Users] RFC Follow Me Find Me script   Andrew D Kirch
• [Asterisk-Users] Hardware timing source for MeetMe   Andrew D Kirch
• [Asterisk-Users] VERY IMPORTANT(TREAT WITH URGENCY)   Andrew D Kirch
• [Asterisk-Users] Polycom 501's for sale   Kevin Kirts
• [Asterisk-Users] Re: gsm picocells   Kevin Kirts
• [Asterisk-Users] Newbie needs audio help   Joe Kletch
• [Asterisk-Users] Newbie needs audio help   Joe Kletch
• [Asterisk-Users] Zaptel compile errors on x86_64   Walter Klomp
• [Asterisk-Users] Zaptel compile errors on x86_64   Walter Klomp
• [Asterisk-Users] Zaptel compile errors on x86_64 - DEFINE_SPINLOCK???   Walter Klomp
• [Asterisk-Users] SIP contexts being confused   Rene Kluwen
• [Asterisk-Users] Weird DTMF issue   Andrew Kohlsmith
• [Asterisk-Users] fax receive using TDM400P   Andrew Kohlsmith
• [Asterisk-Users] Re: transfers/parked calls + polycom 501   Andrew Kohlsmith
• [Asterisk-Users] slinear bandwidth   Andrew Kohlsmith
• [Asterisk-Users] OSHA requirement to "reach a live human" ??   Andrew Kohlsmith
• [Asterisk-Users] Sync Source: Internally clocked   Andrew Kohlsmith
• [Asterisk-Users] Echo canceller data-points   Andrew Kohlsmith
• [Asterisk-Users] Echo canceller data-points   Andrew Kohlsmith
• [Asterisk-Users] OT: Unblocking bloced CID   Andrew Kohlsmith
• [Asterisk-Users] gsm picocells   Andrew Kohlsmith
• [Asterisk-Users] gsm picocells   Andrew Kohlsmith
• [Asterisk-Users] Asterisk Users Mailing List Traffic   Andrew Kohlsmith
• [Asterisk-Users] Asterisk Users Mailing List Traffic   Andrew Kohlsmith
• [Asterisk-Users] answer delay   Andrew Kohlsmith
• [Asterisk-Users] answer delay   Andrew Kohlsmith
• [Asterisk-Users] Problem with chan_iax.c implimentation causes bad audio?   Andrew Kohlsmith
• [Asterisk-Users] Asterisk Users Mailing List Traffic   Andrew Kohlsmith
• [Asterisk-Users] Problem with chan_iax.c implimentation causes bad audio?   Andrew Kohlsmith
• [Asterisk-Users] Problem with chan_iax.c implimentation causesbad audio?   Andrew Kohlsmith
• [Asterisk-Users] Problem with chan_iax.c implimentation causesbadaudio?   Andrew Kohlsmith
• [Asterisk-Users] How to make groups of extensions ???   Andrew Kohlsmith
• [Asterisk-Users] Problem with chan_iax.c implimentationcausesbadaudio?   Andrew Kohlsmith
• [Asterisk-Users] How to make groups of extensions ???   Andrew Kohlsmith
• [Asterisk-Users] Phones were working fine - Now there is noaudiowhen calling between extensions   Andrew Kohlsmith
• [Asterisk-Users] How to make groups of extensions ???   Andrew Kohlsmith
• [Asterisk-Users] How to make groups of extensions ???   Andrew Kohlsmith
• [Asterisk-Users] How to make groups of extensions ???   Andrew Kohlsmith
• [Asterisk-Users] FAX over PRI   Andrew Kohlsmith
• [Asterisk-Users] Asterisk Users Mailing List Traffic   Andrew Kohlsmith
• [Asterisk-Users] FAX over PRI   Andrew Kohlsmith
• [Asterisk-Users] Fw: anybody has SIP realtime working ?   Andrew Kohlsmith
• [Asterisk-Users] Fw: anybody has SIP realtime working ?   Andrew Kohlsmith
• [Asterisk-Users] Snom 360 Hinting tricks   Andrew Kohlsmith
• [Asterisk-Users] Realtime / SIP Peers etc   Andrew Kohlsmith
• [Asterisk-Users] How to make groups of extensions ???   Andrew Kohlsmith
• [Asterisk-Users] Fw: anybody has SIP realtime working ?   Andrew Kohlsmith
• [Asterisk-Users] Fw: anybody has SIP realtime working ?   Andrew Kohlsmith
• [Asterisk-Users] Asterisk Users Mailing List Traffic   Andrew Kohlsmith
• [Asterisk-Users] Fw: anybody has SIP realtime working ?   Andrew Kohlsmith
• [Asterisk-Users] If a packet is dropped, is it possible to replay the previous packet in Asterisk?   Andrew Kohlsmith
• [Asterisk-Users] Asterisk Users Mailing List Traffic   Andrew Kohlsmith
• [Asterisk-Users] Fw: anybody has SIP realtime working ?   Andrew Kohlsmith
• [Asterisk-Users] Re: Fw: anybody has SIP realtime working ?   Andrew Kohlsmith
• [Asterisk-Users] Extension a?   Andrew Kohlsmith
• [Asterisk-Users] Extension a?   Andrew Kohlsmith
• [Asterisk-Users] RE: IAX Incoming/Outgoing   Andrew Kohlsmith
• [Asterisk-Users] RE: Re: OT: Unblocking bloced CID   Andrew Kohlsmith
• [Asterisk-Users] Need help implementing call center features of Asterisk   Naren Koka
• [Asterisk-Users] registration with different username   Tomas Komarek
• [Asterisk-Users] registration with different username   Tomas Komarek
• [Asterisk-Users] Wanted: Cd-bootable Fedora+Asterisk   Bruce Komito
• [Asterisk-Users] Asterisk transfer conflict   Koopmann, Jan-Peter
• [Asterisk-Users] Re: MOH native files   Koopmann, Jan-Peter
• [Asterisk-Users] Oh this is bad.... bindaddr and rtp traffic   Koopmann, Jan-Peter
• [Asterisk-Users] RE: Echo Cancellation   Koopmann, Jan-Peter
• [Asterisk-Users] CDR problem with TAPI   Koopmann, Jan-Peter
• [Asterisk-Users] Native MOH - Convert mp3 to ulaw   Koopmann, Jan-Peter
• [Asterisk-Users] Speed up dial using #?   Koopmann, Jan-Peter
• [Asterisk-Users] Cisco 7970   Agur Koort
• [Asterisk-Users] TFTP problems on FC4   Agur Koort
• [Asterisk-Users] Installing Cisco IP phone 7910   Agur Koort
• [Asterisk-Users] free tollfree termination   Lukas Kortenhorst
• [Asterisk-Users] Unable to authenticate password - VM   Balint Kovacs
• [Asterisk-Users] Polycom Echo   Anton Krall
• [Asterisk-Users] Polycom Echo   Anton Krall
• [Asterisk-Users] Lowering Server Load   Anton Krall
• [Asterisk-Users] Re: Polycom Default Ring Volume [OT]   Anton Krall
• [Asterisk-Users] Lowering Server Load   Anton Krall
• [Asterisk-Users] Polycom 501   Anton Krall
• [Asterisk-Users] Polycom Echo   Anton Krall
• [Asterisk-Users] Lowering Server Load   Anton Krall
• [Asterisk-Users] asterisk management interface   Anton Krall
• [Asterisk-Users] asterisk management interface   Anton Krall
• [Asterisk-Users] Polycom 501   Anton Krall
• [Asterisk-Users] Polycom 501   Anton Krall
• [Asterisk-Users] dtmf tones problem with unicall and E1   Anton Krall
• [Asterisk-Users] Lowering Server Load   Anton Krall
• [Asterisk-Users] Lowering Server Load   Anton Krall
• [Asterisk-Users] asterisk management interface   Anton Krall
• [Asterisk-Users] dtmf tones problem with unicall and E1   Anton Krall
• [Asterisk-Users] spa3000 asterisk fxo gateway   Anton Krall
• [Asterisk-Users] Polycom voice.gain.tx.analog.handset and asteriskecho   Anton Krall
• [Asterisk-Users] dipura 2002 auto dial or intercom   Anton Krall
• [Asterisk-Users] dipura 2002 auto dial or intercom   Anton Krall
• [Asterisk-Users] stop monitor on transfer   Anton Krall
• [Asterisk-Users] stop monitor on transfer   Anton Krall
• [Asterisk-Users] stop monitor on transfer   Anton Krall
• [Asterisk-Users] stop monitor on transfer   Anton Krall
• [Asterisk-Users] stop monitor on transfer   Anton Krall
• [Asterisk-Users] stop monitor on transfer   Anton Krall
• [Asterisk-Users] slinear bandwidth   Anton Krall
• [Asterisk-Users] slinear bandwidth   Anton Krall
• [Asterisk-Users] stop monitor on transfer   Anton Krall
• [Asterisk-Users] fax receive using TDM400P, with Tzafir, Anton, Cosmin, Colin...   Anton Krall
• [Asterisk-Users] Polycom voice.gain.tx.analog.handsetandasteriskecho   Anton Krall
• [Asterisk-Users] queue_log   Anton Krall
• [Asterisk-Users] Polycom hand/head set echo and Zapata config   Anton Krall
• [Asterisk-Users] queue_log   Anton Krall
• [Asterisk-Users] dipura 2002 auto dial or intercom   Anton Krall
• [Asterisk-Users] stop monitor on transfer   Anton Krall
• [Asterisk-Users] Polycom batphone (was dipura 2002 auto dial or intercom)   Anton Krall
• [Asterisk-Users] Polycom batphone (was dipura 2002 auto dial orintercom)   Anton Krall
• [Asterisk-Users] stop monitor on transfer   Anton Krall
• [Asterisk-Users] Polycom 501 Output volume   Anton Krall
• [Asterisk-Users] mixmonitor and how to stop it   Anton Krall
• [Asterisk-Users] Variables in queues.conf   Stephen Kratzer
• [Asterisk-Users] Agents on DND still receiving calls...   Stephen Kratzer
• [Asterisk-Users] about operator   Olivier Krief
• [Asterisk-Users] about operator   Olivier Krief
• [Asterisk-Users] TDM over PCI bus - Electronic payment terminal   Olivier Krief
• [Asterisk-Users] module load order for Junghanns qozap and TDMcard   Olivier Krief
• [Asterisk-Users] Is LDAP working on 1.2.X   Olivier Krief
• [Asterisk-Users] How to monitor dropped frames on a PCI bus   Olivier Krief
• [Asterisk-Users] Re: Asterisk in production as a fax server, anyone?   Olivier Krief
• [Asterisk-Users] Asterisk in production as a fax server, anyone?   Olivier Krief
• [Asterisk-Users] Re: BRI cards, HFC, and bristuff - a general questionto clear up my understanding.   Olivier Krief
• [Asterisk-Users] Asterisk and Hylafax, on the same box   Olivier Krief
• [Asterisk-Users] zaphfc on an 'actual' asterisk?   Olivier Krief
• [Asterisk-Users] Receptionist Phones   Olivier Krief
• [Asterisk-Users] Re: Asterisk in production as a fax server, anyone?   Olivier Krief
• [Asterisk-Users] check call status during call   Arjan Kroon
• [Asterisk-Users] asterisk perl commands   Arjan Kroon
• [Asterisk-Users] RE: Asterisk-Users Digest, Vol 20, Issue 31   Kaleb L. Kunzler
• [Asterisk-Users] RE: Extension 's' in Realtime   Kaleb L. Kunzler
• [Asterisk-Users] Unable to forward frame   Andy Kuo
• [Asterisk-Users] Max retries exceeded to host...   Andy Kuo
• [Asterisk-Users] Dial plan question - exclamtion mark   Andy Kuo
• [Asterisk-Users] ISDN and Analog DIAL UP Connection Through Asterisk and Digium TE405P   Andy Kuo
• [Asterisk-Users] Re: Jittery meetme conference using Linksys 942 phones   LJ
• [Asterisk-Users] G729 codec problems   Rudolf Ladyzhenskii
• [Asterisk-Users] Mailing list problems with gmail!!!!   Rudolf Ladyzhenskii
• [Asterisk-Users] Voicemail to Email   Rudolf Ladyzhenskii
• [Asterisk-Users] seg fault when skinny phone answers   Ryan Laginski
• [Asterisk-Users] seg fault when skinny phone answers   Ryan Laginski
• [Asterisk-Users] Simple php script to monitor asterisk calls   Sig Lange
• [Asterisk-Users] New ncurses Asterisk Manager Interface   Sig Lange
• [Asterisk-Users] Realtime Extensions   Sig Lange
• [Asterisk-Users] New ncurses Asterisk Manager Interface   Sig Lange
• [Asterisk-Users] New ncurses Asterisk Manager Interface   Sig Lange
• [Asterisk-Users] New ncurses Asterisk Manager Interface   Sig Lange
• [Asterisk-Users] assman, the ncurses asterisk manager interface   Sig Lange
• [Asterisk-Users] echo cancellation   Steven Langley
• [Asterisk-Users] router UDP timeout   Steven Langley
• [Asterisk-Users] D4 AMI - No Caller ID   Steve Langstaff
• [Asterisk-Users] Re: Asterisk at large   Kristian Larsson
• [Asterisk-Users] Clustering   Kristian Larsson
• [Asterisk-Users] Clustering   Kristian Larsson
• [Asterisk-Users] Clustering   Kristian Larsson
• [Asterisk-Users] Asterisk Users Mailing List Traffic   Kristian Larsson
• [Asterisk-Users] Clustering   Niklas Larsson
• [Asterisk-Users] Accept Unregistered GK Calls   Abdul Lateef
• [Asterisk-Users] Lowering Server Load   Andrew Latham
• [Asterisk-Users] A general deployment question (OT)   Andrew Latham
• [Asterisk-Users] An FXO version of IAXy?   Andrew Latham
• [Asterisk-Users] VERY IMPORTANT(TREAT WITH URGENCY)   Andrew Latham
• [Asterisk-Users] An FXO version of IAXy?   Andrew Latham
• [Asterisk-Users] An FXO version of IAXy?   Andrew Latham
• [Asterisk-Users] Can this box handle 8 T1s (PSTN) with Asterisk?   Andrew Latham
• [Asterisk-Users] An FXO version of IAXy?   Andrew Latham
• [Asterisk-Users] Can this box handle 8 T1s (PSTN) with Asterisk?   Andrew Latham
• [Asterisk-Users] Mandrake zaptel module not found after compiling   Andrew Latham
• [Asterisk-Users] GSM/DECT handsets (was gsm picocells)   Andrew Latham
• [Asterisk-Users] Please Help Test Quad PRI Using NFAS   Andrew Latham
• [Asterisk-Users] Please Help Test Quad PRI Using NFAS   Andrew Latham
• [Asterisk-Users] Clustering "NEW THREAD", Almost Working   Benjamin Lawetz
• [Asterisk-Users] Routing SIP calls via URI   Bobby Lee
• [Asterisk-Users] Understanding queue timeouts + possible bug found   Lenz
• [Asterisk-Users] music on hold without mpg123   Lenz
• [Asterisk-Users] simple perl-agi - where's the error?   Lenz
• [Asterisk-Users] Re: Agent in multiple queues?   Lenz
• [Asterisk-Users] Re: Agent in multiple queues?   Lenz
• [Asterisk-Users] Re: Re: Agent in multiple queues?   Lenz
• [Asterisk-Users] I'm FED UP with BroadVoice   Ronald Lewis
• [Asterisk-Users] res_config_mysql.so not found   Xin Li
• [Asterisk-Users] Queues Not Reporting Estimated Hold Time   Michael J. Liberatore
• [Asterisk-Users] Queues Not Reporting Estimated Hold Time   Michael J. Liberatore
• [Asterisk-Users] GS BT102 dual ethernet port -bandwidth impact   Michael J. Liberatore
• [Asterisk-Users] Config TE110P and TDM400 with 2 FXS modules   Paul Liew
• [Asterisk-Users] Config TE110P and TDM400 with 2 FXS modules   Paul Liew
• [Asterisk-Users] sprint FNTM(sp?) line   Greg Lim
• [Asterisk-Users] asterisk@home V's Asterisk   Jonathan Lin
• [Asterisk-Users] (no subject)   Larry Linde
• [Asterisk-Users] NEWS: SIP Firmware Available for Cisco 7970   Mailing List
• [Asterisk-Users] NEWS: SIP Firmware Available for Cisco 7970   Mailing List
• [Asterisk-Users] 7940/60 SIP 8.2   Mailing List
• [Asterisk-Users] 7940/60 SIP 8.2   Mailing List
• [Asterisk-Users] 7940/60 SIP 8.2   Mailing List
• [Asterisk-Users] 7940/60 SIP 8.2   Mailing List
• [Asterisk-Users] Some ignorance here, what exactly is a Session Control Border? (Verizonis asking me about this)   Mailing List
• [Asterisk-Users] ADT/Brinks alarm dialing through Asterisk   Lists
• [Asterisk-Users] ADT/Brinks alarm dialing through Asterisk   Lists
• [Asterisk-Users] HW Echo Cancellers   Rob Lith
• [Asterisk-Users] TDM11B Hang up detection not working in France ?   Rob Lith
• [Asterisk-Users] router UDP timeout   Rob Lith
• [Asterisk-Users] Call Monitor   Dan Littlejohn
• [Asterisk-Users] what are these and can they be fixed?   Dan Littlejohn
• [Asterisk-Users] Linksys WIP300 WiFi Phone   Tim Litwiller
• [Asterisk-Users] Re: AAH lost my IVR phrases   Tim Litwiller
• [Asterisk-Users] testing list mail - please ignore   Tim Litwiller
• [Asterisk-Users] Sipura SPA-3000 and PSTN dtmf   Nilesh Londhe
• [Asterisk-Users] Problem compiling zaptel on latest RHEL kernel(2.6.9-34.EL)   Nilesh Londhe
• [Asterisk-Users] Receptionist Phones (was 3Com Phones)   Darrell Long
• [Asterisk-Users] Receptionist Phones   Darrell Long
• [Asterisk-Users] Receptionist Phones   Darrell Long
• [Asterisk-Users] Cannot log into mailbox , guidance requested   Alexander Lopez
• [Asterisk-Users] Same CID on multiple users(friends9 in SIP.conf   Alexander Lopez
• [Asterisk-Users] Milliwatt Analyzer available   Alexander Lopez
• [Asterisk-Users] Problem ChanSpy   Alexander Lopez
• [Asterisk-Users] Destroying a SIP extension doesn't destroyvoicemail box?is this a bug?   Alexander Lopez
• [Asterisk-Users] Oh this is bad.... bindaddr and rtp traffic   Alexander Lopez
• [Asterisk-Users] Oh this is bad.... bindaddr and rtp traffic   Alexander Lopez
• [Asterisk-Users] Dial command   Alexander Lopez
• [Asterisk-Users] cdr data   Alexander Lopez
• [Asterisk-Users] Dial plans and forwarded phones   Alexander Lopez
• [Asterisk-Users] Analog Desktop Phone   Alexander Lopez
• [Asterisk-Users] Polycom - directory dial   Alexander Lopez
• [Asterisk-Users] Action on phone pickup   Alexander Lopez
• [Asterisk-Users] stop monitor on transfer   Alexander Lopez
• [Asterisk-Users] Callerid on transfer   Alexander Lopez
• [Asterisk-Users] stop monitor on transfer   Alexander Lopez
• [Asterisk-Users] Professional Recordings   Alexander Lopez
• [Asterisk-Users] stop monitor on transfer   Alexander Lopez
• [Asterisk-Users] stop monitor on transfer   Alexander Lopez
• [Asterisk-Users] New ncurses Asterisk Manager Interface   Alexander Lopez
• [Asterisk-Users] OSHA requirement to "reach a live human" ??   Alexander Lopez
• [Asterisk-Users] how to show called name on calling polycomdisplay   Alexander Lopez
• [Asterisk-Users] OSHA requirement to "reach a live human" ??   Alexander Lopez
• [Asterisk-Users] how to show called name on calling polycomdisplay   Alexander Lopez
• [Asterisk-Users] OSHA requirement to "reach a live human" ??   Alexander Lopez
• [Asterisk-Users] Unable to forward frame   Alexander Lopez
• [Asterisk-Users] How to configure PSTN lines permissions todifferent extensions ???   Alexander Lopez
• [Asterisk-Users] Problem with System() command.   Alexander Lopez
• [Asterisk-Users] PRI Setup   Alexander Lopez
• [Asterisk-Users] Feedback from VON expo! Info on * HAandPolycomphone!!   Alexander Lopez
• [Asterisk-Users] Authenticate CDR Logging   Alexander Lopez
• [Asterisk-Users] voip-info.... again   Alexander Lopez
• [Asterisk-Users] OT: Unblocking bloced CID   Alexander Lopez
• [Asterisk-Users] DISA alternative   Alexander Lopez
• [Asterisk-Users] NATted phones transferring calls - BUG0003710   Alexander Lopez
• [Asterisk-Users] Dial from php   Alexander Lopez
• [Asterisk-Users] Asterisk end Festival   Luz Lopez
• [Asterisk-Users] PRI Behavior   Matt Loretitsch
• [Asterisk-Users] Snom 360 - Multiple Server BLF Indications   Stuart Elvish - Dallas Delta Corporation Pty Ltd
• [Asterisk-Users] Snom 360 - Multiple Server BLF Indications   Stuart Elvish - Dallas Delta Corporation Pty Ltd
• [Asterisk-Users] Dial out .call files File permissions??   Bruno De Luca
• [Asterisk-Users] (no subject)   Michel Luczak
• [Asterisk-Users] Real Time Asterisk   Fernando Lujan
• [Asterisk-Users] Real Time Asterisk   Fernando Lujan
• [Asterisk-Users] Real Time Asterisk   Fernando Lujan
• [Asterisk-Users] Real Time Asterisk   Fernando Lujan
• [Asterisk-Users] Real Time Asterisk   Fernando Lujan
• [Asterisk-Users] ODBC and VoiceMail messages.   Fernando Lujan
• [Asterisk-Users] Call Recording?   Fernando Lujan
• [Asterisk-Users] Asterisk Between PBX and FXS   Fernando Lujan
• [Asterisk-Users] Asterisk Between PBX and FXS   Fernando Lujan
• [Asterisk-Users] Asterisk Between PBX and FXS   Fernando Lujan
• [Asterisk-Users] Changing REINVITE status of the channel dynamically   Luki
• [Asterisk-Users] Is it possible to reinvite twice?   Luki
• [Asterisk-Users] Unable to open Asterisk database   Luki
• [Asterisk-Users] Asterisk with Vonage   Luki
• [Asterisk-Users] Cisco Callmanager integration with asterisk   Antonio Serrano Luque
• [Asterisk-Users] VERY IMPORTANT(TREAT WITH URGENCY)   Scott Lykens
• [Asterisk-Users] Double-ring tone   Julian Lyndon-Smith
• [Asterisk-Users] Double-ring tone   Julian Lyndon-Smith
• [Asterisk-Users] Double-ring tone   Julian Lyndon-Smith
• [Asterisk-Users] How to enable talking in chanspy while spying?   Julian Lyndon-Smith
• [Asterisk-Users] Cannot log into mailbox , guidance requested   Doug Lytle
• [Asterisk-Users] error messages on /var/log/asterisk/messages   Doug Lytle
• [Asterisk-Users] List disabled notification   Doug Lytle
• [Asterisk-Users] Auto dial feature   Doug Lytle
• [Asterisk-Users] Meetme Participant Announcement   Doug Lytle
• [Asterisk-Users] Polycom voice.gain.tx.analog.handset and asterisk echo   Doug Lytle
• [Asterisk-Users] Comedian Mail Add-ons?   Doug Lytle
• [Asterisk-Users] Polycom voice.gain.tx.analog.handset and asteriskecho   Doug Lytle
• [Asterisk-Users] I can't receive multiple pages with spandsp   Doug Lytle
• [Asterisk-Users] HW Echo Cancellers   Doug Lytle
• [Asterisk-Users] 7940/60 SIP 8.2   Doug Lytle
• [Asterisk-Users] HW Echo Cancellers   Doug Lytle
• [Asterisk-Users] HW Echo Cancellers   Doug Lytle
• [Asterisk-Users] IVR woes   Doug Lytle
• [Asterisk-Users] MWI to 7960's sometimes delayed or lost. Please advise.   Doug Lytle
• [Asterisk-Users] Outbound paging dialplan example?   Doug Lytle
• [Asterisk-Users] echo cancellation   Doug Lytle
• [Asterisk-Users] OSHA requirement to "reach a live human" ??   Doug Lytle
• [Asterisk-Users] Do Not Disturb?   Doug Lytle
• [Asterisk-Users] cisco 7912 not taking config   Doug Lytle
• [Asterisk-Users] asterisk@home V's Asterisk   Doug Lytle
• [Asterisk-Users] Re: transfers/parked calls + polycom 501   Doug Lytle
• [Asterisk-Users] accessing speed dial database   Doug Lytle
• [Asterisk-Users] Call Pickup Woes   Doug Lytle
• [Asterisk-Users] Call Pickup Woes   Doug Lytle
• [Asterisk-Users] Problem with intermittent one-way audio   Doug Lytle
• [Asterisk-Users] Re: Do Not Disturb?   Doug Lytle
• [Asterisk-Users] Voice mail not working with Asteriks 1.2.5   Doug Lytle
• [Asterisk-Users] Native MOH - Convert mp3 to ulaw   Doug Lytle
• [Asterisk-Users] Native MOH - Convert mp3 to ulaw   Doug Lytle
• [Asterisk-Users] Remote dialtone   Doug Lytle
• [Asterisk-Users] Asterisk--->>Autodialling   Doug Lytle
• [Asterisk-Users] Mandrake zaptel module not found after compiling   Doug Lytle
• [Asterisk-Users] Finding the busypattern   Doug Lytle
• [Asterisk-Users] WARNING[5171]: res_musiconhold.c:833 moh_register: Unable to open pseudo channel for timing   Doug Lytle
• [Asterisk-Users] Polarity reversals on a TE100P   Doug Lytle
• [Asterisk-Users] 7940 with Asterisk?   Doug Lytle
• [Asterisk-Users] Polycom 501 Output volume   Doug Lytle
• [Asterisk-Users] Polycom 501 Output volume   Doug Lytle
• [Asterisk-Users] Polycom 501 Output volume   Doug Lytle
• [Asterisk-Users] aah 2.7 / BRI   Doug Lytle
• [Asterisk-Users] AAH Mailing list   Doug Lytle
• [Asterisk-Users] FOP flash panel: how to reload config files when running   Doug Lytle
• [Asterisk-Users] FOP flash panel: how to reload config files when running   Doug Lytle
• [Asterisk-Users] Asterisk out of Media Path - Call Park   Doug Lytle
• [Asterisk-Users] dial plan logic   Doug Lytle
• [Asterisk-Users] Authenticate   Doug Lytle
• [Asterisk-Users] Reporting?   Doug Lytle
• [Asterisk-Users] PRI issues   Doug Lytle
• [Asterisk-Users] Asterisk Referral - Cleanup on Aisle 7   Doug Lytle
• [Asterisk-Users] Problem getting two x200p cards working on 1.2.4   Guillermo Salas M
• [Asterisk-Users] Problem getting two x200p cards working on 1.2.4   Guillermo Salas M
• [Asterisk-Users] Problem getting two x200p cards working on 1.2.4   Guillermo Salas M
• [Asterisk-Users] asterisk@home and H323   Guillermo Salas M
• [Asterisk-Users] Two FXOs getting bridged?   Julian J. M.
• [Asterisk-Users] How to route incoming calls to different contexts?   Julian J. M.
• [Asterisk-Users] Hangup issues   Julian J. M.
• [Asterisk-Users] Action after _caller_ has hungup(cmd Dial 'g'-option)   Julian J. M.
• [Asterisk-Users] asterisk as a fax server   Julian J. M.
• [Asterisk-Users] sipura spa2 + asterisk bug ?   Julian J. M.
• [Asterisk-Users] SMS in Spain (it seems Protocol 2)   Julian J. M.
• [Asterisk-Users] H323 behind a Firewall   Julian J. M.
• [Asterisk-Users] Re: BRI cards, HFC, and bristuff - a general questionto clear up my understanding.   Julian J. M.
• [Asterisk-Users] Asterisk Failover without SER   Bryan Mahin
• [Asterisk-Users] Asterisk Failover without SER   Bryan Mahin
• [Asterisk-Users] Signate Intro to * - London Training March 21-23   Paul Mahler
• [Asterisk-Users] I can't receive multiple pages with spandsp   Marco Maiolini
• [Asterisk-Users] I can't resume a call on hold from zap device   Marco Maiolini
• [Asterisk-Users] TE40X zapata.conf configuration sample   Angelito Manansala
• [Asterisk-Users] FXS channel banks   Angelito Manansala
• [Asterisk-Users] CLI notice: channel.c:2421 __ast_request_and_dial: Don't know what to do with control frame 15   Angelito Manansala
• [Asterisk-Users] Sound quality issue in one direction and wctdm problem with APIC enabled kernel   Geoff Manning
• [Asterisk-Users] Script to Restart Zaptel   Geoff Manning
• [Asterisk-Users] Re: Script to Restart Zaptel   Geoff Manning
• [Asterisk-Users] Toshiba Strata DK-280 support?   Charles Marcus
• [Asterisk-Users] Toshiba Strata DK-280 support?   Charles Marcus
• [Asterisk-Users] Toshiba Strata DK-280 support?   Charles Marcus
• [Asterisk-Users] New ncurses Asterisk Manager Interface   Charles Marcus
• [Asterisk-Users] Re: regexten   Charles Marcus
• [Asterisk-Users] Asterisk Users Mailing List Traffic   Charles Marcus
• [Asterisk-Users] Asterisk Users Mailing List Traffic   Charles Marcus
• [Asterisk-Users] Phones were working fine - Now there is noaudiowhen calling between extensions   Charles Marcus
• [Asterisk-Users] Fw: anybody has SIP realtime working ?   Charles Marcus
• [Asterisk-Users] Call Recording?   Charles Marcus
• [Asterisk-Users] Call Monitoring?   Charles Marcus
• [Asterisk-Users] Re: problems with emailing voicemail   Charles Marcus
• [Asterisk-Users] Call Recording?   Charles Marcus
• [Asterisk-Users] Dumb question - reaching the PSTN   Charles Marcus
• [Asterisk-Users] AAH: DNID not set if caller suppresses CID?   Hans J. Martin
• [Asterisk-Users] AAH: DNID not set if caller suppresses CID?   Hans J. Martin
• [Asterisk-Users] New ncurses Asterisk Manager Interface   Chris Mason
• [Asterisk-Users] [OT] Polycom provisioning   Marty Mastera
• [Asterisk-Users] problem configuring a digium quad E1 card   David Masure
• [Asterisk-Users] problem configuring a digium quad E1 card   David Masure
• [Asterisk-Users] Polling Asterisk for Life   Matt
• [Asterisk-Users] Re: sipura 841 mass provisioning   Matt
• [Asterisk-Users] Polling Asterisk for Life   Matt
• [Asterisk-Users] Polling Asterisk for Life   Matt
• [Asterisk-Users] Polling Asterisk for Life   Matt
• [Asterisk-Users] Polling Asterisk for Life   Matt
• [Asterisk-Users] Softphone for Windows CE 3.0   Matt
• [Asterisk-Users] HW Echo Cancellers   Matt
• [Asterisk-Users] Is everyone getting mails except me?   Matt
• [Asterisk-Users] Dazed and Confused   Matt
• [Asterisk-Users] Professional Recordings   Matt
• [Asterisk-Users] SIP Jitter Buffer for 1.2.5   Matt
• [Asterisk-Users] SIP Jitter Buffer for 1.2.5   Matt
• [Asterisk-Users] SIP Jitter Buffer for 1.2.5   Matt
• [Asterisk-Users] SIP Jitter Buffer for 1.2.5   Matt
• [Asterisk-Users] Asterilink?!?!   Matt
• [Asterisk-Users] Re: Asterilink?!?!   Matt
• [Asterisk-Users] Aastra Questions   Matt
• [Asterisk-Users] Aastra Questions   Matt
• [Asterisk-Users] Aastra Questions   Matt
• [Asterisk-Users] How often do YOU register?   Matt
• [Asterisk-Users] How often do YOU register?   Matt
• [Asterisk-Users] How often do YOU register?   Matt
• [Asterisk-Users] Problem with chan_iax.c implimentation causes bad audio?   Matt
• [Asterisk-Users] Problem with chan_iax.c implimentation causes bad audio?   Matt
• [Asterisk-Users] Problem with chan_iax.c implimentation causes bad audio?   Matt
• [Asterisk-Users] Problem with chan_iax.c implimentation causes bad audio?   Matt
• [Asterisk-Users] Problem with chan_iax.c implimentation causes bad audio?   Matt
• [Asterisk-Users] PSTN to Asterisk VOIP in Manila   Matt
• [Asterisk-Users] SPA-2002 Upgrade Question   Matt
• [Asterisk-Users] what are these and can they be fixed?   Matt
• [Asterisk-Users] Aastra 9331i phones   Matt
• [Asterisk-Users] Aastra 9331i phones   Matt
• [Asterisk-Users] IAX Bridging and not recording CDR correctly   Matt
• [Asterisk-Users] IAX Bridging and not recording CDR correctly   Matt
• [Asterisk-Users] getting your own phone number   Matt
• [Asterisk-Users] Asterisk billing from CDR database   Matt
• [Asterisk-Users] RE: Re: OT: Unblocking bloced CID   Matt
• [Asterisk-Users] Agent in multiple queues?   Matt
• [Asterisk-Users] Re: Agent in multiple queues?   Matt
• [Asterisk-Users] Reporting?   Matt
• [Asterisk-Users] Re: Agent in multiple queues?   Matt
• [Asterisk-Users] Regulatory Ruling about Caller-ID   Matt
• [Asterisk-Users] Regulatory Ruling about Caller-ID   Matt
• [Asterisk-Users] Regulatory Ruling about Caller-ID   Matt
• [Asterisk-Users] Re: Re: Agent in multiple queues?   Matt
• [Asterisk-Users] Reporting?   Matt
• [Asterisk-Users] Confused on Agents and Queues   Matt
• [Asterisk-Users] 1.2.6 doesn't use mpg123?   Matt
• [Asterisk-Users] Confused on Agents and Queues   Matt
• [Asterisk-Users] 1.2.6 doesn't use mpg123?   Matt
• [Asterisk-Users] CRM + Phones   Adam Mattina
• [Asterisk-Users] Realtime Extensions hint priority   Kevin McAllister
• [Asterisk-Users] Lowering Server Load   Ron McCarthy
• [Asterisk-Users] Is everyone getting mails except me?   Ron McCarthy
• [Asterisk-Users] Clustering   Ron McCarthy
• [Asterisk-Users] Clustering   Ron McCarthy
• [Asterisk-Users] Clustering   Ron McCarthy
• [Asterisk-Users] Clustering   Ron McCarthy
• [Asterisk-Users] Clustering   Ron McCarthy
• [Asterisk-Users] Clustering   Ron McCarthy
• [Asterisk-Users] Clustering   Ron McCarthy
• [Asterisk-Users] [Fwd: Over 40 destinations for FREE!]   trixter aka Bret McDanel
• [Asterisk-Users] Numbered Voicemails even with delete option!   trixter aka Bret McDanel
• [Asterisk-Users] Regulatory Ruling about Caller-ID   trixter aka Bret McDanel
• [Asterisk-Users] 'quit' isn't in the CLI's 'help'   Bob McDowell
• [Asterisk-Users] res_mysql.conf & DNS SRV lookup   Bob McDowell
• [Asterisk-Users] Oh this is bad.... bindaddr and rtp traffic   Bob McDowell
• [Asterisk-Users] Asterisk programmer needed   Bob McDowell
• [Asterisk-Users] Asterisk programmer needed   Bob McDowell
• [Asterisk-Users] Clustering   Bob McDowell
• [Asterisk-Users] stop monitor on transfer   Bob McDowell
• [Asterisk-Users] Clustering   Bob McDowell
• [Asterisk-Users] Looking for docs on adjusting txgain/rxgain   Bob McDowell
• [Asterisk-Users] OT call recording (was stop monitor on transfer)   Bob McDowell
• [Asterisk-Users] List Rules   Bob McDowell
• [Asterisk-Users] Asterisk Native Sounds - in case you missed it...   Bob McDowell
• [Asterisk-Users] Asterisk to receive fax   Bob McDowell
• [Asterisk-Users] echo cancellation   Bob McDowell
• [Asterisk-Users] OSHA requirement to "reach a live human" ??   Bob McDowell
• [Asterisk-Users] OSHA requirement to "reach a live human" ??   Bob McDowell
• [Asterisk-Users] automatic fax detection in asteriskathome   Bob McDowell
• [Asterisk-Users] RE: choppy recorded sounds in asterisk   Bob McDowell
• [Asterisk-Users] Question about advanced IVR   Bob McDowell
• [Asterisk-Users] Aastra Questions   Bob McDowell
• [Asterisk-Users] Aastra Questions   Bob McDowell
• [Asterisk-Users] OT: Good Vendor?   Bob McDowell
• [Asterisk-Users] Native MOH - Convert mp3 to ulaw   Bob McDowell
• [Asterisk-Users] Native MOH - Convert mp3 to ulaw   Bob McDowell
• [Asterisk-Users] Native MOH - Convert mp3 to ulaw   Bob McDowell
• [Asterisk-Users] Native MOH - Convert mp3 to ulaw   Bob McDowell
• [Asterisk-Users] ADT/Brinks alarm dialing through Asterisk   Bob McDowell
• [Asterisk-Users] ADT/Brinks alarm dialing through Asterisk   Bob McDowell
• [Asterisk-Users] RE: Asterisk Users   Bob McDowell
• [Asterisk-Users] Re: problems with emailing voicemail   Bob McDowell
• [Asterisk-Users] Aastra 9331i phones   Bob McDowell
• FW: [Asterisk-Users] Asterisk Users   Bob McDowell
• FW: [Asterisk-Users] Asterisk Users   Bob McDowell
• [Asterisk-Users] RE: Asterisk Failover without SER   Bob McDowell
• [Asterisk-Users] Alarmreciver   Bob McDowell
• [Asterisk-Users] Alarmreciver   Bob McDowell
• [Asterisk-Users] Receptionist Phones   Bob McDowell
• [Asterisk-Users] Receptionist Phones   Bob McDowell
• [Asterisk-Users] Asterisk with Vonage   Bob McDowell
• [Asterisk-Users] Marketing Materials   Bob McDowell
• [Asterisk-Users] Marketing Materials   Bob McDowell
• [Asterisk-Users] Dumb question - reaching the PSTN   Bob McDowell
• [Asterisk-Users] Marketing Materials   Bob McDowell
• [Asterisk-Users] Marketing Materials   Bob McDowell
• [Asterisk-Users] RE: Asterisk as Voicemail Server for Option 61c?   Bob McDowell
• [Asterisk-Users] multiple auto attendants   Bob McDowell
• [Asterisk-Users] Marketing Materials   Bob McDowell
• [Asterisk-Users] IAX: Auto-congesting call due to slow response   Bob McDowell
• [Asterisk-Users] Asterisk and Hylafax, on the same box   Bob McDowell
• [Asterisk-Users] Exchange 12 Unified Messaging   Dr Rodney G. McDuff
• [Asterisk-Users] Confusion about construction of RURIs from contact headers for BYEs generated by *   Dr. Rodney G. McDuff
• [Asterisk-Users] Do Not Disturb?   Brian McEntire
• [Asterisk-Users] Testing asterisk faxing functionality   Corey S. McFadden
• [Asterisk-Users] DTMF leak with IAXy & call waiting Bug?   Zachary McGibbon
• [Asterisk-Users] Preferred editor(s) dialplan coding?   S McGowan
• [Asterisk-Users] Preferred editor(s) dialplan coding?   S McGowan
• [Asterisk-Users] pap2 Dial plan   S McGowan
• [Asterisk-Users] Re: Asterisk-Users Digest, Vol 20, Issue 153   patty McHenry
• [Asterisk-Users] G729 and Meetme   Raymond McKay
• [Asterisk-Users] IVR woes   Robert P. McKenzie
• [Asterisk-Users] IVR woes   Robert P. McKenzie
• [Asterisk-Users] IVR woes   Robert P. McKenzie
• [Asterisk-Users] IVR woes   Robert P. McKenzie
• [Asterisk-Users] Configs for Gradwell and inWeb   Robert P. McKenzie
• [Asterisk-Users] Configs for Gradwell and inWeb   Robert P. McKenzie
• [Asterisk-Users] IVR dial by extension option..   Robert P. McKenzie
• [Asterisk-Users] IVR dial by extension option..   Robert P. McKenzie
• [Asterisk-Users] IVR weirdness   Robert P. McKenzie
• [Asterisk-Users] 7940/60 SIP 8.2   Rob McKrill
• [Asterisk-Users] Clustering "NEW THREAD", Almost Working   Jeremy McNamara
• [Asterisk-Users] Free g729   Jeremy McNamara
• [Asterisk-Users] Preferred editor(s) dialplan coding?   David McNett
• [Asterisk-Users] Spontaneous reloads   McQuiggan, Mark xt46480
• [Asterisk-Users] Problem with libpri?   McQuiggan, Mark xt46480
• [Asterisk-Users] Re: Problem with libpri?   McQuiggan, Mark xt46480
• [Asterisk-Users] Incompatible switchtypes   McQuiggan, Mark xt46480
• [Asterisk-Users] Calls not tearing down properly   McQuiggan, Mark xt46480
• [Asterisk-Users] echo problem + choppy sound   McQuiggan, Mark xt46480
• [Asterisk-Users] [FOLLOWUP]: Calls not tearing down properly   McQuiggan, Mark xt46480
• [Asterisk-Users] Festival tts   Antoine Megalla
• [Asterisk-Users] License for asterisk-addons?   C. J. Meidlinger
• [Asterisk-Users] License for asterisk-addons?   C. J. Meidlinger
• [Asterisk-Users] Can't get incoming call from POTS line...   Tripp Meister
• [Asterisk-Users] echo problem + choppy sound   Giovanni Miano
• [Asterisk-Users] echo problem + choppy sound   Giovanni Miano
• [Asterisk-Users] Config File Management   Giovanni Miano
• [Asterisk-Users] Authorization by ip   Giovanni Miano
• [Asterisk-Users] Call Simulator   Giovanni Miano
• [Asterisk-Users] Re: G729 and Meetme   Bill Michaelson
• [Asterisk-Users] Polycom 501's for sale   Bill Michaelson
• [Asterisk-Users] problem with incoming peer (cisco as5400)   Miguel
• [Fwd: Re: [Asterisk-Users] problem with incoming peer (cisco as5400)]   Miguel
• [Asterisk-Users] ring (hunt?) group   Miguel
• [Asterisk-Users] ring (hunt?) group   Miguel
• [Asterisk-Users] ring (hunt?) group   Miguel
• [Asterisk-Users] Development news :: T38 passthrough support   Miguel
• [Asterisk-Users] sip show inuse not accurate   Miguel
• [Asterisk-Users] Extension a?   Mike
• [Asterisk-Users] FW: Extension a?   Mike
• [Asterisk-Users] Extension a?   Mike
• [Asterisk-Users] Extension a?   Mike
• [Asterisk-Users] Asterisk and Hylafax, on the same box   Mike
• [Asterisk-Users] Asterisk and Hylafax, on the same box   Mike
• [Asterisk-Users] MultiBRI in Australia - found one - maybe   Avi Miller
• [Asterisk-Users] Single E1 with HW Echo Can?   Avi Miller
• [Asterisk-Users] MultiBRI in Australia - found one - maybe   Avi Miller
• [Asterisk-Users] MultiBRI in Australia - found one - maybe   Avi Miller
• [Asterisk-Users] EICON Diva 4BRI   Avi Miller
• [Asterisk-Users] Zaptel compile errors on x86_64   Avi Miller
• [Asterisk-Users] Re: regexten   Avi Miller
• [Asterisk-Users] Call Pickup Woes   Avi Miller
• [Asterisk-Users] best CENTOS to use for latest asterisk   Avi Miller
• [Asterisk-Users] problems with DTMF   Avi Miller
• [Asterisk-Users] Re: best CENTOS to use for latest asterisk   Avi Miller
• [Asterisk-Users] RE: Asterisk Users   Avi Miller
• [Asterisk-Users] [SOLVED] TDM400 FXO module not answering or dialing out.   Avi Miller
• [Asterisk-Users] Re: problems with emailing voicemail   Avi Miller
• [Asterisk-Users] Re: problems with emailing voicemail   Avi Miller
• [Asterisk-Users] Re: problems with emailing voicemail   Avi Miller
• [Asterisk-Users] Re: problems with emailing voicemail   Avi Miller
• [Asterisk-Users] [OT] Polycom provisioning   Avi Miller
• [Asterisk-Users] [OT] Polycom provisioning   Avi Miller
• [Asterisk-Users] [OT] Polycom provisioning   Avi Miller
• [Asterisk-Users] Polycom IP 301 is slow   Avi Miller
• [Asterisk-Users] Polycom IP 301 is slow   Avi Miller
• [Asterisk-Users] Re: AAH lost my IVR phrases   Avi Miller
• [Asterisk-Users] OT: Polycom IP501 and Speed Dials   Avi Miller
• [Asterisk-Users] Reporting?   Avi Miller
• [Asterisk-Users] Echo cancellation problem   Avi Miller
• [Asterisk-Users] Asterisk coder conflicts   Dan Miller
• [Asterisk-Users] PLEASE respond: how to get Asterisk to change coders on RTP handoff??   Dan Miller
• [Asterisk-Users] Re: PLEASE respond: how to get Asterisk to change coders on RTP handoff?? HELLO???   Dan Miller
• [Asterisk-Users] news-reading question   Dan Miller
• [Asterisk-Users] Re: transfers/parked calls + polycom 501   Noah Miller
• [Asterisk-Users] Re: transfers/parked calls + polycom 501   Noah Miller
• [Asterisk-Users] Re: Asterisk large scale, help needed   Noah Miller
• [Asterisk-Users] Re: Asterisk-Users Digest, Vol 20, Issue 91   Noah Miller
• [Asterisk-Users] Re: OSHA requirement to "reach a live human" ??   Noah Miller
• [Asterisk-Users] Re: how to show called name on calling polycom display   Noah Miller
• [Asterisk-Users] Re: how to show called name on calling polycomdisplay   Noah Miller
• [Asterisk-Users] Re: Asterisk Native Sounds - in case you missed it...   Noah Miller
• [Asterisk-Users] Re: transfers/parked calls + polycom 501   Noah Miller
• [Asterisk-Users] Re: Remote MWI over IAX2   Noah Miller
• [Asterisk-Users] Re: [OT] Polycom provisioning   Noah Miller
• [Asterisk-Users] Re: IAX Incoming/Outgoing   Noah Miller
• [Asterisk-Users] Searchable forums   Noah Miller
• [Asterisk-Users] cannot set outgoing cid   Noah Miller
• [Asterisk-Users] OT: Polycom IP501 and Speed Dials   Noah Miller
• [Asterisk-Users] Realtime Users/Peers/Friends - Ick   Noah Miller
• [Asterisk-Users] IAX: Auto-congesting call due to slow response   Noah Miller
• [Asterisk-Users] Asterisk on hosted server   John Millican
• [Asterisk-Users] Asterisk transfer conflict   Mimmus
• [Asterisk-Users] Native attended transfer: taking again original conversation   Mimmus
• [Asterisk-Users] Two PBX   Mimmus
• [Asterisk-Users] Unable to make hints function properly   Mimmus
• [Asterisk-Users] Two Asterisk server   Mimmus
• [Asterisk-Users] echo problem + choppy sound   Mimmus
• [Asterisk-Users] asteriskathome maximun channels per trunk   Mimmus
• [Asterisk-Users] Attended call transfer with GXP-2000   Mimmus
• [Asterisk-Users] Call pickup between different protocols   Mimmus
• [Asterisk-Users] New ncurses Asterisk Manager Interface   Mimmus
• [Asterisk-Users] Countries supporting SMS on PSTN (ISDN)   Mimmus
• [Asterisk-Users] Call Pickup Woes   Mimmus
• [Asterisk-Users] pickup problem   Mimmus
• [Asterisk-Users] pickup problem   Mimmus
• [Asterisk-Users] Zap<-->IAX codec?   Mimmus
• [Asterisk-Users] Zap<-->IAX codec?   Mimmus
• [Asterisk-Users] Zap<-->IAX codec?   Mimmus
• [Asterisk-Users] Failed to read gains: Invalid argument   Mimmus
• [Asterisk-Users] Failed to read gains: Invalid argument   Mimmus
• [Asterisk-Users] automatic callback when busy   Mimmus
• [Asterisk-Users] automatic callback when busy   Mimmus
• [Asterisk-Users] Set caller ID for outgoing PRI calls   Mimmus
• [Asterisk-Users] IAX: Auto-congesting call due to slow response   Mimmus
• [Asterisk-Users] IAX: Auto-congesting call due to slow response   Mimmus
• [Asterisk-Users] Re: [asterisk-biz] Professional Recordings   Stephen Misel
• [Asterisk-Users] status on jitter buffer for SIP/RTP? (OT?)   Adam Moffett
• [Asterisk-Users] impact of qualify=yes   Adam Moffett
• [Asterisk-Users] Jitter buffer for SIP channels (OT?)   Adam Moffett
• [Asterisk-Users] embedded hardware for Asterisk?   Adam Moffett
• [Asterisk-Users] SIP Jitter Buffer for 1.2.5   Adam Moffett
• [Asterisk-Users] hunt groups   Adam Moffett
• [Asterisk-Users] what are these and can they be fixed?   Adam Moffett
• [Asterisk-Users] call drops after one ring   Adam Moffett
• [Asterisk-Users] * dials out zap line first 6 digits, pause, then last digit   Mojo with Horan & Company, LLC
• [Asterisk-Users] Changing caller id on transfer   Mojo with Horan & Company, LLC
• [Asterisk-Users] Changing caller id on transfer   Mojo with Horan & Company, LLC
• [Asterisk-Users] Problem getting two x200p cards working on 1.2.4   Mojo with Horan & Company, LLC
• [Asterisk-Users] Problem getting two x200p cards working on 1.2.4   Mojo with Horan & Company, LLC
• [Asterisk-Users] Unable to start Asterisk 1.2.5 with Asterisk-Addons 1.2.1   Mojo with Horan & Company, LLC
• [Asterisk-Users] Polycom 501   Mojo with Horan & Company, LLC
• [Asterisk-Users] Upgrading Asterisk witk G729 license installed   Mojo with Horan & Company, LLC
• [Asterisk-Users] Re: transfers/parked calls + polycom 501   Mojo with Horan & Company, LLC
• [Asterisk-Users] Simple php script to monitor asterisk calls   Mojo with Horan & Company, LLC
• [Asterisk-Users] Voip-Info   Mojo with Horan & Company, LLC
• [Asterisk-Users] Simple php script to monitor asterisk calls   Mojo with Horan & Company, LLC
• [Asterisk-Users] Simple php script to monitor asterisk calls   Mojo with Horan & Company, LLC
• [Asterisk-Users] Simple php script to monitor asterisk calls   Mojo with Horan & Company, LLC
• [Asterisk-Users] Simple php script to monitor asterisk calls   Mojo with Horan & Company, LLC
• [Asterisk-Users] echo problem + choppy sound   Mojo with Horan & Company, LLC
• [Asterisk-Users] Action on phone pickup   Mojo with Horan & Company, LLC
• [Asterisk-Users] Re: transfers/parked calls + polycom 501   Mojo with Horan & Company, LLC
• [Asterisk-Users] Queues Not Reporting Estimated Hold Time   Mojo with Horan & Company, LLC
• [Asterisk-Users] Feedback from VON expo! Info on * HA and Polycom phone!!   Mojo with Horan & Company, LLC
• [Asterisk-Users] 计划生育的无耻宣传该结束了   Mojo with Horan & Company, LLC
• [Asterisk-Users] 计划生育的无耻宣传该结束了   Mojo with Horan & Company, LLC
• [Asterisk-Users] simple perl-agi - where's the error?   Mojo with Horan & Company, LLC
• [Asterisk-Users] An FXO version of IAXy?   Mojo with Horan & Company, LLC
• [Asterisk-Users] VERY IMPORTANT(TREAT WITH URGENCY)   Mojo with Horan & Company, LLC
• [Asterisk-Users] 计划生育的无耻宣传该结束了   Mojo with Horan & Company, LLC
• [Asterisk-Users] G729 License questions   Mojo with Horan & Company, LLC
• [Asterisk-Users] Is it possible to turn off password for transfers on FOP   Mojo with Horan & Company, LLC
• [Asterisk-Users] Master.csv Shell Script   Mojo with Horan & Company, LLC
• [Asterisk-Users] Call Waiting Issues   Mojo with Horan & Company, LLC
• [Asterisk-Users] Authorization by ip   Mojo with Horan & Company, LLC
• [Asterisk-Users] Authorization by ip   Mojo with Horan & Company, LLC
• [Asterisk-Users] Echo cancellation   Mojo with Horan & Company, LLC
• [Asterisk-Users] Echo cancellation   Mojo with Horan & Company, LLC
• [Asterisk-Users] How to send announcement after called has picked up the phone?   Mojo with Horan & Company, LLC
• [Asterisk-Users] RXgain   Mojo with Horan & Company, LLC
• [Asterisk-Users] Disable polycom call waiting?   Mojo with Horan & Company, LLC
• [Asterisk-Users] OT: Polycom IP501 and Speed Dials   Mojo with Horan & Company, LLC
• [Asterisk-Users] IAXY codec support and questions..   Mojo with Horan & Company, LLC
• [Asterisk-Users] Dial from php   Mojo with Horan & Company, LLC
• [Asterisk-Users] OT: ad-hoc polycom network   Mojo with Horan & Company, LLC
• [Asterisk-Users] Sending 2 CallingNumbers   Markus Monka
• [Asterisk-Users] Festival tts   Vladimir Montealegre
• [Asterisk-Users] RE: Predictive Dialer   Vladimir Montealegre
• [Asterisk-Users] RE: Predictive Dialer   Vladimir Montealegre
• [Asterisk-Users] Keeping the user name in sip INVITE with fixed IP host routing.   Ricardo Monteiro
• [Asterisk-Users] DTMF recognition inconsistent in Asterisk   Monty
• [Asterisk-Users] gsm picocells   Justin Moore
• [Asterisk-Users] Plain Old Answering Machine   Justin Moore
• [Asterisk-Users] Receptionist Phones (was 3Com Phones)   Justin Moore
• [Asterisk-Users] Extensions base policy   Melcon Moraes
• [Asterisk-Users] Can I avoid configuring FXS part in zaptel.conf and zapata.conf   Melcon Moraes
• [Asterisk-Users] How to assign channels for asterisk   Melcon Moraes
• [Asterisk-Users] Extracting info from the $EXTEN variable   Melcon Moraes
• [Asterisk-Users] difference between records in CDR and real duration of call   Melcon Moraes
• [Asterisk-Users] difference between records in CDR and real duration of call   Melcon Moraes
• [Asterisk-Users] Call Pickup Woes   Melcon Moraes
• [Asterisk-Users] IAX Bridging and not recording CDR correctly   Melcon Moraes
• [Asterisk-Users] help on mfc/r2   Melcon Moraes
• [Asterisk-Users] help on mfc/r2   Melcon Moraes
• [Asterisk-Users] compiling Zaptel-1.2.4 on CentOS 4.3   Melcon Moraes
• [Asterisk-Users] help on mfc/r2   Melcon Moraes
• [Asterisk-Users] Disable timeout for answered queue calls?   Melcon Moraes
• [Asterisk-Users] iax limit question   Melcon Moraes
• [Asterisk-Users] Disable timeout for answered queue calls?   Melcon Moraes
• [Asterisk-Users] help on mfc/r2   Melcon Moraes
• [Asterisk-Users] help on mfc/r2   Melcon Moraes
• [Asterisk-Users] help on mfc/r2   Melcon Moraes
• [Asterisk-Users] Alarm on Unicall   Melcon Moraes
• [Asterisk-Users] Re: Alarm on Unicall   Melcon Moraes
• [Asterisk-Users] Re: Alarm on Unicall   Melcon Moraes
• [Asterisk-Users] Re: Alarm on Unicall   Melcon Moraes
• [Asterisk-Users] Asterisk Between PBX and FXS   Melcon Moraes
• [Asterisk-Users] Blacklist out bound numbers from file   Melcon Moraes
• [Asterisk-Users] Please Help Test Quad PRI Using NFAS   Melcon Moraes
• [Asterisk-Users] multiple auto attendants   Melcon Moraes
• [Asterisk-Users] Please Help Test Quad PRI Using NFAS   Melcon Moraes
• [Asterisk-Users] Bad quality between SIP and TDM   Filipe Mordhorst
• RES: [Asterisk-Users] pap2 Dial plan   Filipe Mordhorst
• RES: [Asterisk-Users] Inserting access codes as prefixes to CID   Filipe Mordhorst
• [Asterisk-Users] DTFM or FSK   Filipe Mordhorst
• RES: [Asterisk-Users] DTFM or FSK   Filipe Mordhorst
• RES: [Asterisk-Users] pap2 Dial plan   Filipe Mordhorst
• RES: [Asterisk-Users] DTFM or FSK   Filipe Mordhorst
• [Asterisk-Users] Maximum retries exceeded on transmission   Filipe Mordhorst
• RES: [Asterisk-Users] reload - restart   Filipe Mordhorst
• [Asterisk-Users] res_musiconhold.c: Only wrote -1 of 640 bytes to pipe // no queue music   MorelCom
• [Asterisk-Users] Max retries exceeded to host...   Dan Morin
• [Asterisk-Users] Max retries exceeded to host...   Dan Morin
• [Asterisk-Users] Routing SIP calls via URI   Shad Mortazavi
• [Asterisk-Users] Routing SIP calls via URI   Shad Mortazavi
• [Asterisk-Users] Routing SIP calls via URI   Shad Mortazavi
• [Asterisk-Users] How do you perform a Variable Substitution In Asterisk   Shad Mortazavi
• [Asterisk-Users] Re: MeetMe 'i' option not working correctly?   Tony Mountifield
• [Asterisk-Users] Re: MeetMe Listen Only flag (|m)   Tony Mountifield
• [Asterisk-Users] Re: regexten   Tony Mountifield
• [Asterisk-Users] Re: problems with emailing voicemail   Tony Mountifield
• [Asterisk-Users] Re: Question about meetme app   Tony Mountifield
• [Asterisk-Users] Re: Local Channel   Tony Mountifield
• [Asterisk-Users] Re: FreePBX & AAH   Tony Mountifield
• [Asterisk-Users] nwebmail   Marco Mouta
• [Asterisk-Users] Destroying a SIP extension doesn't destroy voicemail box?is this a bug?   Marco Mouta
• [Asterisk-Users] Receiving Multiple calls on asterisk at home   Marco Mouta
• [Asterisk-Users] Conference room owner Changing his room password? Ast@Home   Marco Mouta
• [Asterisk-Users] Clock is runing too fast, Asterisk@home2.5 Ztdummy and VMware workstation   Marco Mouta
• [Asterisk-Users] 2-Asterisk@Home Servers Connecting Portugal to Brazil (offices)   Marco Mouta
• [Asterisk-Users] Stress Tests from AsteriskGur with Asterisk@Home   Marco Mouta
• [Asterisk-Users] Development news :: T38 passthrough support   Marco Mouta
• [Asterisk-Users] I don't listen first seconds of audio from call - Asterisk integration with old PBX   Marco Mouta
• [Asterisk-Users] 10minutes to restart Asterisk@home 2.7   Marco Mouta
• [Asterisk-Users] Directory doesn't work well Asterisk@home2.7- try from PSTN with Digital recepcionist- Directory based on Last name   Marco Mouta
• [Asterisk-Users] Phones were working fine - Now there is noaudiowhen calling between extensions   Marco Mouta
• [Asterisk-Users] * Meetme Freeze patch found   Marco Mouta
• [Asterisk-Users] Dial out .call files File permissions??   Marco Mouta
• [Asterisk-Users] Dial out .call files File permissions??   Marco Mouta
• [Asterisk-Users] Dial out .call files File permissions??   Marco Mouta
• [Asterisk-Users] Dial out .call files File permissions??   Marco Mouta
• [Asterisk-Users] SJphone Do not send silence - option ? Should be disabled for Asterisk   Marco Mouta
• [Asterisk-Users] Sjphone looses registry in my Asterisk Server then i need to restart Sjphone PC, why?   Marco Mouta
• [Asterisk-Users] Sjphone looses registry in my Asterisk Server then i need to restart Sjphone PC, why?   Marco Mouta
• [Asterisk-Users] Problem with setting ringtones on Cisco 7960 phone.   Greg Mudd
• [Asterisk-Users] Connecting a Grandstream Handytone 486 to Asterisk   Ralf Mueller
• [Asterisk-Users] Asterisk Tools for OSX   Devraj Mukherjee
• [Asterisk-Users] Call Monitoring / Call Takeover with Asterisk   Devraj Mukherjee
• [Asterisk-Users] An FXO version of IAXy?   Steve Murphy
• [Asterisk-Users] OT: Flash/web site developer in Boca Raton FL required   Brian J. Murrell
• [Asterisk-Users] res_features pickupexten   Dinesh Nair
• [Asterisk-Users] Native music on hold - Error   Dinesh Nair
• [Asterisk-Users] Changing caller id on transfer   Dinesh Nair
• [Asterisk-Users] Child PID's   Dinesh Nair
• [Asterisk-Users] Changing caller id on transfer   Dinesh Nair
• [Asterisk-Users] Asterisk 1.2.5 Released   Dinesh Nair
• [Asterisk-Users] Bad Meetme() Bug   Dinesh Nair
• [Asterisk-Users] Meetme Participant Announcement   Dinesh Nair
• [Asterisk-Users] Sync Source: Internally clocked   Dinesh Nair
• [Asterisk-Users] Re: MeetMe 'i' option not working correctly?   Dinesh Nair
• [Asterisk-Users] cdr data   Dinesh Nair
• [Asterisk-Users] IVR woes   Dinesh Nair
• [Asterisk-Users] Dial Out IVR   Dinesh Nair
• [Asterisk-Users] HITBSecConf2006 - Malaysia: Call for Papers   Dinesh Nair
• [Asterisk-Users] Numbered Voicemails when you still delete them.   Dinesh Nair
• [Asterisk-Users] Pickupexten not working   Dinesh Nair
• [Asterisk-Users] VERY IMPORTANT(TREAT WITH URGENCY)   Dinesh Nair
• [Asterisk-Users] VERY IMPORTANT(TREAT WITH URGENCY)   Dinesh Nair
• [Asterisk-Users] Remote dialtone   Dinesh Nair
• [Asterisk-Users] Failed to read gains: Invalid argument   Dinesh Nair
• [Asterisk-Users] How to create [new_context] in extensions.conf?   Dinesh Nair
• [Asterisk-Users] RTP frame size location?   Dinesh Nair
• [Asterisk-Users] Benchmarking an Asterisk Server with 14k users   Dinesh Nair
• [Asterisk-Users] Asterisk, QSIG and Tenovis PBX?   Dinesh Nair
• [Asterisk-Users] transfer incoming call to VM without answering call   Jonathan Nalley
• [Asterisk-Users] Dialogic d/4 PCI   Rene Nelson
• [Asterisk-Users] AstCC   Il Neofita
• [Asterisk-Users] TDM400P busy   Il Neofita
• [Asterisk-Users] TDM400P busy   Il Neofita
• [Asterisk-Users] H323 Info   Il Neofita
• [Asterisk-Users] H323 behind a Firewall   Il Neofita
• [Asterisk-Users] E911 from Remote Office via PRI   Peder at NetworkOblivion
• [Asterisk-Users] RE:Toshiba DK424 / Asterisk / DTMF problems   chan (Alpha Trilogies Networks)
• [Asterisk-Users] G729A   chan (Alpha Trilogies Networks)
• [Asterisk-Users] RE: Problems with installing a TE110P on a Dell Poweredge 850   chan (Alpha Trilogies Networks)
• [Asterisk-Users] GUI Web interface   chan (Alpha Trilogies Networks)
• [Asterisk-Users] PRI Setup   chan (Alpha Trilogies Networks)
• [Asterisk-Users] isdn out of band signalling   Michael Neuhauser
• [Asterisk-Users] MeetMe freezes machine with Junghanns QuadBRI cards   Michael Neuhauser
• [Asterisk-Users] 7970 Configs   Charles A. Newcomer
• [Asterisk-Users] res_mysql.conf & DNS SRV lookup   Ramin Nikaeen
• [Asterisk-Users] MWI & Asterisk Realtime Architecture   Ramin Nikaeen
• [Asterisk-Users] really need help with outgoing calls..PSTN errors   John Novack
• [Asterisk-Users] really need help with outgoing calls..PSTN errors   John Novack
• [Asterisk-Users] Aastra Questions   John Novack
• [Asterisk-Users] Asterisk Users Mailing List Traffic   John Novack
• [Asterisk-Users] DISA alternative   John Novack
• [Asterisk-Users] Problem with chan_iax.c implimentation causes bad audio?   John Novack
• [Asterisk-Users] Native MOH - Convert mp3 to ulaw   John Novack
• [Asterisk-Users] Asterisk Users Mailing List Traffic   John Novack
• [Asterisk-Users] Asterisk Users Mailing List Traffic   John Novack
• [Asterisk-Users] Receptionist Phones   John Novack
• [Asterisk-Users] Asterisk with Vonage   John Novack
• [Asterisk-Users] Receptionist Phones   John Novack
• [Asterisk-Users] IAXY codec support and questions..   John Novack
• [Asterisk-Users] Asterisk Referral - Cleanup on Aisle 7   John Novack
• [Asterisk-Users] 1.2.6 doesn't use mpg123?   John Novack
• [Asterisk-Users] monitor outgoing calls in queue / campaings   Jordan Novak
• [Asterisk-Users] wake up calls   Jordan Novak
• [Asterisk-Users] RE: Asterisk-Users Digest, Vol 20, Issue 13   Jordan Novak
• [Asterisk-Users] web meetme instructions   Jordan Novak
• [Asterisk-Users] webvmail problems   Jordan Novak
• [Asterisk-Users] webvmail   Jordan Novak
• [Asterisk-Users] hotel vmail and iax trouble   Jordan Novak
• [Asterisk-Users] choppy recorded sounds in asterisk   Jordan Novak
• [Asterisk-Users] hunt groups   Jordan Novak
• [Asterisk-Users] Web-ex type solution for use with asterisk   Jordan Novak
• [Asterisk-Users] Mandrake zaptel module not found after compiling   Jordan Novak
• [Asterisk-Users] Mandrake zaptel module not found after compiling   Jordan Novak
• [Asterisk-Users] RXgain   Jordan Novak
• [Asterisk-Users] Callid on T-1 trunk   Jordan Novak
• [Asterisk-Users] asterisk-stat and webmeetme by areski   Jordan Novak
• [Asterisk-Users] Voice problem   Andrew Nowrot
• [Asterisk-Users] Voice problem   Andrew Nowrot
• [Asterisk-Users] Voice problem   Andrew Nowrot
• [Asterisk-Users] Voice problem   Andrew Nowrot
• [Asterisk-Users] Alarmreciver   Andrew Nowrot
• [Asterisk-Users] Alarmreciver   Andrew Nowrot
• [Asterisk-Users] Sangoma A200 error   Michael Kenjie Nukui
• [Asterisk-Users] Music On Hold playback   Amund Nygaard
• [Asterisk-Users] Good VoIP providers that support Asterisk PBX's   Matthew T. O'Connor
• [Asterisk-Users] metermaid patch   Matthew T. O'Connor
• [Asterisk-Users] Sound issues on SIP-SIP calls   Bjorn O
• [Asterisk-Users] Monitoring question   Bjorn O
• [Asterisk-Users] IAX - only one way traffic   Bjorn O
• [Asterisk-Users] TDM11B Hang up detection not working in France ?   Pascal OFFREDO
• [Asterisk-Users] Mitel SX-2000 and Asterisk integration   Richard OSS
• [Asterisk-Users] Mitel SX-2000 <--> TE210P Red Alarm   Richard OSS
• [Asterisk-Users] Failed to read gains: Invalid argument   Richard OSS
• [Asterisk-Users] Failed to read gains: Invalid argument   Richard OSS
• [Asterisk-Users] problem configuring a digium quad E1 card   Richard OSS
• [Asterisk-Users] Mitel 3300 PRI problems   Richard OSS
• [Asterisk-Users] OT - Cisco IP Phone and PC in diferent VLANs (with 802.1x)   Greg Oliver
• [Asterisk-Users] OT - Cisco IP Phone and PC in diferent VLANs (with 802.1x)   Greg Oliver
• [Asterisk-Users] wake up calls   Greg Oliver
• [Asterisk-Users] Program Buttons on Cisco 79xx Phones   Greg Oliver
• [Asterisk-Users] NEWS: SIP Firmware Available for Cisco 7970   Greg Oliver
• [Asterisk-Users] call manager integration   Greg Oliver
• [Asterisk-Users] call manager integration   Greg Oliver
• [Asterisk-Users] NEWS: SIP Firmware Available for Cisco 7970   Greg Oliver
• [Asterisk-Users] NEWS: SIP Firmware Available for Cisco 7970   Greg Oliver
• [Asterisk-Users] Cisco 7960 SIP - Displaying Time   Greg Oliver
• [Asterisk-Users] 7940/60 SIP 8.2   Greg Oliver
• [Asterisk-Users] 7970 Configs   Greg Oliver
• [Asterisk-Users] 7970 Configs   Greg Oliver
• [Asterisk-Users] voip-info.... again   Greg Oliver
• [Asterisk-Users] Cisco POS 3-08-2   Greg Oliver
• [Asterisk-Users] Cisco 7970 SIP Image   Greg Oliver
• [Asterisk-Users] 7970 8.x firmware speeddials   Greg Oliver
• [Asterisk-Users] Re: gsm picocells   Greg Oliver
• [Asterisk-Users] Asterisk 1.2.6, VMWare, & Playback/Background GSM prompts   Greg Oliver
• [Asterisk-Users] Cracked and sleepy voice   Taiwo Oluyemi
• [Asterisk-Users] Inaudible voice and sleepy voice   Taiwo Oluyemi
• [Asterisk-Users] Implementing VoIP for first time with Packetcable   Carlos Alberto Bernat Orozco
• [Asterisk-Users] Asterisk and PacketCable   Carlos Alberto Bernat Orozco
• [Asterisk-Users] Polycom 501 power over ethernet   Chad Osmond
• [Asterisk-Users] Callid on T-1 trunk   Chad Osmond
• [Asterisk-Users] 7960 Cisco SIP Phone TFTP Files   P.A. Oudakker
• [Asterisk-Users] Junghanns, Germany ISDN settings   Florian Overkamp
• [Asterisk-Users] Junghanns, Germany ISDN settings   Florian Overkamp
• [Asterisk-Users] Native MOH - Convert mp3 to ulaw   From PH
• [Asterisk-Users] Asterisk integration with office PBX   John Padovano
• [Asterisk-Users] How is Teliax ?   Andres Paglayan
• [Asterisk-Users] sending SIP text messages to capable phones from an app   Andres Paglayan
• [Asterisk-Users] Voicemail limit?   Ryan Pagquil
• [Asterisk-Users] Voicemail limit?   Ryan Pagquil
• [Asterisk-Users] Voicemail limit?   Ryan Pagquil
• [Asterisk-Users] Voicemail limit?   Ryan Pagquil
• [Asterisk-Users] Voicemail limit?   Ryan Pagquil
• [Asterisk-Users] Development news :: T38 passthrough support   George Pajari
• [Asterisk-Users] Development news :: T38 passthrough support   George Pajari
• [Asterisk-Users] T.38 Passthrough testing -- IAX problem   George Pajari
• [Asterisk-Users] One-Way SIP Audio with SVN Codebase   George Pajari
• [Asterisk-Users] One-Way SIP Audio with SVN Codebase (CANCEL)   George Pajari
• [Asterisk-Users] T38 Passthrough testing -- unknown media type error   George Pajari
• [Asterisk-Users] Stability and motherboard questions with TE406P and TE410P   George Pajari
• [Asterisk-Users] Server freeze with meetme and sip GSM users and ztdummy   Benoit Panizzon
• [Asterisk-Users] Set CallerID to a specific Queue Member   Benoit Panizzon
• [Asterisk-Users] Re: Server freeze with meetme and sip GSM users   Benoit Panizzon
• [Asterisk-Users] Anonymous sip calls getting into wrong context?   Benoit Panizzon
• [Asterisk-Users] Re: Server freeze with meetme and sip GSM users   Benoit Panizzon
• [Asterisk-Users] * Meetme Freeze patch found   Benoit Panizzon
• [Asterisk-Users] * Meetme Freeze patch found   Benoit Panizzon
• [Asterisk-Users] How to send announcement after called has picked up the phone?   Benoit Panizzon
• [Asterisk-Users] zaphfc on an 'actual' asterisk?   Benoit Panizzon
• [Asterisk-Users] zaphfc on an 'actual' asterisk?   Benoit Panizzon
• [Asterisk-Users] mISDN Problem   Benoit Panizzon
• [Asterisk-Users] misdn timeout?   Benoit Panizzon
• [Asterisk-Users] misdn timeout?   Benoit Panizzon
• [Asterisk-Users] my zap channel not ringing   Tim Panton
• [Asterisk-Users] Child PID's   Tim Panton
• [Asterisk-Users] IAX / Firefly handshake problem   Tim Panton
• [Asterisk-Users] Voice problem   Tim Panton
• [Asterisk-Users] Multiple IAX clients behind a firewall   Tim Panton
• [Asterisk-Users] RE: Multiple IAX clients behind a firewall   Tim Panton
• [Asterisk-Users] IAX choppy sound   Tim Panton
• [Asterisk-Users] IAX choppy sound   Tim Panton
• [Asterisk-Users] IAX choppy sound   Tim Panton
• [Asterisk-Users] Realtime SIP users/peers - Screwed?   Tim Panton
• [Asterisk-Users] Annoying Asterisk Realtime Limitation   Tim Panton
• [Asterisk-Users] pickup problem   Tim Panton
• [Asterisk-Users] Problem with chan_iax.c implimentation causes bad audio?   Tim Panton
• [Asterisk-Users] Problem with chan_iax.c implimentationcausesbadaudio?   Tim Panton
• [Asterisk-Users] Problem with chan_iax.cimplimentationcausesbadaudio?   Tim Panton
• [Asterisk-Users] router UDP timeout   Tim Panton
• [Asterisk-Users] RE: IAX Incoming/Outgoing   Tim Panton
• [Asterisk-Users] Using Voicemail with MP3 files...   Tim Panton
• [Asterisk-Users] Re: ON DEMAND call Recording   Tomislav Parcina
• [Asterisk-Users] Re: ON DEMAND call Recording   Tomislav Parcina
• [Asterisk-Users] Asterisk sip and radius authentication   Tomislav Parcina
• [Asterisk-Users] news-reading question   Tomislav Parcina
• [Asterisk-Users] Cisco 7960 SIP - Displaying Time   Tomislav Parcina
• [Asterisk-Users] cdr data   Tomislav Parcina
• [Asterisk-Users] Page about 70 users crash my Asterisk   Alvaro Parres
• [Asterisk-Users] Page about 70 users crash my Asterisk   Alvaro Parres
• [Asterisk-Users] Page about 70 users crash my Asterisk   Alvaro Parres
• [Asterisk-Users] Voicemail limit?   Alvaro Parres
• [Asterisk-Users] Asterisk transfer conflict   Tom Paseka
• [Asterisk-Users] Problem compiling ztdummy on centos 4, 2.6 kernel   Patrick
• [Asterisk-Users] Problem compiling ztdummy on centos 4, 2.6 kernel   Patrick
• [Asterisk-Users] Voice problem   Patrick
• [Asterisk-Users] Stuck. Extenions.conf? Realtime? MySQL? Grrrrr!   Patrick
• [Asterisk-Users] Development news :: T38 passthrough support   Patrick
• [Asterisk-Users] Aterisk with Realtime   Patrick
• [Asterisk-Users] Linksys and sip   Thomas Patterson
• [Asterisk-Users] what version s this??   Paul
• [Asterisk-Users] Need help implementing call center features of Asterisk   Paul
• [Asterisk-Users] MWI to 7960's sometimes delayed or lost. Please advise.   Paul
• [Asterisk-Users] VERY IMPORTANT(TREAT WITH URGENCY)   Paul
• [Asterisk-Users] Asterisk with Vonage   Paul
• [Asterisk-Users] Fw: 2 real phone numbers on one SIP account   Tic Pavlin
• [Asterisk-Users] IAX2 + Sonicwall   Francesco Peeters
• [Asterisk-Users] IAX2 + Sonicwall   Francesco Peeters
• [Asterisk-Users] OT - Cisco IP Phone and PC in diferent VLANs (with 802.1x)   Joao Pereira
• [Asterisk-Users] OT - Cisco IP Phone and PC in diferent VLANs (with 802.1x)   Joao Pereira
• [Asterisk-Users] OT - Cisco IP Phone and PC in diferent VLANs(with 802.1x)   Joao Pereira
• [Asterisk-Users] OT - force Cisco phones to reboot   Joao Pereira
• [Asterisk-Users] OT - force Cisco phones to reboot   Joao Pereira
• [Asterisk-Users] spa3000 asterisk fxo gateway   Roberto Pereyra
• [Asterisk-Users] What codec extensions using now?   Roberto Pereyra
• [Asterisk-Users] hosted billing service   Roberto Pereyra
• [Asterisk-Users] g729 and latency measures   Erick Perez
• [Asterisk-Users] g729 and latency measures   Erick Perez
• [Asterisk-Users] welltech Wellgate 3804 in SIP mode   Erick Perez
• [Asterisk-Users] welltech Wellgate 3804 in SIP mode   Erick Perez
• [Asterisk-Users] WARNING[5171]: res_musiconhold.c:833 moh_register: Unable to open pseudo channel for timing   Erick Perez
• [Asterisk-Users] WARNING[5171]: res_musiconhold.c:833 moh_register: Unable to open pseudo channel for timing   Erick Perez
• [Asterisk-Users] Small - Medium Billing Software needed   Erick Perez
• [Asterisk-Users] Searchable forums   Erick Perez
• [Asterisk-Users] Searchable forums   Erick Perez
• [Asterisk-Users] Intel Compilation Questions - Asterisk 1.2.5/6   Erick Perez
• [Asterisk-Users] Re: Small - Medium Billing Software needed   Erick Perez
• [Asterisk-Users] Unable to open Asterisk database   Erick Perez
• [Asterisk-Users] Unable to open Asterisk database   Erick Perez
• [Asterisk-Users] src/chan_h323.c: Failed to initialize OOH323   Erick Perez
• [Asterisk-Users] welltech Wellgate 3804 in SIP mode   Erick Perez
• [Asterisk-Users] welltech Wellgate 3804 in SIP mode   Erick Perez
• [Asterisk-Users] Problem with two cards Digium   Infobox Peru
• [Asterisk-Users] problem with tdm22b   Infobox Peru
• [Asterisk-Users] TDM400P busy   Infobox Peru
• [Asterisk-Users] MultiBRI in Australia - found one - maybe   David Phelan
• [Asterisk-Users] MultiBRI in Australia - found one - maybe   David Phelan
• [Asterisk-Users] MultiBRI in Australia - found one - maybe   David Phelan
• [Asterisk-Users] MultiBRI in Australia - found one - maybe   David Phelan
• [Asterisk-Users] MultiBRI in Australia - found one - maybe   David Phelan
• [Asterisk-Users] HFC USB (was MultiBRI in Australia - found one -maybe)   David Phelan
• [Asterisk-Users] VERY IMPORTANT(TREAT WITH URGENCY)   David Phelan
• [Asterisk-Users] GSM/DECT handsets (was gsm picocells)   David Phelan
• [Asterisk-Users] LCDPROC cient for Asterisk   Mark Phillips
• [Asterisk-Users] Avaya IP Office 412   Mark Phillips
• [Asterisk-Users] Problem with poud key (#)   Matteo Piazza
• [Asterisk-Users] Polycom voice.gain.tx.analog.handset and asterisk echo   Wilson Pickett
• [Asterisk-Users] Polycom voice.gain.tx.analog.handset and asteriskecho   Wilson Pickett
• [Asterisk-Users] Odd CID issue calling SIP to SIP DID - anyone have this or can explain it?   Wilson Pickett
• [Asterisk-Users] cidname via IAX2?   Wilson Pickett
• [Asterisk-Users] Dial Out IVR   Wilson Pickett
• [Asterisk-Users] Multiple IAX clients behind a firewall   Wilson Pickett
• [Asterisk-Users] VERY IMPORTANT(TREAT WITH URGENCY)   Wilson Pickett
• [Asterisk-Users] Asterisk Users Mailing List Traffic   Wilson Pickett
• [Asterisk-Users] FOP flash panel: how to reload config files when running   Wilson Pickett
• [Asterisk-Users] about operator   Bartosz Piec
• [Asterisk-Users] how to run asterisk?   Bartosz Piec
• [Asterisk-Users] Calls forwarding to numbers only in user's context   Bartosz Piec
• [Asterisk-Users] Preferred editor(s) dialplan coding?   Bartosz Piec
• [Asterisk-Users] Calls forwarding to numbers only in user's context   Bartosz Piec
• [Asterisk-Users] How to transmit Video   Bartosz Piec
• [Asterisk-Users] accessing speed dial database   Ian Pilkington
• [Asterisk-Users] Action on phone pickup   Karl O. Pinc
• [Asterisk-Users] Zaptel not compiling on lastest Centos 4.2 kernel.   Karl O. Pinc
• [Asterisk-Users] SIP Jitter Buffer for 1.2.5   Rosario Pingaro
• [Asterisk-Users] why incoming DATA CALLS are answered as VOICE by asterisk IVR?   Pisac
• [Asterisk-Users] DATA CALLS annoying my system   Pisac
• [Asterisk-Users] OSHA requirement to "reach a live human" ??   Scott Plante
• [Asterisk-Users] OSHA requirement to "reach a live human" ??   Scott Plante
• [Asterisk-Users] invoking a macro doesn't work   Don Pobanz
• [Asterisk-Users] Remote dialtone   Don Pobanz
• [Asterisk-Users] Call Recording?   Don Pobanz
• [Asterisk-Users] ISDN and Analog DIAL UP Connection Through Asterisk and Digium TE405P   Don Pobanz
• [Asterisk-Users] ISDN and Analog DIAL UP Connection Through Asterisk and Digium TE405P   Don Pobanz
• [Asterisk-Users] Re: Asterisk in production as a fax server, anyone?   Don Pobanz
• [Asterisk-Users] Polycom IP501 Buddy List   Mike Pollitt
• [Asterisk-Users] Asterisk as Voicemail Server for Option 61c?   Porier, Jeremy M.
• [Asterisk-Users] Problems Configuring Cisco 12SP+   Juanjo Portela
• [Asterisk-Users] Problems Configuring Cisco 12SP+   Juanjo Portela
• [Asterisk-Users] VoIP prepaid billing   Voipers Portugal
• [Asterisk-Users] VoIP prepaid billing   Voipers Portugal
• [Asterisk-Users] Flash zap trunk from Softphone or IP Handsets...   Marc-andre Poupier
• [Asterisk-Users] HITBSecConf2006 - Malaysia: Call for Papers   Praburaajan
• [Asterisk-Users] Development news :: T38 passthrough   Paolo Prandini
• [Asterisk-Users] new beta Grandstream firmware HT488_496_386   Russ Price
• [Asterisk-Users] Problem compiling zaptel on latest RHEL kernel (2.6.9-34.EL)   Russ Price
• [Asterisk-Users] Hangup issues   Carlos Prieto
• [Asterisk-Users] Polling Asterisk for Life   Cosmin Prund
• [Asterisk-Users] Changing caller id on transfer   Cosmin Prund
• [Asterisk-Users] Re: Asterisk at large   Cosmin Prund
• [Asterisk-Users] Changing caller id on transfer   Cosmin Prund
• [Asterisk-Users] Changing caller id on transfer   Cosmin Prund
• [Asterisk-Users] What hardware to use for ISDN in Romania   Cosmin Prund
• [Asterisk-Users] What hardware to use for ISDN in Romania   Cosmin Prund
• [Asterisk-Users] Changing caller id on transfer   Cosmin Prund
• [Asterisk-Users] Changing caller id on transfer   Cosmin Prund
• [Asterisk-Users] Problem with System() command.   Cosmin Prund
• [Asterisk-Users] snom 320 MWI light   Joe Pukepail
• [Asterisk-Users] Send One Touch Record to mail   Joe Pukepail
• [Asterisk-Users] Asterisk integration with office PBX   Joe Pukepail
• [Asterisk-Users] web meetme instructions   Ben Q
• [Asterisk-Users] Asterisk Users   QUICK, RANDY
• [Asterisk-Users] compiling Zaptel-1.2.4 on CentOS 4.3   Mark Quitoriano
• [Asterisk-Users] compiling Zaptel-1.2.4 on CentOS 4.3   Mark Quitoriano
• [Asterisk-Users] Voicemail limit?   Antonio Rabena
• [Asterisk-Users] Voicemail limit?   Antonio Rabena
• [Asterisk-Users] Incoming Asterisk SIP DID Calls   Antonio Rabena
• [Asterisk-Users] Info about F1000G   Aryanto Rachmad
• [Asterisk-Users] Re: Info about F1000G   Aryanto Rachmad
• [Asterisk-Users] 3Com Phones   Radcliffe
• [Asterisk-Users] asterisk@home V's Asterisk   David Rahn
• [Asterisk-Users] Asterisk Users Mailing List Traffic   David Rahn
• [Asterisk-Users] An FXO version of IAXy?   David Rahn
• [Asterisk-Users] ISDN BRI and UK Premium Rate Numbers   Faris Raouf
• [Asterisk-Users] Queues Not Reporting Estimated Hold Time   Faris Raouf
• [Asterisk-Users] ISDN BRI and UK Premium Rate Numbers   Faris Raouf
• [Asterisk-Users] Asterisk @ Home 2.6 Call hangs up   Jerry Rasmussen
• [Asterisk-Users] Asterisk @ Home 2.6 Call hangs up   Jerry Rasmussen
• [Asterisk-Users] Asterisk @ Home 2.6 Call hangs up   Jerry Rasmussen
• [Asterisk-Users] Oneway voice   Jerry Rasmussen
• [Asterisk-Users] Exchange 12 Unified Messaging   Jerry Rasmussen
• [Asterisk-Users] Asterisk 1.2.6, VMWare, & Playback/Background GSM prompts   Jerry Rasmussen
• [Asterisk-Users] Nortel Meridian Opt 81C/11c and PRI   Steve Rawlings
• [Asterisk-Users] Nortel Meridian Opt 81C/11c and PRI   Steve Rawlings
• [Asterisk-Users] Howto cut the first digit   Trevor Raynsford
• [Asterisk-Users] Same CID on multiple users(friends9 in SIP.conf   Tele Cost Price Reducer
• [Asterisk-Users] dial plan !!   Tele Cost Price Reducer
• [Asterisk-Users] How to route incoming calls to different contexts?   Tele Cost Price Reducer
• [Asterisk-Users] 160 analogue phones..   Tele Cost Price Reducer
• [Asterisk-Users] Dial plans and forwarded phones   Tele Cost Price Reducer
• [Asterisk-Users] Grandstream unit HT-488   Tele Cost Price Reducer
• [Asterisk-Users] FXS channel banks   Tele Cost Price Reducer
• [Asterisk-Users] FXS channel banks   Tele Cost Price Reducer
• [Asterisk-Users] Set caller ID for outgoing PRI calls   Tele Cost Price Reducer
• [Asterisk-Users] Connecting a Grandstream Handytone 486 to Asterisk   Tele Cost Price Reducer
• [Asterisk-Users] asterisk turn key solution   Tele Cost Price Reducer
• [Asterisk-Users] Polycom IP 301 is slow   Walt Reed
• [Asterisk-Users] Primary D-Channel on span 1 down   Christian Reelfs
• [Asterisk-Users] Howto Cut the first digit   Christian Reelfs
• [Asterisk-Users] Howto cut the first digit   Christian Reelfs
• [Asterisk-Users] perl AGI won't run from extensions.conf   Richard Reina
• [Asterisk-Users] asterisk doesn't wait for whole extension   Sebastian Reitenbach
• [Asterisk-Users] asterisk doesn't wait for whole extension   Sebastian Reitenbach
• [Asterisk-Users] asterisk doesn't wait for whole extension   Sebastian Reitenbach
• [Asterisk-Users] cannot set outgoing cid   Sebastian Reitenbach
• [Asterisk-Users] transfer calls via Manager Api   Stefan Reuter
• [Asterisk-Users] Programming the Manager API   Stefan Reuter
• [Asterisk-Users] transfer calls via Manager Api   Stefan Reuter
• [Asterisk-Users] NEWS: SIP Firmware Available for Cisco 7970   John Reynolds
• [Asterisk-Users] How is Teliax ?   John Reynolds
• [Asterisk-Users] chan_nbs and nbs   Christopher Rhodes
• [Asterisk-Users] [OT?]SCCP image for cisco 7905g   Simone Ricci
• [Asterisk-Users] TDM400 FXO module not answering or dialing out.   Hadley Rich
• [Asterisk-Users] TDM400 FXO module not answering or dialing out.   Hadley Rich
• [Asterisk-Users] TDM400 FXO module not answering or dialing out.   Hadley Rich
• [Asterisk-Users] [SOLVED] TDM400 FXO module not answering or dialing out.   Hadley Rich
• [Asterisk-Users] "Not Found" in archive   Hadley Rich
• [Asterisk-Users] How to create [new_context] in extensions.conf?   Hadley Rich
• [Asterisk-Users] Lowering Server Load   Gary Richardson
• [Asterisk-Users] OT - Cisco IP Phone and PC in diferent VLANs (with 802.1x)   Gary Richardson
• [Asterisk-Users] MixMonitor Problems -- sssshh, don't be too loud   Gary Richardson
• [Asterisk-Users] Autofill phonebook??   Gary Richardson
• [Asterisk-Users] Autofill phonebook??   Gary Richardson
• [Asterisk-Users] Program Buttons on Cisco 79xx Phones   Gary Richardson
• [Asterisk-Users] MixMonitor Problems -- sssshh, don't be too loud   Gary Richardson
• [Asterisk-Users] MixMonitor Problems -- sssshh, don't be too loud   Gary Richardson
• [Asterisk-Users] Asterisk Users   Gary Richardson
• [Asterisk-Users] Asterisk Users   Gary Richardson
• [Asterisk-Users] Call Monitoring?   Gary Richardson
• FW: [Asterisk-Users] Asterisk Users   Gary Richardson
• [Asterisk-Users] Config File Management   Gary Richardson
• [Asterisk-Users] Testing asterisk faxing functionality   Gary Richardson
• [Asterisk-Users] DUNDi Public and Private Key Question   JR Richardson
• [Asterisk-Users] Clustering   JR Richardson
• [Asterisk-Users] Clustering   JR Richardson
• [Asterisk-Users] Regexten & Regcontext   JR Richardson
• [Asterisk-Users] Re: Regexten & Regcontext, working now   JR Richardson
• [Asterisk-Users] Clustering "NEW THREAD", Almost Working   JR Richardson
• [Asterisk-Users] Re: Clustering "NEW THREAD", Almost Working   JR Richardson
• [Asterisk-Users] Clustering "NEW THREAD", Almost Working   JR Richardson
• [Asterisk-Users] Regcontext, only 1 context available?   JR Richardson
• [Asterisk-Users] RE: Clustering   JR Richardson
• [Asterisk-Users] RE: DUNDi .... Halfway and CLUSTERING   JR Richardson
• [Asterisk-Users] RE: Clustering   JR Richardson
• [Asterisk-Users] Remote MWI over IAX2   JR Richardson
• [Asterisk-Users] VoiceMailMain(@context) Problem with Option 5 (Advanced)   JR Richardson
• [Asterisk-Users] Fw: anybody has SIP realtime working ?   JR Richardson
• [Asterisk-Users] RE: VoiceMailMain(@context) Problem with Option 5 (Advanced)   JR Richardson
• [Asterisk-Users] RE: IAX Incoming/Outgoing   JR Richardson
• [Asterisk-Users] RE: Asterisk-Users Digest, Vol 20, Issue 184   JR Richardson
• [Asterisk-Users] RE: Hopefully a Simple Question?   JR Richardson
• [Asterisk-Users] Asterisk to MySQL Data Lookup Warning Message?   JR Richardson
• [Asterisk-Users] MOH native files   Steven Ringwald
• [Asterisk-Users] [1.2.5] DTMF not being set correctly (RESEND)   Steven Ringwald
• [Asterisk-Users] [1.2.5] DTMF not being set correctly (RESEND)   Steven Ringwald
• [Asterisk-Users] Child PID's   Luigi Rizzo
• [Asterisk-Users] regexten   Luigi Rizzo
• [Asterisk-Users] regexten   Luigi Rizzo
• [Asterisk-Users] regexten   Luigi Rizzo
• [Asterisk-Users] How to create [new_context] in extensions.conf?   Luigi Rizzo
• [Asterisk-Users] MixMonitor   Alex Robertson
• [Asterisk-Users] Re: [asterisk-biz] Professional Recordings   Adam Robins
• [Asterisk-Users] Festival tts   Adam Robins
• [Asterisk-Users] Festival tts   Adam Robins
• [Asterisk-Users] Festival tts   Adam Robins
• [Asterisk-Users] Problem with chan_iax.c implimentation causesbad audio?   Adam Robins
• [Asterisk-Users] Problem with chan_iax.c implimentation causesbadaudio?   Adam Robins
• [Asterisk-Users] Problem with chan_iax.c implimentationcausesbadaudio?   Adam Robins
• [Asterisk-Users] Problem with chan_iax.cimplimentationcausesbadaudio?   Adam Robins
• [Asterisk-Users] Problem with chan_iax.cimplimentationcausesbadaudio?   Adam Robins
• [Asterisk-Users] Inter-Asterisk Using SIP   Adam Robins
• [Asterisk-Users] Countries supporting SMS on PSTN (ISDN)   Tim Robinson
• [Asterisk-Users] zaphfc on an 'actual' asterisk?   Tim Robinson
• [Asterisk-Users] IAX Video and Meetme   Hagen Rode
• [Asterisk-Users] RE: Multiple IAX clients behind a firewall   Hagen Rode
• [Asterisk-Users] RE: Echo Cancellation   Hagen Rode
• [Asterisk-Users] Re: router UDP timeout   Hagen Rode
• [Asterisk-Users] Asterisk + SE Linux   Anthony Rodgers
• [Asterisk-Users] IAXy (S101) echo?   Anthony Rodgers
• [Asterisk-Users] Problem compiling zaptel on latest RHEL kernel (2.6.9-34.EL)   Anthony Rodgers
• [Asterisk-Users] (no subject)   Anthony Rodgers
• [Asterisk-Users] Problem compiling zaptel on latest RHEL kernel(2.6.9-34.EL)   Anthony Rodgers
• [Asterisk-Users] how to connect 3 or more servers via IAX ?   Anthony Rodgers
• [Asterisk-Users] CDR Bug?   Anthony Rodgers
• [Asterisk-Users] PRI DMS100 -> Nortel Meridian Option 81   Anthony Rodgers
• [Asterisk-Users] Nortel Meridian Opt 81C/11c and PRI   Anthony Rodgers
• [Asterisk-Users] PRI DMS100 -> Nortel Meridian Option 81   Anthony Rodgers
• [Asterisk-Users] Asterisk Users   Anthony Rodgers
• [Asterisk-Users] Autofill phonebook??   Amaury Rodriguez
• [Asterisk-Users] Can One FXO Support Multiple Phone Lines?   James Ronald
• [Asterisk-Users] IAX / Firefly handshake problem   Michael van Rooyen
• [Asterisk-Users] Re: MOH native files   Matt Roth
• [Asterisk-Users] Milliwatt Analyzer available   Matt Roth
• [Asterisk-Users] Lowering Server Load   Matt Roth
• [Asterisk-Users] Brainstorming dual-core and Asterisk   Matt Roth
• [Asterisk-Users] Asterisk programmer needed   Matt Roth
• [Asterisk-Users] music on hold without mpg123   Matt Roth
• [Asterisk-Users] VERY IMPORTANT(TREAT WITH URGENCY)   Matt Roth
• [Asterisk-Users] Asterisk & SMP: Is irqbalance Redundant on 2.6 Kernels?   Matt Roth
• [Asterisk-Users] Asterisk & SMP: Is irqbalance Redundant on 2.6 Kernels? - Resolved   Matt Roth
• [Asterisk-Users] Asterisk & SMP: Is irqbalance Redundant on 2.6 Kernels? - Resolved   Matt Roth
• [Asterisk-Users] Re: Asterisk in production as a fax server, anyone?   Matt Roth
• [Asterisk-Users] setmusiconhold doesn't work between 2 SIP phones   Joseph Rothstein
• [Asterisk-Users] Queues - calls going to agents lised as "In use"   Joseph Rothstein
• [Asterisk-Users] TFTP problems on FC4   Joseph Rothstein
• [Asterisk-Users] Netgear FS116P and Cisco 79XX phones   Joseph Rothstein
• [Asterisk-Users] dropping voice frame ulaw - slin?   Joseph Rothstein
• [Asterisk-Users] Problem with Queue periodic announcemnets   Joseph Rothstein
• [Asterisk-Users] MixMonitor Problems -- sssshh, don't be too loud   Brian Roy
• [Asterisk-Users] IAX2 + Sonicwall   Brian Roy
• [Asterisk-Users] IAX2 + Sonicwall   Brian Roy
• [Asterisk-Users] Toshiba Strata DK-280 support?   Brian Roy
• [Asterisk-Users] Asterisk Users Mailing List Traffic   Brian Roy
• [Asterisk-Users] cdr_odbc appears to have fields missing   Brian Roy
• [Asterisk-Users] cdr_odbc appears to have fields missing   Brian Roy
• [Asterisk-Users] most common VOIP echo simulaton for research purposes ?   Robert Rozman
• [Asterisk-Users] Professional Recordings   Waldo Rubinstein
• [Asterisk-Users] Arcavox / AV-Global   Waldo Rubinstein
• [Asterisk-Users] FreePBX & AAH   Waldo Rubinstein
• [Asterisk-Users] Asterisk and LCR   Waldo Rubinstein
• [Asterisk-Users] Asterisk and LCR   Waldo Rubinstein
• [Asterisk-Users] Test   RumaTech
• [Asterisk-Users] (no subject)   Vitaliy S
• [Asterisk-Users] Program Buttons on Cisco 79xx Phones   Omar A. Sabek
• [Asterisk-Users] Cisco 7960 SIP - Displaying Time   Omar A. Sabek
• [Asterisk-Users] 7940/60 SIP 8.2   Omar A. Sabek
• [Asterisk-Users] Cisco 7960 SIP - Displaying Time   Omar A. Sabek
• [Asterisk-Users] Cisco 7960 SIP - Displaying Time   Omar A. Sabek
• [Asterisk-Users] Cisco 7960 SIP - Displaying Time   Omar A. Sabek
• [Asterisk-Users] Cisco 7960 8.2 callerID lists proxy?   Omar A. Sabek
• [Asterisk-Users] Authenticated SIP NOtify with 1.2.4?   Alberto Sagredo
• [Asterisk-Users] Can log into the mailbox from Soft-phone , but not from Hardware Phone   Alberto Sagredo
• [Asterisk-Users] Development news :: T38 passthrough support   Alberto Sagredo
• [Asterisk-Users] Ability to put call on hold via manager?   Alberto Sagredo
• [Asterisk-Users] SMS in Spain (it seems Protocol 2)   Alberto Sagredo
• [Asterisk-Users] H323 behind a Firewall   Alberto Sagredo
• [Asterisk-Users] Extensions base policy   Mohammad Salaque
• [Asterisk-Users] Extensions base policy   Mohammad Salaque
• [Asterisk-Users] Extensions base policy   Mohammad Salaque
• [Asterisk-Users] Extensions base policy   Mohammad Salaque
• [Asterisk-Users] Asterisk and gateway   Mohammad Salaque
• [Asterisk-Users] Asterisk and gateway   Mohammad Salaque
• [Asterisk-Users] Asterisk and gateway   Mohammad Salaque
• [Asterisk-Users] Dialling Problem   Mohammad Salaque
• [Asterisk-Users] Dialling Problem   Mohammad Salaque
• [Asterisk-Users] WARNING[3675] samples/codec_g723.c: Received a G.723.1 frame that was 4 bytes from RTP   Mohammad Salaque
• [Asterisk-Users] What codec extensions using now?   Mohammad Salaque
• [Asterisk-Users] asterisk silence suppression?   Juan Salas
• [Asterisk-Users] a=fmtp:18 annexb=no   Juan Salas
• [Asterisk-Users] G729, G729 annex A or G729 annex B?   Juan Salas
• [Asterisk-Users] How to transmit Video   Juan Salas
• [Asterisk-Users] oh323 signal update support   Juan Salas
• [Asterisk-Users] agi channel status   Danish Samad
• [Asterisk-Users] Asterisk transfer conflict   Michael Sampson
• [Asterisk-Users] channels appear to be stuck   Michael Sampson
• [Asterisk-Users] Sipura RMA   Michael Sampson
• [Asterisk-Users] Echo Cancelation on TE110P   Michael Sampson
• [Asterisk-Users] Asterisk Fax Question   Michael Sampson
• [Asterisk-Users] 160 analogue phones..   Michael Sampson
• [Asterisk-Users] Flash on Unicall Channel   Paulo Scardine
• [Asterisk-Users] Reverse group in zapata.conf   Greg Scasny
• [Asterisk-Users] RE: Stable Hardware Combination Experiences   Scheller, Bob
• [Asterisk-Users] PRI/T-1   Greg Schiedler
• [Asterisk-Users] Re: chan_capi-cm-0.6.4   Armin Schindler
• [Asterisk-Users] Re: chan_capi-cm-0.6.4   Armin Schindler
• [Asterisk-Users] Working Asterisk with Austrian ISDN p2p   Armin Schindler
• [Asterisk-Users] Re: chan_capi-cm-0.6.4   Armin Schindler
• [Asterisk-Users] Working Asterisk with Austrian ISDN p2p   Armin Schindler
• [Asterisk-Users] Working Asterisk with Austrian ISDN p2p   Armin Schindler
• [Asterisk-Users] MultiBRI in Australia - found one - maybe   Armin Schindler
• [Asterisk-Users] EICON Diva 4BRI   Armin Schindler
• [Asterisk-Users] AVM C2 chan_capi-cm-0.6.4 Error on Dial   Armin Schindler
• [Asterisk-Users] AVM C2 chan_capi-cm-0.6.4 Error on Dial   Armin Schindler
• [Asterisk-Users] capiHOLD missing in BRIstuff 0.3.0   Armin Schindler
• [Asterisk-Users] Asterisk and Hylafax, on the same box   Armin Schindler
• [Asterisk-Users] Asterisk and Hylafax, on the same box   Armin Schindler
• [Asterisk-Users] Asterisk and Hylafax, on the same box   Armin Schindler
• [Asterisk-Users] Echo still present with Eicon Diva Server 4 Bri   Armin Schindler
• [Asterisk-Users] Asterisk and Hylafax, on the same box   Armin Schindler
• [Asterisk-Users] Asterisk and Hylafax, on the same box   Armin Schindler
• [Asterisk-Users] Re: chan_capi-cm-0.6.4   Ralf Schlatterbeck
• [Asterisk-Users] Re: chan_capi-cm-0.6.4   Ralf Schlatterbeck
• [Asterisk-Users] Working Asterisk with Austrian ISDN p2p   Ralf Schlatterbeck
• [Asterisk-Users] Re: chan_capi-cm-0.6.4   Ralf Schlatterbeck
• [Asterisk-Users] Echo Cancellation   Keith Schmidt
• [Asterisk-Users] capiHOLD missing in BRIstuff 0.3.0   Peer Oliver Schmidt
• [Asterisk-Users] Receptionist Phones   Peer Oliver Schmidt
• [Asterisk-Users] Milliwatt Analyzer available   Roger Schreiter
• [Asterisk-Users] remote IP address in channel?   Roger Schreiter
• [Asterisk-Users] test call quality   Roger Schreiter
• [Asterisk-Users] Re: Milliwatt Analyzer available   Roger Schreiter
• [Asterisk-Users] slinear bandwidth   Roger Schreiter
• [Asterisk-Users] slinear bandwidth   Roger Schreiter
• [Asterisk-Users] FAX over PRI   Roger Schreiter
• [Asterisk-Users] How to nice agi scripts?   Roger Schreiter
• [Asterisk-Users] How to disable event_log?   Roger Schreiter
• [Asterisk-Users] Child PID's   Matt Schulte
• [Asterisk-Users] ooh323 Gatekeeper Bug   Richard Scobie
• [Asterisk-Users] dial plan logic   Miles Scruggs
• [Asterisk-Users] dial plan logic   Miles Scruggs
• [Asterisk-Users] incoming triggers seperate outbound   Miles Scruggs
• [Asterisk-Users] FXS Caller ID?   Lee Chit Seong
• [Asterisk-Users] webvmail   Kris Seraphine
• [Asterisk-Users] Transfer to specific park number   Kris Seraphine
• [Asterisk-Users] Stability and motherboard questions with TE406P and TE410P   Kyle Sexton
• [Asterisk-Users] Asterisk--->>Autodialling   Kyle Sexton
• [Asterisk-Users] Stability and motherboard questions with TE406P and TE410P   Kyle Sexton
• [Asterisk-Users] Stability and motherboard questions with TE406P and TE410P   Kyle Sexton
• [Asterisk-Users] Zap not installing   Curt Shaffer
• [Asterisk-Users] Re: Zap not installing   Curt Shaffer
• [Asterisk-Users] Failed installing zaptel   Curt Shaffer
• [Asterisk-Users] FXS channel banks   Curt Shaffer
• [Asterisk-Users] FXS channel banks   Curt Shaffer
• [Asterisk-Users] 3Com Phones   Curt Shaffer
• [Asterisk-Users] Receptionist Phones (was 3Com Phones)   Curt Shaffer
• [Asterisk-Users] Asterisk with Vonage   Curt Shaffer
• [Asterisk-Users] eyeBeam v1.1   Curt Shaffer
• [Asterisk-Users] Dialplan : Forwarding call to voicemail after one ring iif extension is busy   Navneet Shah
• [Asterisk-Users] MWI, SER and asterisk   Sharon
• [Asterisk-Users] GXP-2000 Volume Issue   Clint Sharp
• [Asterisk-Users] GXP-2000 Volume Issue   Clint Sharp
• [Asterisk-Users] caller anounce   Shaun
• [Asterisk-Users] Re: caller anounce   Shaun
• [Asterisk-Users] Polycom Echo   Stagg Shelton
• [Asterisk-Users] TE411P VPM   Stagg Shelton
• [Asterisk-Users] ALSA channel (console/dsp) problem   Mohammad Shokuie
• [Asterisk-Users] Polling Asterisk for Life   Andreas Sikkema
• [Asterisk-Users] IVR woes   Wiley Siler
• [Asterisk-Users] Asterisk transfer conflict   Moises Silva
• [Asterisk-Users] Preferred editor(s) dialplan coding?   Moises Silva
• [Asterisk-Users] Programming the Manager API   Moises Silva
• [Asterisk-Users] Failed to read gains: Invalid argument   Moises Silva
• [Asterisk-Users] ring noise at the background   Baris Simsek
• [Asterisk-Users] Toll free nos   San Singhania
• [Asterisk-Users] Passing Digits between ISDN PBX and Asterisk   Garth van Sittert
• [Asterisk-Users] Zaptel compile errors   Garth van Sittert
• [Asterisk-Users] Zaptel compile errors   Garth van Sittert
• [Asterisk-Users] Development news :: T38 passthrough support   James Sizemore
• [Asterisk-Users] Any new Voice Recognition devs?   Neil Skowronek
• [Asterisk-Users] Forward from SER to asterisk can't hang up   Bart J. Smit
• [Asterisk-Users] Sample SER + Asterisk conf?   Bart J. Smit
• [Asterisk-Users] SIP routing over IAX2   Bart J. Smit
• [Asterisk-Users] SIP routing over IAX2   Bart J. Smit
• [Asterisk-Users] Calls to SIP providers   Bart J. Smit
• [Asterisk-Users] Auto dial feature   Kevin Smith
• [Asterisk-Users] Agents and agent counts   Kevin Smith
• [Asterisk-Users] Polycom - directory dial   Kevin Smith
• [Asterisk-Users] Polycom hand/head set echo and Zapata config   Kevin Smith
• [Asterisk-Users] Polycom 601 Message Center   Kevin Smith
• [Asterisk-Users] Polycom 601 Message Center   Kevin Smith
• [Asterisk-Users] Polycom 601 Message Center   Kevin Smith
• [Asterisk-Users] Polycom 601 Message Center   Kevin Smith
• [Asterisk-Users] PRI issues   Kevin Smith
• [Asterisk-Users] PRI issues   Kevin Smith
• [Asterisk-Users] Confused on Agents and Queues   Kevin Smith
• [Asterisk-Users] Disappearing voicemail   Richard Smith
• [Asterisk-Users] LNP / DID Service - Louisianna / Virginia   Rick Smith
• [Asterisk-Users] Polycom 501's for sale   Rick Smith
• [Asterisk-Users] Polycom 501's for sale   Rick Smith
• [Asterisk-Users] AstriCon Europe: Early Bird Open / Speakers & Papers Wanted   Steven Sokol
• [Asterisk-Users] SER & Asterisk with DID incoming and out going   Andrei Sotirov
• [Asterisk-Users] Asterisk RealTime Question, Please help   Peter Spikings
• [Asterisk-Users] Asterisk RealTime Question, Please help   Peter Spikings
• [Asterisk-Users] ANI configuration   Nripendra Srivastava
• [Asterisk-Users] Program Buttons on Cisco 79xx Phones   Kevin Steil
• [Asterisk-Users] Program Buttons on Cisco 79xx Phones   Kevin Steil
• [Asterisk-Users] Program Buttons on Cisco 79xx Phones   Kevin Steil
• [Asterisk-Users] An FXO version of IAXy?   Kevin Steil
• [Asterisk-Users] reload - restart   Johann Steinwendtner
• [Asterisk-Users] MOH native files   Chris Stenton
• [Asterisk-Users] Re: MOH native files   Chris Stenton
• [Asterisk-Users] Cisco 7960 SIP - Displaying Time   Chris Stenton
• [Asterisk-Users] Cisco 7960 SIP - Displaying Time   Chris Stenton
• [Asterisk-Users] Cisco 7960 8.2 callerID lists proxy?   Chris Stenton
• [Asterisk-Users] music on hold without mpg123   Chris Stenton
• [Asterisk-Users] Budgetone strange problem - have to press hold on and off to connect call.   Chris Stenton
• [Asterisk-Users] Festival tts   Steven
• [Asterisk-Users] Re: Outbound paging dialplan example?   Steven
• [Asterisk-Users] Re: Asterisk Native Sounds - in case you missed it...   Steven
• [Asterisk-Users] best CENTOS to use for latest asterisk   Steven
• [Asterisk-Users] Re: best CENTOS to use for latest asterisk   Steven
• [Asterisk-Users] Re: compiling Zaptel-1.2.4 on CentOS 4.3   Steven
• [Asterisk-Users] Re: AAH lost my IVR phrases   Steven
• [Asterisk-Users] Re: Re: AAH lost my IVR phrases   Steven
• [Asterisk-Users] Re: Asterisk in production as a fax server, anyone?   Steven
• [Asterisk-Users] 7940 with Asterisk?   Skeeve Stevens
• [Asterisk-Users] snom 320 MWI light   Christian Stredicke
• [Asterisk-Users] Receptionist Phones   Christian Stredicke
• [Asterisk-Users] incoming calls dropout on PRI over TE110p   James Sturges
• [Asterisk-Users] Unable to forward frame   James Sturges
• [Asterisk-Users] Unable to forward frame   James Sturges
• [Asterisk-Users] AMP and ABE   James Sturges
• [Asterisk-Users] Asterisk Users Mailing List Traffic   James Sturges
• [Asterisk-Users] How to enable talking in chanspy while spying?   James Sturges
• [Asterisk-Users] AMP and ABE   James Sturges
• [Asterisk-Users] Call Wrap-Up   James Sturges
• [Asterisk-Users] 20 seconds til voice transmission starts   Cornelius Suermann
• [Asterisk-Users] Re: 20 seconds til voice transmission starts   Cornelius Suermann
• [Asterisk-Users] anonymous caller id causes crash   Cornelius Suermann
• [Asterisk-Users] sipura spa2 + asterisk bug ?   Tofik Suleymanov
• [Asterisk-Users] sipura spa2 + asterisk bug ?   Tofik Suleymanov
• [Asterisk-Users] sipura spa2 + asterisk bug ?   Tofik Suleymanov
• [Asterisk-Users] sipura spa2 + asterisk bug ?   Tofik Suleymanov
• [Asterisk-Users] sipura spa2 + asterisk bug ?   Tofik Suleymanov
• [Asterisk-Users] ISDN and Analog DIAL UP Connection Through Asterisk and Digium TE405P   Tofik Suleymanov
• [Asterisk-Users] Chinaroby VOIP phones?   Darko Sundek
• [Asterisk-Users] Chinaroby VOIP phones? SECOND TIME!   Darko Sundek
• [Asterisk-Users] static kernel   Paolo Supino
• [Asterisk-Users] multiple auto attendants   Paolo Supino
• [Asterisk-Users] Professional Recordings   Technical Support
• [Asterisk-Users] problems with emailing voicemail   Technical Support
• [Asterisk-Users] txfax problem   Technical Support
• [Asterisk-Users] Asterisk 1.2.6, VMWare, & Playback/Background GSM prompts   Technical Support
• [Asterisk-Users] Asterisk 1.2.6, VMWare, & Playback/BackgroundGSM prompts   Technical Support
• [Asterisk-Users] [HELP] Outbound Channel next priority on originator disconnect   Asterisk Supporter
• [Asterisk-Users] [HELP] dial plan continue for outbound channel on disconnect   Asterisk Supporter
• [Asterisk-Users] OT: Unblocking bloced CID   Shanon Swafford
• [Asterisk-Users] asterisk crash too much?   TC
• [Asterisk-Users] RE : RE : [asterisk-dev] iax failure?   Administrator TOOTAI
• SV: [Asterisk-Users] IAX - only one way traffic   Administrator TOOTAI
• [Asterisk-Users] Re: uniqueid   Carlo Taguinod
• [Asterisk-Users] How to configure PSTN lines permissions to different extensions ???   Carlo Taguinod
• [Asterisk-Users] RE: Snom 360 problems   Usman Tahir
• [Asterisk-Users] RE: Snom 360 problems   Usman Tahir
• [Asterisk-Users] RE: Snom 360 problems   Usman Tahir
• [Asterisk-Users] GSM phone reception range extendor   Sam Tam
• [Asterisk-Users] Anyone using the GSMgateway from CyberTelecom ?   Sam Tam
• RE: [Asterisk-Users] 计划生育的无耻宣传该结束了   Sam Tam
• [Asterisk-Users] Authorization by ip   Sam Tam
• [Asterisk-Users] Installing the none commercial intel g729 codecsinto Asterisk@Home 2.2?   Sam Tam
• [Asterisk-Users] Trouble installing a digium TE405P card   Mads B. Tandrup
• [Asterisk-Users] Re: Asterisk at large   Joseph Tanner
• [Asterisk-Users] Re: Asterisk at large   Joseph Tanner
• [Asterisk-Users] Polling Asterisk for Life   Joseph Tanner
• [Asterisk-Users] Changing caller id on transfer   Joseph Tanner
• [Asterisk-Users] Polling Asterisk for Life   Joseph Tanner
• [Asterisk-Users] Polling Asterisk for Life   Joseph Tanner
• [Asterisk-Users] Polling Asterisk for Life   Joseph Tanner
• [Asterisk-Users] Polling Asterisk for Life   Joseph Tanner
• [Asterisk-Users] Polling Asterisk for Life   Joseph Tanner
• [Asterisk-Users] Does an entry in AstDB stay after reboot?   Joseph Tanner
• [Asterisk-Users] How to route incoming calls to different contexts?   Joseph Tanner
• [Asterisk-Users] really need help with outgoing calls..PSTN errors   Joseph Tanner
• [Asterisk-Users] really need help with outgoing calls..PSTN errors   Joseph Tanner
• [Asterisk-Users] Two asterisks on one machine   Joseph Tanner
• [Asterisk-Users] Asterisk download file locations   Joseph Tanner
• [Asterisk-Users] Question from a newbie on finding digium hosts   Joseph Tanner
• [Asterisk-Users] Any way to change dns timeout value? Asterisk hangs if internet unreachable   Joseph Tanner
• [Asterisk-Users] System Design   Joseph Tanner
• [Asterisk-Users] PAP2 won't make two g729 calls at the same time   Joseph Tanner
• [Asterisk-Users] Memory Problems   Joseph Tanner
• [Asterisk-Users] Memory Problems   Joseph Tanner
• [Asterisk-Users] System Design   Joseph Tanner
• [Asterisk-Users] HW Echo Cancellers   Joseph Tanner
• [Asterisk-Users] Asterisk @ Home 2.6 Call hangs up   Joseph Tanner
• [Asterisk-Users] Asterisk Native Sounds - in case you missed it...   Joseph Tanner
• [Asterisk-Users] Can anybody get me setup with a hosted asterisk@home box or virtual server in the next 24 hours?   Joseph Tanner
• [Asterisk-Users] Polycom 501   Tarpo, Louie
• [Asterisk-Users] Inserting access codes as prefixes to CID   AR Tarzi
• [Asterisk-Users] Dialplan - strip IDD prefix and insert another   AR Tarzi
• [Asterisk-Users] Dialplan - strip IDD prefix and insert another   AR Tarzi
• [Asterisk-Users] difference between records in CDR and realduration of call   AR Tarzi
• [Asterisk-Users] stop monitor on transfer   AR Tarzi
• [Asterisk-Users] GSM/DECT handsets (was gsm picocells)   AR Tarzi
• [Asterisk-Users] Polycom IP 301 is slow   AR Tarzi
• [Asterisk-Users] oh323 - unable to install   AR Tarzi
• [Asterisk-Users] my zap channel not ringing   Viktor Tatianin
• [Asterisk-Users] asterisk@home and H323   Viktor Tatianin
• [Asterisk-Users] asterisk@home and H323   Viktor Tatianin
• [Asterisk-Users] Asterisk 1.2.6 and Zaptel 1.2.5 Released   Asterisk Development Team
• [Asterisk-Users] Asterisk 1.2.6 and Zaptel 1.2.5 Released   Asterisk Development Team
• [Asterisk-Users] Asterisk 1.2.5 Released   The Asterisk Development Team
• [Asterisk-Users] Asterisk-addons 1.2.2 released   The Asterisk Development Team
• [Asterisk-Users] Polycom 501   MBIT Technologies
• [Asterisk-Users] Polycom 501   MBIT Technologies
• [Asterisk-Users] Polycom 501   MBIT Technologies
• [Asterisk-Users] asterisk@home and H323   MBIT Technologies
• [Asterisk-Users] Polycom 501 Output volume   MBIT Technologies
• [Asterisk-Users] MFCR2   Rodrigo P. Telles
• [Asterisk-Users] ANNOUNCE: WIST - Web Interface for SIP Trace   Rodrigo P. Telles
• [Asterisk-Users] What is asterisk   Melisa Teoh
• [Asterisk-Users] Does Atcom AU-200 work with XLite?   Melisa Teoh
• [Asterisk-Users] Hopefully a Simple Question?   Clint Tevlin
• [Asterisk-Users] Re: Best GUI for basic HostedPBX service   Peeramate at SIPPhone Thailand
• [Asterisk-Users] RE: Delay in ringing   Ash Thakrar
• [Asterisk-Users] Analog Desktop Phone   Thczv F. Thczv
• [Asterisk-Users] Analog Desktop Phone   Thczv F. Thczv
• [Asterisk-Users] Analog Desktop Phone   Thczv F. Thczv
• [Asterisk-Users] Analog Desktop Phone   Thczv F. Thczv
• [Asterisk-Users] Analog Desktop Phone   Thczv F. Thczv
• [Asterisk-Users] Analog Desktop Phone   Thczv F. Thczv
• [Asterisk-Users] I'm FED UP with BroadVoice   Thczv F. Thczv
• [Asterisk-Users] Queue and busy/congested ZAP channels   Christian Theune
• [Asterisk-Users] call files and cdr I need src different from CallerID(number)   Thomas
• [Asterisk-Users] Hardware Requirements for 1M minutes   David Thomas
• [Asterisk-Users] Hardware Requirements for 1M minutes   David Thomas
• [Asterisk-Users] MWI, SER and asterisk   David Thomas
• [Asterisk-Users] MWI, SER and asterisk   David Thomas
• [Asterisk-Users] Feedback from VON expo! Info on * HA and Polycom phone!!   David Thomas
• [Asterisk-Users] Re: DUNDi .... Halfway and CLUSTERING   David Thomas
• [Asterisk-Users] Sticky Problem SER/Asterisk   David Thomas
• [Asterisk-Users] Re: DUNDi .... Halfway and CLUSTERING   David Thomas
• [Asterisk-Users] Re: DUNDi .... Halfway and CLUSTERING   David Thomas
• [Asterisk-Users] Feedback from VON expo! Info on * HAandPolycomphone!!   David Thomas
• [Asterisk-Users] Realtime SIP Persistency   David Thomas
• [Asterisk-Users] IVR woes   Rob Thomas
• [Asterisk-Users] Re: Asterisk perms in manager.conf   Stefan Tichy
• [Asterisk-Users] Managed Switches QoS to deal with network bottleneck   Mark Tinka
• [Asterisk-Users] 7960 Cisco SIP Phone TFTP Files   Mark Tinka
• [Asterisk-Users] 7960 Cisco SIP Phone TFTP Files   Mark Tinka
• [Asterisk-Users] sporadic voicemail delete problems   Paul Tinsley
• [Asterisk-Users] RTP frame size location?   John Todd
• [Asterisk-Users] How to install the cdr_odbc module.   Jan du Toit
• [Asterisk-Users] Problems loading res_odbc.so and cdr_odbc.so   Jan du Toit
• [Asterisk-Users] Professional Recordings   Tom
• [Asterisk-Users] Sangoma A101 T1/E1 (PRI) voip card available for testing   Tom
• [Asterisk-Users] MeetMe - Causes * to crash :/   Brent Torrenga
• [Asterisk-Users] RE: MeetMe freezes machine with Junghanns   Brent Torrenga
• [Asterisk-Users] Re: * Meetme Freeze patch found   Brent Torrenga
• [Asterisk-Users] Re: How is Teliax ?   Brent Torrenga
• [Asterisk-Users] Preferred editor(s) dialplan coding?   Steve Totaro
• [Asterisk-Users] re: Sixtel Services   Steve Totaro
• [Asterisk-Users] Getting True ANI not Caller ID   Steve Totaro
• [Asterisk-Users] Getting True ANI not Caller ID   Steve Totaro
• [Asterisk-Users] FXS channel banks   Steve Totaro
• [Asterisk-Users] RE: IAX Incoming/Outgoing   Steve Totaro
• [Asterisk-Users] RE: IAX Incoming/Outgoing   Steve Totaro
• [Asterisk-Users] RE: IAX Incoming/Outgoing   Steve Totaro
• [Asterisk-Users] Web based voicemail client   Steve Totaro
• [Asterisk-Users] Polycom IP 301 is slow   Steve Totaro
• [Asterisk-Users] Call Simulator   Steve Totaro
• [Asterisk-Users] * Meetme Freeze patch found   Steve Totaro
• [Asterisk-Users] Ability to put call on hold via manager?   Steve Totaro
• [Asterisk-Users] Ability to put call on hold via manager?   Steve Totaro
• [Asterisk-Users] Call Monitoring / Call Takeover with Asterisk   Steve Totaro
• [Asterisk-Users] Please Help Test Quad PRI Using NFAS   Steve Totaro
• [Asterisk-Users] Please Help Test Quad PRI Using NFAS   Steve Totaro
• [Asterisk-Users] Quintum Tenor DX4060   Steve Totaro
• [Asterisk-Users] How to enable talking in chanspy while spying?   Steven Totaro
• [Asterisk-Users] OT - Cisco IP Phone and PC in diferent VLANs(with 802.1x)   Wojciech Tryc
• [Asterisk-Users] OT - Cisco IP Phone and PC in different VLANs(with802.1x)   Wojciech Tryc
• [Asterisk-Users] Polycom - directory dial   Wojciech Tryc
• [Asterisk-Users] chan_h323 problem   Ganbold Tsagaankhuu
• [Asterisk-Users] HOWTO initialize new kernel & kernel source without reboot   Justin Tunney
• [Asterisk-Users] ODBC and VoiceMail messages.   Justin Tunney
• [Asterisk-Users] Asterisk eating CPU   Justin Tunney
• [Asterisk-Users] TFTP problems on FC4   Justin Tunney
• [Asterisk-Users] Regulatory Ruling about Caller-ID   Justin Tunney
• [Asterisk-Users] Regulatory Ruling about Caller-ID   Justin Tunney
• [Asterisk-Users] [OT] Polycom provisioning   Alyed Tzompa
• [Asterisk-Users] kernel recompilation on a asterisk server   Alyed Tzompa
• [Asterisk-Users] User Extension Custom Voicemail   Alyed Tzompa
• [Asterisk-Users] Asterisk and LCR   Alyed Tzompa
• [Asterisk-Users] Dumb question - reaching the PSTN   Alyed Tzompa
• [Asterisk-Users] Asterisk Between PBX and FXS   Alyed Tzompa
• [Asterisk-Users] G.726 codec - can we select bandwidth?   Steve Underwood
• [Asterisk-Users] ADPCM - vs - G.726   Steve Underwood
• [Asterisk-Users] If a packet is dropped, is it possible to replay the previous packet in Asterisk?   Steve Underwood
• [Asterisk-Users] Alarm on Unicall   Steve Underwood
• [Asterisk-Users] Span monitoring   Steve Underwood
• [Asterisk-Users] Re: Asterisk in production as a fax server, anyone?   Steve Underwood
• [Asterisk-Users] H.264   Erick Weber V.
• [Asterisk-Users] H.264   Erick Weber V.
• [Asterisk-Users] Question about Polycom 601 and expansion module.   Olger Merlos V.
• [Asterisk-Users] On site installtion Tech. wanted   VOICEIN at aol.com
• [Asterisk-Users] Asterisk Users Mailing List Traffic   Jens Vagelpohl
• [Asterisk-Users] Which Mac OSX softphone with IAX2 support?   Jens Vagelpohl
• [Asterisk-Users] SIP trunk problem   George Vagenas
• [Asterisk-Users] SIP trunk problem   George Vagenas
• [Asterisk-Users] 3Com Phones   Jared Valentine
• [Asterisk-Users] Help with Digium TE210P, TDM400P card in Dell PE830   Ondrej Valousek
• [Asterisk-Users] 7970 Configs   Joel Vandal
• [Asterisk-Users] Help! Connecting two Astersik via SIP channels   Alejandro Vargas
• [Asterisk-Users] How to receive faxes with SPA3000 and Asterisk setup   Alejandro Vargas
• [Asterisk-Users] What is asterisk   Alejandro Vargas
• [Asterisk-Users] sending text to display of sip phones   Alejandro Vargas
• [Asterisk-Users] mediatrix 1102   Alejandro Vargas
• [Asterisk-Users] pap2 Dial plan   Alejandro Vargas
• [Asterisk-Users] spa 3000/2100 noise   Alejandro Vargas
• [Asterisk-Users] Asterisk to receive fax   Alejandro Vargas
• [Asterisk-Users] external modem   Alejandro Vargas
• [Asterisk-Users] asteriskathome maximun channels per trunk   Alejandro Vargas
• [Asterisk-Users] SIP routing over IAX2   Alejandro Vargas
• [Asterisk-Users] Re: did from sip trunk   Alejandro Vargas
• [Asterisk-Users] asterisk configurations   Alejandro Vargas
• [Asterisk-Users] automatic fax detection in asteriskathome   Alejandro Vargas
• [Asterisk-Users] asterisk and skype - asterisk newbie   Alejandro Vargas
• [Asterisk-Users] asterisk and skype - asterisk newbie   Alejandro Vargas
• [Asterisk-Users] Asterisk and CISCO 7970 color   Diego Mariano Velo
• [Asterisk-Users] Grandstream unit HT-488   Oliver Vermeulen
• [Asterisk-Users] DID's Now Offering Romania Bucharest 4021+ and 4031+   Oliver Vermeulen
• [Asterisk-Users] Configuration call hijack for users in a hunting group ?   Maxim Vexler
• [Asterisk-Users] Autodial   Vidar
• [Asterisk-Users] Autodial   Vidar
• [Asterisk-Users] Call Parking Grandstream   Alfie Viechweg
• [Asterisk-Users] Parked calls delay   Tom Vile
• [Asterisk-Users] Sipura RMA   Tom Vile
• [Asterisk-Users] Sipura RMA   Tom Vile
• [Asterisk-Users] new beta Grandstream firmware HT488_496_386   Tom Vile
• [Asterisk-Users] new beta Grandstream firmware HT488_496_386   Tom Vile
• [Asterisk-Users] low call volume   Tom Vile
• [Asterisk-Users] new beta Grandstream firmware HT488_496_386   Tom Vile
• [Asterisk-Users] new beta Grandstream firmware HT488_496_386   Tom Vile
• [Asterisk-Users] PAP2 won't make two g729 calls at the same time   Tom Vile
• [Asterisk-Users] PAP2 won't make two g729 calls at the same time   Tom Vile
• [Asterisk-Users] asterisk@home V's Asterisk   Tom Vile
• [Asterisk-Users] asterisk@home V's Asterisk   Tom Vile
• [Asterisk-Users] asterisk@home V's Asterisk   Tom Vile
• [Asterisk-Users] asterisk@home V's Asterisk   Tom Vile
• [Asterisk-Users] Re: problems with emailing voicemail   Tom Vile
• [Asterisk-Users] Call Pickup Woes   Tom Vile
• [Asterisk-Users] Comments on Gafachi   Tom Vile
• [Asterisk-Users] FreePBX & AAH   Tom Vile
• [Asterisk-Users] sending SIP text messages to capable phones from an app   Tom Vile
• [Asterisk-Users] caller anounce   Tom Vile
• [Asterisk-Users] cannot set outgoing cid   Tom Vile
• [Asterisk-Users] Dial from php   Tom Vile
• [Asterisk-Users] Play wav while in connection with a caller   Tom Vile
• [Asterisk-Users] Problem with HT-286 & BT-101   Todd Vinson
• [Asterisk-Users] icmp 36: 192.168.30.32 udp port 5004 unreachable   Todd Vinson
• [Asterisk-Users] Sipura SPA-3000 and PSTN dtmf   Vladyslav
• [Asterisk-Users] Sipura SPA-3000 and PSTN dtmf   Vladyslav
• [Asterisk-Users] Predictive Dialer   Adam Vocks
• [Asterisk-Users] RE: Predictive Dialer   Adam Vocks
• [Asterisk-Users] RE: Predictive Dialer   Adam Vocks
• [Asterisk-Users] RE: Predictive Dialer   Adam Vocks
• [Asterisk-Users] Callerid on transfer   Ronald Voermans
• [Asterisk-Users] Callerid on transfer   Ronald Voermans
• [Asterisk-Users] time update (7905)   Tomislav Vojvodic
• [Asterisk-Users] Dial out .call files File permissions??   Tomislav Vojvodic
• [Asterisk-Users] Dial out .call files File permissions??   Tomislav Vojvodic
• [Asterisk-Users] Dial out .call files File permissions??   Tomislav Vojvodic
• [Asterisk-Users] time update (7905)   Tomislav Vojvodic
• [Asterisk-Users] Set caller ID for outgoing PRI calls   Tomislav Vojvodic
• [Asterisk-Users] Set caller ID for outgoing PRI calls   Tomislav Vojvodic
• [Asterisk-Users] Set caller ID for outgoing PRI calls   Tomislav Vojvodic
• [Asterisk-Users] open source queue analyzer   Terry Wade
• [Asterisk-Users] Random Zap port going crazy When channel released after a flash.   Dennis Walker
• [Asterisk-Users] IAXY codec support and questions..   Michael Wallette
• [Asterisk-Users] Re: Asterisk-Users Digest, Vol 20, Issue 226   Michael Wallette
• [Asterisk-Users] priorityjumping=no   Watkins, Bradley
• [Asterisk-Users] Asterisk Native Sounds - in case you missed it...   Watkins, Bradley
• [Asterisk-Users] DUNDi .... Halfway   Watkins, Bradley
• [Asterisk-Users] RE: DUNDi .... Halfway and CLUSTERING   Watkins, Bradley
• [Asterisk-Users] Re: DUNDi .... Halfway and CLUSTERING   Watkins, Bradley
• [Asterisk-Users] Re: DUNDi .... Halfway and CLUSTERING   Watkins, Bradley
• [Asterisk-Users] Re: DUNDi .... Halfway and CLUSTERING   Watkins, Bradley
• [Asterisk-Users] Annoying Asterisk Realtime Limitation   Watkins, Bradley
• [Asterisk-Users] VoiceMailMain(@context) Problem with Option 5 (Advanced)   Watkins, Bradley
• [Asterisk-Users] RE: VoiceMailMain(@context) Problem with Opt ion 5 (Advanced)   Watkins, Bradley
• [Asterisk-Users] License for asterisk-addons?   Watkins, Bradley
• [Asterisk-Users] Voicemail limit?   Watkins, Bradley
• [Asterisk-Users] Voicemail limit?   Watkins, Bradley
• [Asterisk-Users] Dell 2850 w/TDM2400?   Watkins, Bradley
• [Asterisk-Users] Question about advanced IVR   Devon Watkins
• [Asterisk-Users] EICON Diva 4BRI   David Waugh
• [Asterisk-Users] Numbered Voicemails when you still delete them.   David Waugh
• [Asterisk-Users] Numbered Voicemails even with delete option!   David Waugh
• [Asterisk-Users] Numbered Voicemails even with delete option!   David Waugh
• [Asterisk-Users] OT: Snom 320, displaying text on the screenfrom *   Franklin Webb
• [Asterisk-Users] Oh this is bad.... bindaddr and rtp traffic   Robert Webb
• [Asterisk-Users] SIP Jitter Buffer for 1.2.5   Robert Webb
• [Asterisk-Users] SIP Jitter Buffer for 1.2.5   Robert Webb
• [Asterisk-Users] openSUSE 10.0 and zaptel init script   Robert Webb
• [Asterisk-Users] List of transcoding combinations   Robert Webb
• [Asterisk-Users] MeetMe 'i' option not working correctly?   Jon Webster
• [Asterisk-Users] Marketing Materials   Jon Webster
• [Asterisk-Users] tsu-600   Dave Weis
• [Asterisk-Users] Replicate 486 Sip Response Code   Jon Weisman
• [Asterisk-Users] Replicate 486 Sip Response Code   Jon Weisman
• [Asterisk-Users] Program Buttons on Cisco 79xx Phones   Ron Wellsted
• [Asterisk-Users] Cisco 7960 SIP - Displaying Time   Ron Wellsted
• [Asterisk-Users] Asterisk Users Mailing List Traffic   Ron Wellsted
• [Asterisk-Users] Double Call Progress tones   Ron Wellsted
• [Asterisk-Users] Double Call Progress tones   Ron Wellsted
• [Asterisk-Users] Problems with wcte11xp module   Ron Wellsted
• [Asterisk-Users] Problem with cdr_odbc   Ron Wellsted
• [Asterisk-Users] Multi node call center   Michael Welter
• [Asterisk-Users] Polycom 501 power over ethernet   Michael Welter
• [Asterisk-Users] Polycom 501 power over ethernet   Michael Welter
• [Asterisk-Users] Analog Desktop Phone   Michael Welter
• [Asterisk-Users] Testing IAX links   Michael Welter
• [Asterisk-Users] Polycom IP501 Buddy List   Michael Welter
• [Asterisk-Users] SIP - Problem with audio clipping   Michael Welter
• [Asterisk-Users] "Not Found" in archive   Michael Welter
• [Asterisk-Users] Re: How is Teliax ?   Michael Welter
• [Asterisk-Users] PRI issues   Michael Welter
• [Asterisk-Users] multiple auto attendants   Michael Welter
• [Asterisk-Users] Re: How is Teliax ?   Michael Welter
• [Asterisk-Users] Music on hold and conferencing on OS X   Dustin Wenz
• [Asterisk-Users] Asterisk transfer conflict   BJ Weschke
• [Asterisk-Users] G729 and Meetme   BJ Weschke
• [Asterisk-Users] voip-info.... again   BJ Weschke
• [Asterisk-Users] Web-ex type solution for use with asterisk   BJ Weschke
• [Asterisk-Users] WiFi phones and WDS (Wireless Distribution System)   BJ Weschke
• [Asterisk-Users] Re: Asterisk-Users Digest, Vol 20, Issue 153   BJ Weschke
• [Asterisk-Users] How to create [new_context] in extensions.conf?   BJ Weschke
• [Asterisk-Users] RE: MeetMe freezes machine with Junghanns   BJ Weschke
• [Asterisk-Users] Re: Fw: anybody has SIP realtime working ?   BJ Weschke
• [Asterisk-Users] [OT] Polycom provisioning   BJ Weschke
• [Asterisk-Users] Re: Fw: anybody has SIP realtime working ?   BJ Weschke
• [Asterisk-Users] Re: Fw: anybody has SIP realtime working ?   BJ Weschke
• [Asterisk-Users] Polycom 501's for sale   BJ Weschke
• [Asterisk-Users] Page about 70 users crash my Asterisk   BJ Weschke
• [Asterisk-Users] Problem with MeetMe Conference!!!   BJ Weschke
• [Asterisk-Users] Page about 70 users crash my Asterisk   BJ Weschke
• [Asterisk-Users] Re: Fw: anybody has SIP realtime working ?   BJ Weschke
• [Asterisk-Users] Maximum Queue Name Length   BJ Weschke
• [Asterisk-Users] Copying SIP Subscriptions   BJ Weschke
• [Asterisk-Users] after-queues   BJ Weschke
• [Asterisk-Users] Polycoms and hints   BJ Weschke
• [Asterisk-Users] Agents on DND still receiving calls...   BJ Weschke
• [Asterisk-Users] Any new Voice Recognition devs?   BJ Weschke
• [Asterisk-Users] Confused on Agents and Queues   BJ Weschke
• [Asterisk-Users] Asterisk spanDSP / Faxing problem   Thys de Wet
• [Asterisk-Users] G.726 codec - can we select bandwidth?   Whisker, Peter
• [Asterisk-Users] ADPCM - vs - G.726   Whisker, Peter
• [Asterisk-Users] ADPCM - vs - G.726   Whisker, Peter
• [Asterisk-Users] cisco 7912 ringlist.dat file format   Zachary C. Whitley
• [Asterisk-Users] cisco 7912 ringlist.dat file format   Zachary C. Whitley
• [Asterisk-Users] AMP and ABE   Derek Whitten
• [Asterisk-Users] Asterisk Users Mailing List Traffic   Derek Whitten
• [Asterisk-Users] callerid= in zapata.conf   Derek Whitten
• [Asterisk-Users] Attachment Blocking Notification   Derek Whitten
• [Asterisk-Users] VERY IMPORTANT(TREAT WITH URGENCY)   Derek Whitten
• [Asterisk-Users] type of incoming lines   Derek Whitten
• [Asterisk-Users] Asterisk and "Commercial Unix"   Derek Whitten
• [Asterisk-Users] Asterisk and "Commercial Unix"   Derek Whitten
• [Asterisk-Users] Problems Configuring Cisco 12SP+   Derek Whitten
• [Asterisk-Users] Local Channel   Darren Wiebe
• [Asterisk-Users] Grabbing the billsec and duration after a hangup.   Darren Wiebe
• [Asterisk-Users] asterisk billing   Darren Wiebe
• [Asterisk-Users] Asterisk billing from CDR database   Darren Wiebe
• [Asterisk-Users] Asterisk billing from CDR database   Darren Wiebe
• [Asterisk-Users] Re: Asterisk at large   Eric "ManxPower" Wieling
• [Asterisk-Users] what version s this??   Eric "ManxPower" Wieling
• [Asterisk-Users] Same CID on multiple users(friends9 in SIP.conf   Eric "ManxPower" Wieling
• [Asterisk-Users] ignore a DID?   Eric "ManxPower" Wieling
• [Asterisk-Users] my zap channel not ringing   Eric "ManxPower" Wieling
• [Asterisk-Users] priorityjumping=no   Eric "ManxPower" Wieling
• [Asterisk-Users] echo cancellation   Eric "ManxPower" Wieling
• [Asterisk-Users] Fake Ring Tone/Compile Addon   Eric "ManxPower" Wieling
• [Asterisk-Users] Sync Source: Internally clocked   Eric "ManxPower" Wieling
• [Asterisk-Users] Asterisk Users Mailing List Traffic   Eric "ManxPower" Wieling
• [Asterisk-Users] Getting True ANI not Caller ID   Eric "ManxPower" Wieling
• [Asterisk-Users] Changing codec.   Eric "ManxPower" Wieling
• [Asterisk-Users] Re: How to nice agi scripts?   Eric "ManxPower" Wieling
• [Asterisk-Users] Call terminated after 60 seconds   Eric "ManxPower" Wieling
• [Asterisk-Users] Extension a?   Eric "ManxPower" Wieling
• SV: [Asterisk-Users] re: Sound issues on SIP-SIP calls   Eric "ManxPower" Wieling
• [Asterisk-Users] RE: IAX Incoming/Outgoing   Eric "ManxPower" Wieling
• [Asterisk-Users] 3Com Phones   Eric "ManxPower" Wieling
• [Asterisk-Users] Polycom IP 301 is slow   Eric "ManxPower" Wieling
• [Asterisk-Users] Authorization by ip   Eric "ManxPower" Wieling
• [Asterisk-Users] Dialogic d/4 PCI   Eric "ManxPower" Wieling
• [Asterisk-Users] Voicemail limit?   Eric "ManxPower" Wieling
• [Asterisk-Users] Routing SIP calls via URI   Eric "ManxPower" Wieling
• SV: [Asterisk-Users] IAX - only one way traffic   Eric "ManxPower" Wieling
• SV: [Asterisk-Users] IAX - only one way traffic   Eric "ManxPower" Wieling
• [Asterisk-Users] Re: How is Teliax ?   Eric "ManxPower" Wieling
• [Asterisk-Users] Nortel BCM and Asterisk as SIP Extension   Andrew Willerding
• [Asterisk-Users] Xorcom TS-1 T1 installs?   Mark Willis
• [Asterisk-Users] best pre paid for astreisk?   Walter Willis
• [Asterisk-Users] Re: [asterisk-dev] Is there a way to define an outbound proxy in sip.conf ?   Terry Wilson
• [Asterisk-Users] Best budget IP phone at the moment?   WipeOut
• [Asterisk-Users] IAXY codec support and questions..   WipeOut
• [Asterisk-Users] IAXY codec support and questions..   WipeOut
• [Asterisk-Users] Dial command   Ronald Wiplinger
• [Asterisk-Users] Dial command   Ronald Wiplinger
• [Asterisk-Users] Re: ON DEMAND call Recording   Ronald Wiplinger
• [Asterisk-Users] Audio problem   Ronald Wiplinger
• [Asterisk-Users] How to check if a phone / line is used?   Ronald Wiplinger
• [Asterisk-Users] Anyone using the GSMgateway from CyberTelecom ?   Wireless
• [Asterisk-Users] Sipura SPA-3000 vs Linksys SPA3000   Wireless
• [Asterisk-Users] RE: Delay in ringing   Wireless
• [Asterisk-Users] IPv6   Hans Witvliet
• [Asterisk-Users] gsm picocells   Hans Witvliet
• [Asterisk-Users] aah 2.7 / BRI   Hans Witvliet
• [Asterisk-Users] OT - Cisco IP Phone and PC in diferent VLANs (with 802.1x)   Wojtek
• [Asterisk-Users] Polycom 4000 results?   Wojtek
• [Asterisk-Users] Voip-Info   Tobias Wolf
• [Asterisk-Users] Re: digium.com redesign   Tobias Wolf
• [Asterisk-Users] voip-info.... again   Tobias Wolf
• [Asterisk-Users] Authorization by ip   Scott Wolfe
• [Asterisk-Users] 160 analogue phones..   Conrad Wood
• [Asterisk-Users] Problem with NAT!!!   Conrad Wood
• [Asterisk-Users] 160 analogue phones..   Conrad Wood
• [Asterisk-Users] Realtime SIP   Simon Woodhead
• [Asterisk-Users] Echo Cancelation on TE110P   Darren Wright
• [Asterisk-Users] NEWS: SIP Firmware Available for Cisco 7970   Darren Wright
• [Asterisk-Users] NEWS: SIP Firmware Available for Cisco 7970   Darren Wright
• [Asterisk-Users] NEWS: SIP Firmware Available for Cisco 7970   Darren Wright
• [Asterisk-Users] HW Echo Cancellers   Darren Wright
• [Asterisk-Users] More Voicemail prompts   Darren Wright
• [Asterisk-Users] 7940/60 SIP Call Park Button   Darren Wright
• [Asterisk-Users] An FXO version of IAXy?   Darren Wright
• [Asterisk-Users] Re: G729 and Meetme   Wai Wu
• [Asterisk-Users] Re: G729 and Meetme   Wai Wu
• [Asterisk-Users] Call recording   Wai Wu
• [Asterisk-Users] DTFM or FSK   Wai Wu
• [Asterisk-Users] DTFM or FSK   Wai Wu
• [Asterisk-Users] Clustering   Wai Wu
• [Asterisk-Users] Clustering   Wai Wu
• [Asterisk-Users] Clustering   Wai Wu
• [Asterisk-Users] Need help implementing call center features ofAsterisk   Wai Wu
• [Asterisk-Users] Need help implementing call center featuresofAsterisk   Wai Wu
• [Asterisk-Users] Clustering   Wai Wu
• [Asterisk-Users] Clustering "NEW THREAD", Almost Working   Wai Wu
• [Asterisk-Users] channel bridging   Wai Wu
• [Asterisk-Users] Clustering "NEW THREAD", Almost Working   Wai Wu
• [Asterisk-Users] External transfer   Wai Wu
• [Asterisk-Users] Re: regexten   Wai Wu
• [Asterisk-Users] How to enable talking in chanspy while spying?   Wai Wu
• [Asterisk-Users] Can this box handle 8 T1s (PSTN) with Asterisk?   Wai Wu
• [Asterisk-Users] Can this box handle 8 T1s (PSTN) with Asterisk?   Wai Wu
• [Asterisk-Users] Can this box handle 8 T1s (PSTN) with Asterisk?   Wai Wu
• [Asterisk-Users] Can this box handle 8 T1s (PSTN) with Asterisk?   Wai Wu
• [Asterisk-Users] polycom queue bug   Wai Wu
• [Asterisk-Users] transfer calls via Manager Api   Wai Wu
• [Asterisk-Users] How to create [new_context] in extensions.conf?   Wai Wu
• [Asterisk-Users] Changing codec.   Wai Wu
• [Asterisk-Users] Changing codec.   Wai Wu
• [Asterisk-Users] Callid on T-1 trunk   Wai Wu
• [Asterisk-Users] Span monitoring   Wai Wu
• [Asterisk-Users] Span monitoring   Wai Wu
• [Asterisk-Users] Span monitoring   Wai Wu
• [Asterisk-Users] Questions on call recording and conference.   Wai Wu
• [Asterisk-Users] meetme option 'e'   Wai Wu
• [Asterisk-Users] Marketing Materials   Charles Wyble
• [Asterisk-Users] Sipura SPA-3000 and PSTN dtmf   Vahan Yerkanian
• [Asterisk-Users] Sipura SPA-3000 and PSTN dtmf   Vahan Yerkanian
• [Asterisk-Users] Re: sipura 841 mass provisioning   Vahan Yerkanian
• [Asterisk-Users] Info about F1000G   Vahan Yerkanian
• [Asterisk-Users] Sipura SPA-3000 and PSTN dtmf   Vahan Yerkanian
• [Asterisk-Users] Info about F1000G   Vahan Yerkanian
• [Asterisk-Users] Re: sipura 841 mass provisioning   Vahan Yerkanian
• [Asterisk-Users] Sipura 3000 DMTF   Vahan Yerkanian
• [Asterisk-Users] sipura spa2 + asterisk bug ?   Vahan Yerkanian
• [Asterisk-Users] sipura spa2 + asterisk bug ?   Vahan Yerkanian
• [Asterisk-Users] 7940/60 SIP 8.2   Shane Young
• [Asterisk-Users] Re: how to show called name on calling polycom display   Shane Young
• [Asterisk-Users] ADT/Brinks alarm dialing through Asterisk   Shane Young
• [Asterisk-Users] ADT/Brinks alarm dialing through Asterisk   Shane Young
• [Asterisk-Users] Alarmreciver   Shane Young
• [Asterisk-Users] help with asterisk installation   (pg) Zeeshan
• [Asterisk-Users] Display Name   Edward de Zeeuw
• [Asterisk-Users] Settings for Yuxin Phones...   Juergen K. Zick
• [Asterisk-Users] What hardware to use for ISDN in Romania   Juergen K. Zick
• [Asterisk-Users] What hardware to use for ISDN in Romania   Juergen K. Zick
• [Asterisk-Users] Development news :: T38 passthrough support   Zoa
• [Asterisk-Users] Corrupt CDR records in Asterisk 1.2.x   Jeroen Zwarts
• [Asterisk-Users] Working Asterisk with Austrian ISDN p2p   Wolfgang Zweimueller
• [Asterisk-Users] Asterisk, QSIG and Tenovis PBX?   Wolfgang Zweimueller
• [Asterisk-Users] Howto cut the first digit   Wolfgang Zweimueller
• [Asterisk-Users] Asterisk, QSIG and Tenovis PBX?   Wolfgang Zweimueller
• [Asterisk-Users] Lowering Server Load   Matt Riddell [NZ]
• [Asterisk-Users] Changing caller id on transfer   Matt Riddell [NZ]
• [Asterisk-Users] Polling Asterisk for Life   Matt Riddell [NZ]
• [Asterisk-Users] IAX Video and Meetme   Matt Riddell [NZ]
• [Asterisk-Users] Re: G729 and Meetme   Matt Riddell [NZ]
• [Asterisk-Users] test call quality   Matt Riddell [NZ]
• [Asterisk-Users] Meetme Timing Interface   Matt Riddell [NZ]
• [Asterisk-Users] Oh this is bad.... bindaddr and rtp traffic   Matt Riddell [NZ]
• [Asterisk-Users] Echo Cancelation on TE110P   Matt Riddell [NZ]
• [Asterisk-Users] MixMonitor Problems -- sssshh, don't be too loud   Matt Riddell [NZ]
• [Asterisk-Users] can i get the script   Matt Riddell [NZ]
• [Asterisk-Users] Oh this is bad.... bindaddr and rtp traffic   Matt Riddell [NZ]
• [Asterisk-Users] res_mysql.conf & DNS SRV lookup   Matt Riddell [NZ]
• [Asterisk-Users] sending text to display of sip phones   Matt Riddell [NZ]
• [Asterisk-Users] parking slot lights - testers wanted   Matt Riddell [NZ]
• [Asterisk-Users] What port mpg123 uses for MoH?   Matt Riddell [NZ]
• [Asterisk-Users] Re: PLEASE respond: how to get Asterisk to change coders on RTP handoff?? HELLO???   Matt Riddell [NZ]
• [Asterisk-Users] Professional Recordings   Matt Riddell [NZ]
• [Asterisk-Users] H.264   Matt Riddell [NZ]
• [Asterisk-Users] Numbered Voicemails when you still delete them.   Matt Riddell [NZ]
• [Asterisk-Users] french sounds in asterisk   Matt Riddell [NZ]
• [Asterisk-Users] asterisk having problem in playing sounds   aRUnaR
• [Asterisk-Users] Critical Problem with asterisk   aRUnaR
• [Asterisk-Users] Alarm on Unicall   acriollo
• [Asterisk-Users] Re: Alarm on Unicall   acriollo
• [Asterisk-Users] Re: Alarm on Unicall   acriollo
• [Asterisk-Users] Re: Alarm on Unicall   acriollo
• [Asterisk-Users] Re: Alarm on Unicall   acriollo
• [Asterisk-Users] Re: Alarm on Unicall   acriollo
• [Asterisk-Users] Preferred editor(s) dialplan coding?   adibar
• [Asterisk-Users] Anyone using the GSMgateway from CyberTelecom ?   adibar
• [Asterisk-Users] Anyone using the GSMgateway from CyberTelecom ?   adibar
• [Asterisk-Users] Anyone using the GSMgateway from CyberTelecom ?   adibar
• [Asterisk-Users] Countries supporting SMS on PSTN (ISDN)   adibar
• [Asterisk-Users] dipura 2002 auto dial or intercom   guido amendano
• [Asterisk-Users] Settings for Yuxin Phones...   andrew.stock at sheltervalley.ca
• [Asterisk-Users] problems in changing Festival's Default Voice in Asterisk   arun arora
• [Asterisk-Users] PSTN to Asterisk VOIP in Manila   asterisk-user at dualcall.com
• [Asterisk-Users]chan_zap.c:6570 handle_init_event error   asterisk183
• [Asterisk-Users]Hangup with error   asterisk183
• [Asterisk-Users] How hard to create Asterisk for Compact Flash?   asterisk at anime.net
• [Asterisk-Users] Zoom 5801 problems with *   asterisk at anime.net
• [Asterisk-Users] Asterisk with T1 card on laptop   asterisk at anime.net
• [Asterisk-Users] Sipura SPA-3000 and PSTN dtmf   asterisk at anime.net
• [Asterisk-Users] Zoom 5801 problems with *   asterisk at anime.net
• [Asterisk-Users] Zoom 5801 problems with * (Wellgate 3701a)   asterisk at anime.net
• [Asterisk-Users] Zoom 5801 problems with * (Wellgate 3701a)   asterisk at anime.net
• [Asterisk-Users] Zoom 5801 problems with * (Wellgate 3701a)   asterisk at anime.net
• [Asterisk-Users] Zoom 5801 problems with * (Wellgate 3701a)   asterisk at anime.net
• [Asterisk-Users] Sipura SPA-3000 and PSTN dtmf   asterisk at anime.net
• [Asterisk-Users] Re: sipura 841 mass provisioning   asterisk at anime.net
• [Asterisk-Users] Sipura SPA-3000 vs Linksys SPA3000   asterisk at anime.net
• [Asterisk-Users] CON-SNT-CP7970 resellers?   asterisk at anime.net
• [Asterisk-Users] Program Buttons on Cisco 79xx Phones   asterisk at anime.net
• [Asterisk-Users] Preferred editor(s) dialplan coding?   asterisk at anime.net
• [Asterisk-Users] Program Buttons on Cisco 79xx Phones   asterisk at anime.net
• [Asterisk-Users] Program Buttons on Cisco 79xx Phones   asterisk at anime.net
• [Asterisk-Users] NEWS: SIP Firmware Available for Cisco 7970   asterisk at anime.net
• [Asterisk-Users] How to receive faxes with SPA3000 and Asterisk setup   asterisk at anime.net
• [Asterisk-Users] sending text to display of sip phones   asterisk at anime.net
• [Asterisk-Users] overlap dialing with polycom?   asterisk at anime.net
• [Asterisk-Users] Analog Desktop Phone   asterisk at anime.net
• [Asterisk-Users] Re: Asterisk-Users Digest, Vol 20, Issue 91   asterisk at anime.net
• [Asterisk-Users] Echo Cancellation   asterisk at anime.net
• [Asterisk-Users] Feedback from VON expo! Info on * HA and Polycom phone!!   asterisk at anime.net
• [Asterisk-Users] Re: RE: Re: Cisco phones and Linksys SRW224P   asterisk at anime.net
• [Asterisk-Users] An FXO version of IAXy?   asterisk at anime.net
• [Asterisk-Users] polycom queue bug   asterisk at anime.net
• [Asterisk-Users] 7970 SIP Firmware; SIP 8.2 for 7940/7960   asterisk at anime.net
• [Asterisk-Users] Cisco 7970 SIP Image - hint lines   asterisk at anime.net
• [Asterisk-Users] Polycom IP 301 is slow   asterisk at anime.net
• [Asterisk-Users] RE: Snom 360 problems   asterisk at anime.net
• [Asterisk-Users] Disable timeout for answered queue calls?   asterisk at anime.net
• [Asterisk-Users] Snom 360 problems   asterisk at anime.net
• [Asterisk-Users] Disable timeout for answered queue calls?   asterisk at anime.net
• [Asterisk-Users] Disable timeout for answered queue calls?   asterisk at anime.net
• [Asterisk-Users] RE: Snom 360 problems   asterisk at anime.net
• [Asterisk-Users] queue caveats   asterisk at anime.net
• [Asterisk-Users] How is Teliax ?   asterisk at anime.net
• [Asterisk-Users] Disable polycom call waiting?   asterisk at anime.net
• [Asterisk-Users] Disable polycom call waiting?   asterisk at anime.net
• [Asterisk-Users] Re: How is Teliax ?   asterisk at anime.net
• [Asterisk-Users] Re: How is Teliax ?   asterisk at anime.net
• [Asterisk-Users] Re: How is Teliax ?   asterisk at anime.net
• [Asterisk-Users] about operator   asterisk at frameweb.it
• [Asterisk-Users] about operator   asterisk at frameweb.it
• [Asterisk-Users] misdn   asterisk at frameweb.it
• [Asterisk-Users] misdn   asterisk at frameweb.it
• [Asterisk-Users] misdn problem   asterisk at frameweb.it
• [Asterisk-Users] misdn problem   asterisk at frameweb.it
• [Asterisk-Users] AMP backup-restore problem   asterisk at frameweb.it
• [Asterisk-Users] Feedback from VON expo!Infoon*HAandPolycomphone!!   asterisk at gafana.com
• [Asterisk-Users] RE: Clustering   asterisk at gafana.com
• [Asterisk-Users] Help! Directing Inbound calls to different extensions   asterisk at totemspace.com
• [Asterisk-Users] ADT/Brinks alarm dialing through Asterisk   asterisk at totemspace.com
• [Asterisk-Users] MINNESOTA: TwinCities Asterisk Users Group - Saturday 03/11/2006   asterisk_help at iwishi.nu
• [Asterisk-Users] Re: [asterisk-biz] Professional Recordings   asterisk_help at iwishi.nu
• [Asterisk-Users] dial plan !!   vikram b
• [Asterisk-Users] Sync Source: Internally clocked   bails
• [Asterisk-Users] Sync Source: Internally clocked   bails
• [Fwd: Re: [Asterisk-Users] Sync Source: Internally clocked]   bails
• [Asterisk-Users] UK pri almost working   bails
• [Asterisk-Users] aah 2.7 / BRI   bails
• [Asterisk-Users] # (send immediately) and dialplan broken on PAP2?   barton-lists at gcmcomputers.com
• [Asterisk-Users] # (send immediately) and dialplan broken on PAP2?   barton-lists at gcmcomputers.com
• [Asterisk-Users] low call volume   billy
• [Asterisk-Users] Slow outgoing pstn calls   billy
• [Asterisk-Users] How to assign channels for asterisk   bixbite at tom.com
• [Asterisk-Users] How to assign channels for asterisk   bixbite at tom.com
• [Asterisk-Users] Stuck. Extenions.conf? Realtime? MySQL? Grrrrr!   brett at websmyths.com
• [Asterisk-Users] Native MOH - Convert mp3 to ulaw   brett at websmyths.com
• [Asterisk-Users] How to disable event_log?   brett at websmyths.com
• [Asterisk-Users] (no subject)   btb
• [Asterisk-Users] change voicemail folders   btb
• [Asterisk-Users] Asterisk Prepaid Card   leonimar cape
• [Asterisk-Users] ON DEMAND call Recording   leonimar cape
• [Asterisk-Users] asterisk@home and H323   leonimar cape
• [Asterisk-Users] asterisk@home and H323   leonimar cape
• [Asterisk-Users] Asterisk Prepaid Card   leonimar cape
• [Asterisk-Users] Adding entries on company directory   leonimar cape
• [Asterisk-Users] asterisk and iptables   leonimar cape
• [Asterisk-Users] Adding entries on company directory   leonimar cape
• [Asterisk-Users] One Extension - Two Calls?   casasterisk at valnet.com
• [Asterisk-Users] Re: One Extension - Two Calls?   casasterisk at valnet.com
• [Asterisk-Users] G729 codec   chan at isysnetsolution.com
• [Asterisk-Users] Panasonic KXTD 1232 6   charles
• [Asterisk-Users] Speeding up the dial of DTMF's in SIP channel   chip at innovates.com
• [Asterisk-Users] WiFi phones and WDS (Wireless Distribution System)   chip at innovates.com
• [Asterisk-Users] voicetronix and asterisk@home   sean cook
• [Asterisk-Users] really need help with outgoing calls..PSTN errors   John covici
• [Asterisk-Users] asterisk 1.2.5 cannot call a zap channel extension   John covici
• [Asterisk-Users] Failed installing zaptel   John covici
• [Asterisk-Users] meetme recording very loud   John covici
• [Asterisk-Users] how to connect 3 or more servers via IAX ?   Jean-Louis curty
• [Asterisk-Users] a2billing problem with call duration   d_pejic at inecco.net
• [Asterisk-Users] PLEASE HELP ,a2billing problem with call duration   d_pejic at inecco.net
• [Asterisk-Users] Question: "When i Diall a group"   didier
• [Asterisk-Users] Background timeout and Read questions   didier
• [Asterisk-Users] Want to record call and put into users voicemail   duane.pudenz at shastaindustries.com
• [Asterisk-Users] Asterisk 1.2.6, VMWare, & Playback/Background GSM prompts   dveith1 at comcast.net
• [kpj@junghanns.net: Re: [asterisk@frameweb.it: RE: [Asterisk-Users] Corrupt CDR records in Asterisk 1.2.x]]   eric
• [Asterisk-Users] Re: One Extension - Two Calls?   ewr at erols.com
• [Asterisk-Users] Polycom voice.gain.tx.analog.handset andasteriskecho   ewr at erols.com
• [Asterisk-Users] T1 > GSM or DECT ? Searching for a complete microcell solution   f6hqz-m at hamwlan.net
• [Asterisk-Users] TDM11B Hang up detection not working in France ?   f6hqz-m at hamwlan.net
• [Asterisk-Users] Voice problem   f6hqz-m at hamwlan.net
• [Asterisk-Users] Echo Cancellation   f6hqz-m at hamwlan.net
• [Asterisk-Users] Best budget IP phone at the moment?   f6hqz-m at hamwlan.net
• [Asterisk-Users] IAX Phone?   f6hqz-m at hamwlan.net
• [Asterisk-Users] Sipura 3000 DMTF   f6hqz-m at hamwlan.net
• [Asterisk-Users] TDM 2400 With 24 FXO   f6hqz-m at hamwlan.net
• [Asterisk-Users] Asterisk Users Mailing List Traffic   f6hqz-m at hamwlan.net
• [Asterisk-Users] RE : RE : [asterisk-dev] iax failure?   f6hqz-m at hamwlan.net
• [Asterisk-Users] Stability of Asterisk with 2 x TDM400P cards (6analogue lines)   f6hqz-m at hamwlan.net
• [Asterisk-Users] FXS channel banks   f6hqz-m at hamwlan.net
• [Asterisk-Users] FXS channel banks   f6hqz-m at hamwlan.net
• [Asterisk-Users] FXS channel banks   f6hqz-m at hamwlan.net
• [Asterisk-Users] Stability of Asterisk with 2 x TDM400P cards (6analogue lines)   f6hqz-m at hamwlan.net
• [Asterisk-Users] Echo cancellation   f6hqz-m at hamwlan.net
• [Asterisk-Users] Analog Desktop Phone   fischer at snom.de
• [Asterisk-Users] really need help with outgoing calls..PSTN errors   sdgesa gaeharth
• [Asterisk-Users] really need help with outgoing calls..PSTN errors   sdgesa gaeharth
• [Asterisk-Users] really need help with outgoing calls..PSTN errors   sdgesa gaeharth
• [Asterisk-Users] really need help with outgoing calls..PSTN errors   sdgesa gaeharth
• [Asterisk-Users] really need help with outgoing calls..PSTN errors   sdgesa gaeharth
• [Asterisk-Users] really need help with outgoing calls..PSTN errors   sdgesa gaeharth
• [Asterisk-Users] really need help with outgoing calls..PSTN errors   sdgesa gaeharth
• [Asterisk-Users] transfers/parked calls + polycom 501   sdgesa gaeharth
• [Asterisk-Users] echo problem + choppy sound   sdgesa gaeharth
• [Asterisk-Users] echo problem + choppy sound   sdgesa gaeharth
• [Asterisk-Users] echo problem + choppy sound   sdgesa gaeharth
• [Asterisk-Users] echo problem + choppy sound   sdgesa gaeharth
• [Asterisk-Users] Re: transfers/parked calls + polycom 501   sdgesa gaeharth
• [Asterisk-Users] Re: transfers/parked calls + polycom 501   sdgesa gaeharth
• [Asterisk-Users] Re: transfers/parked calls + polycom 501   sdgesa gaeharth
• [Asterisk-Users] Re: transfers/parked calls + polycom 501   sdgesa gaeharth
• [Asterisk-Users] Re: transfers/parked calls + polycom 501   sdgesa gaeharth
• [Asterisk-Users] asterisk and skype - asterisk newbie   adriano ghezzi
• [Asterisk-Users] Info about F1000G   wendell hamilton
• [Asterisk-Users] Bluetooth headset in handsfree mode with SJPhoneor X-lite   wendell hamilton
• [Asterisk-Users] Bluetooth headset in handsfree modewith SJPhoneor X-lite   wendell hamilton
• [Asterisk-Users] Bluetooth headset in handsfree modewith SJPhoneorX-lite   wendell hamilton
• [Asterisk-Users] Bluetooth headsetin handsfree modewith SJPhoneorX-lite   wendell hamilton
• [Asterisk-Users] Any new Voice Recognition devs?   wendell hamilton
• [Asterisk-Users] how to dial to hardphone by kiax   zhang hao
• [Asterisk-Users] TDM04B sound volume   jonny hashem
• [Asterisk-Users] how to get separate CDR for inbound and outbound legs of a call   dorn hetzel
• [Asterisk-Users] Redirect a sip outbound requests to a sip proxy   hgaillac-sip at yahoo.fr
• [Asterisk-Users] Outbound Proxy Support   hgaillac-sip at yahoo.fr
• [Asterisk-Users] Re: [asterisk-dev] Is there a way to define an outbound proxy in sip.conf ?   hgaillac-sip at yahoo.fr
• [Asterisk-Users] MWI, SER and asterisk   hgaillac-sip at yahoo.fr
• [Asterisk-Users] app_queue and ARA   hgaillac-sip at yahoo.fr
• [Asterisk-Users] TIMEOUT(s)   hgaillac-sip at yahoo.fr
• [Asterisk-Users] Setting up announcement on reply to 4xx 5xx 6xx messages   hgaillac-sip at yahoo.fr
• [Asterisk-Users] Asterisk Compatible Server Architecture   hugolivude
• [Asterisk-Users] problems with emailing voicemail   hugolivude
• [Asterisk-Users] Re: problems with emailing voicemail   hugolivude
• [Asterisk-Users] Problem w/ Dial Command on Zap channel   hugolivude
• [Asterisk-Users] Re: problems with emailing voicemail   hugolivude
• [Asterisk-Users] How to make caller groups ???   hugolivude
• [Asterisk-Users] How to make caller groups ???   hugolivude
• [Asterisk-Users] Re: problems with emailing voicemail   hugolivude
• [Asterisk-Users] How to make groups of extensions ???   hugolivude
• [Asterisk-Users] Re: problems with emailing voicemail   hugolivude
• [Asterisk-Users] Re: problems with emailing voicemail   hugolivude
• [Asterisk-Users] Re: problems with emailing voicemail   hugolivude
• [Asterisk-Users] USING MMS STREAM FOR MOH   listas iPfone
• RES: [Asterisk-Users] pap2 Dial plan   indsat
• [Asterisk-Users] can i get the script   pali ismail
• [Asterisk-Users] mysql asterik   pali ismail
• [Asterisk-Users] Seperate music on hold for SIP extensions   james.texter at cox.net
• [Asterisk-Users] Re: Seperate music on hold for SIP extensions   james.texter at cox.net
• [Asterisk-Users] Analog POTS line -> Rhino FXO Channel Bank -> No Hangup   james.texter at cox.net
• [Asterisk-Users] D4 AMI - No Caller ID   james.texter at cox.net
• [Asterisk-Users] D4 AMI - No Caller ID   james.texter at cox.net
• [Asterisk-Users] Analog POTS line -> Rhino FXO Channel Bank -> No Hangup   james.texter at cox.net
• [Asterisk-Users] Asterisk with TOPEX GSM Gateway   jddr at nexus-sistemas.com
• [Asterisk-Users] Re: One Extension - Two Calls?   jeff
• [Asterisk-Users] Comedian Mail Add-ons?   jeff
• [Asterisk-Users] User Extension Custom Voicemail   jglucky at blueware.net
• [Asterisk-Users] Echo and static when dialing Asterisk   jglucky at blueware.net
• [Asterisk-Users] FreePBX & AAH   jglucky at blueware.net
• [Asterisk-Users] Asterisk with Vonage ATA question follow-up   jglucky at blueware.net
• [Asterisk-Users] Agent login   nelson jimenez
• [Asterisk-Users] 7970 8.x firmware speeddials   john
• [Asterisk-Users] 7970 8.x firmware speeddials   john
• [Asterisk-Users] spa3000 asterisk fxo gateway   john at cusick.ws
• [Asterisk-Users] Upgrading AAH   john at cusick.ws
• [Asterisk-Users] Sip Realtime Configs Samples with MySQL   jr.richardson at cox.net
• [Asterisk-Users] WiFi phones and WDS (Wireless Distribution System)   jason justman
• [Asterisk-Users] Cisco 7970   jason justman
• [Asterisk-Users] Re: Cisco 7970   jason justman
• [Asterisk-Users] Dell 2850 w/TDM2400?   jason justman
• [Asterisk-Users] Development news :: T38 passthrough   amer karim
• [Asterisk-Users] Disable timeout for answered queue calls?   amer karim
• [Asterisk-Users] Which g729 codec to download for a P4?   kaze0010 at umn.edu
• [Asterisk-Users] How to enable talking in chanspy while spying?   atik khan
• [Asterisk-Users] Get no busy signal on my analog line   kibeki at gmx.net
• [Asterisk-Users] pickup last ringing phone   erkan kolemen
• [Asterisk-Users] pickup problem   erkan kolemen
• [Asterisk-Users] Asterisk large scale, help needed   venkat kumar
• [Asterisk-Users] Understanding queue timeouts + possible bug found   lenz
• [Asterisk-Users] music on hold without mpg123   lenz
• [Asterisk-Users] Call Wrap-Up   lenz
• [Asterisk-Users] Tearing my hair out with Queues   lenz
• [Asterisk-Users] queue caveats   lenz
• [Asterisk-Users] Agent in multiple queues?   lenz
• [Asterisk-Users] Monitoring question   lenz
• [Asterisk-Users] Softphone accepting URL   lenz
• [Asterisk-Users] Newbie config help? Wellgate 3701a   kevin ling
• [Asterisk-Users] What SW/HW phones support sendtext feature (tryingto send speech recognition results back to user)?   kevin ling
• [Asterisk-Users] aah 2.7 / BRI   kevin ling
• [Asterisk-Users] wellgate 38XX FX & FXS voip gateways with outgoingcall files   kevin ling
• [Asterisk-Users] welltech Wellgate 3804 in SIP mode   kevin ling
• [Asterisk-Users] welltech Wellgate 3804 in SIP mode   kevin ling
• [Asterisk-Users] Re: FreePBX & AAH   kevin ling
• [Asterisk-Users] Invitation   varadhan m
• [Asterisk-Users] PBX-VPN-SIP-Asterisk trouble   artifex maximus
• [Asterisk-Users] PBX-VPN-SIP-Asterisk trouble   artifex maximus
• [Asterisk-Users] Dumb question (hang up detection/Zapata.conf)   artifex maximus
• [Asterisk-Users] welltech Wellgate 3804 in SIP mode   artifex maximus
• [Asterisk-Users] (no subject)   Hector medina
• [Asterisk-Users] Dimensions of TDM 24 card   eugenio mendez
• [Asterisk-Users] Problem with NAT!!!   serge messa
• [Asterisk-Users] RE: Asterisk-Users Digest, Vol 20, Issue 20   serge messa
• [Asterisk-Users] to configure asterisk to work with the nathelper module of openser   serge messa
• [Asterisk-Users] french sounds in asterisk   serge messa
• [Asterisk-Users] Problem with MeetMe Conference!!!   serge messa
• [Asterisk-Users] No voice heard in festivalassociated with asterisk!!!   serge messa
• [Asterisk-Users] decrease the speed of reading text!!!   serge messa
• [Asterisk-Users] Asterisk at large   mkumar at mantragroup.com
• [Asterisk-Users] Re: Asterisk at large   mkumar at mantragroup.com
• [Asterisk-Users] Asterisk Fax Question   mkumar at mantragroup.com
• [Asterisk-Users] Asterisk Fax Question   mkumar at mantragroup.com
• [Asterisk-Users] Extension 's' in Realtime   mkumar at mantragroup.com
• [Asterisk-Users] Asterisk RealTime Question, Please help   mkumar at mantragroup.com
• [Asterisk-Users] Asterisk RealTime Question, Please help   mkumar at mantragroup.com
• [Asterisk-Users] Asterisk RealTime Question, Please help   mkumar at mantragroup.com
• [Asterisk-Users] Asterisk RealTime Question, Please help   mkumar at mantragroup.com
• [Asterisk-Users] Asterisk RealTime Question, Please help   mkumar at mantragroup.com
• [Asterisk-Users] How hard to create Asterisk for Compact Flash?   mustardman29
• [Asterisk-Users] Re: TDM400P digium card   mustardman29
• [Asterisk-Users] Echo Cancelation on TE110P   mustardman29
• [Asterisk-Users] Echo Cancellation   mustardman29
• [Asterisk-Users] Echo Cancellation   mustardman29
• [Asterisk-Users] choppy recorded sounds in asterisk   mustardman29
• [Asterisk-Users] Aastra Questions   mustardman29
• [Asterisk-Users] Fw: anybody has SIP realtime working ?   mustardman29
• [Asterisk-Users] Re: Fw: anybody has SIP realtime working ?   mustardman29
• [Asterisk-Users] Re: Fw: anybody has SIP realtime working ?   mustardman29
• [Asterisk-Users] Re: Subscription state after reload (New subject)   mustardman29
• [Asterisk-Users] Aastra 9331i phones   mustardman29
• [Asterisk-Users] Re: Fw: anybody has SIP realtime working ?   mustardman29
• [Asterisk-Users] Re: Fw: anybody has SIP realtime working ?   mustardman29
• [Asterisk-Users] Re: Fw: anybody has SIP realtime working ?   mustardman29
• [Asterisk-Users] Voicemail to Email   mustardman29
• [Asterisk-Users] Re: Fw: anybody has SIP realtime working ?   mustardman29
• [Asterisk-Users] Receptionist Phones   mustardman29
• [Asterisk-Users] Asterisk with Vonage   mustardman29
• [Asterisk-Users] Receptionist Phones   mustardman29
• [Asterisk-Users] Receptionist Phones   mustardman29
• [Asterisk-Users] Asterisk with Vonage   mustardman29
• [Asterisk-Users] Building Asterisk embedded device   mustardman29
• [Asterisk-Users] problems with MOH   nik600
• [Asterisk-Users] monitor outgoing calls in queue / campaings   nik600
• [Asterisk-Users] login/logout agents in a specific queue   nik600
• [Asterisk-Users] broken pipe, restart asterisk   nik600
• [Asterisk-Users] monitor/statistic web interface for cdr   nik600
• [Asterisk-Users] difference between records in CDR and real duration of call   nik600
• [Asterisk-Users] queue and service period   nik600
• [Asterisk-Users] difference between records in CDR and real duration of call   nik600
• [Asterisk-Users] difference between records in CDR and realduration of call   nik600
• [Asterisk-Users] asterisk crash too much?   nik600
• [Asterisk-Users] GUI Web interface   nik600
• [Asterisk-Users] open source queue analyzer   nik600
• [Asterisk-Users] open source queue analyzer   nik600
• [Asterisk-Users] transfer calls via Manager Api   nik600
• [Asterisk-Users] asterisk as a fax server   nik600
• [Asterisk-Users] transfer calls via Manager Api   nik600
• [Asterisk-Users] kernel recompilation on a asterisk server   nik600
• [Asterisk-Users] transfer calls via Manager Api   nik600
• [Asterisk-Users] kernel recompilation on a asterisk server   nik600
• [Asterisk-Users] Problem with cdr_odbc   onespell at yandex.ru
• [Asterisk-Users] Testing asterisk faxing functionality   patryk
• [Asterisk-Users] Testing asterisk faxing functionality   patryk
• [Asterisk-Users] Injecting faxes directly to queue?   patryk
• [Asterisk-Users] Polycom 501   pdhales at optusnet.com.au
• [Asterisk-Users] Re: Asterisk Question   pdhales at optusnet.com.au
• [Asterisk-Users] Re: Asterisk Question   pdhales at optusnet.com.au
• [Asterisk-Users] incoming calls dropout on PRI over TE110p   pdhales at optusnet.com.au
• [Asterisk-Users] incoming calls dropout on PRI over TE110p   pdhales at optusnet.com.au
• [Asterisk-Users] Polycom 501 power over ethernet   pdhales at optusnet.com.au
• [Asterisk-Users] Polycom 501 power over ethernet   pdhales at optusnet.com.au
• [Asterisk-Users] Polycom 501 power over ethernet   pdhales at optusnet.com.au
• [Asterisk-Users] IVR woes   pdhales at optusnet.com.au
• [Asterisk-Users] IVR woes   pdhales at optusnet.com.au
• [Asterisk-Users] MultiBRI in Australia - found one - maybe   pdhales at optusnet.com.au
• [Asterisk-Users] Need help implementing call center features ofAsterisk   pdhales at optusnet.com.au
• [Asterisk-Users] MultiBRI in Australia - found one - maybe   pdhales at optusnet.com.au
• [Asterisk-Users] gsm picocells   pdhales at optusnet.com.au
• [Asterisk-Users] Asterisk Users Mailing List Traffic   pdhales at optusnet.com.au
• [Asterisk-Users] How to make extension groups ???   pdhales at optusnet.com.au
• [Asterisk-Users] Realtime / SIP Peers etc   pdhales at optusnet.com.au
• [Asterisk-Users] Native MOH - Convert mp3 to ulaw   pdhales at optusnet.com.au
• [Asterisk-Users] connecting Avaya Partnet with asterisk , TE205P   pdhales at optusnet.com.au
• [Asterisk-Users] Stability of Asterisk with 2 x TDM400P cards (6 analogue lines)   pdhales at optusnet.com.au
• [Asterisk-Users] G729 codec problems   pdhales at optusnet.com.au
• [Asterisk-Users] 3Com Phones   pdhales at optusnet.com.au
• [Asterisk-Users] GSM/DECT handsets (was gsm picocells)   pdhales at optusnet.com.au
• [Asterisk-Users] Hopefully a Simple Question?   pdhales at optusnet.com.au
• [Asterisk-Users] GSM/DECT handsets (was gsm picocells)   pdhales at optusnet.com.au
• [Asterisk-Users] Polycom IP 301 is slow   pdhales at optusnet.com.au
• [Asterisk-Users] Snom 360 - Multiple Server BLF Indications   pdhales at optusnet.com.au
• [Asterisk-Users] Snom 360 - Multiple Server BLF Indications   pdhales at optusnet.com.au
• [Asterisk-Users] Snom 360 - Multiple Server BLF Indications   pdhales at optusnet.com.au
• [Asterisk-Users] Polycom IP 301 is slow   pdhales at optusnet.com.au
• [Asterisk-Users] GSM/DECT handsets (was gsm picocells)   pdhales at optusnet.com.au
• [Asterisk-Users] Receptionist Phones   pdhales at optusnet.com.au
• [Asterisk-Users] Receptionist Phones   pdhales at optusnet.com.au
• [Asterisk-Users] Receptionist Phones   pdhales at optusnet.com.au
• [Asterisk-Users] Problems with wcte11xp module   pdhales at optusnet.com.au
• [Asterisk-Users] Config TE110P and TDM400 with 2 FXS modules   pdhales at optusnet.com.au
• [Asterisk-Users] Problems with installing a TE110P on a Dell Poweredge 850 running Fedora Core 4   phil.dawson at marnock.com
• [Asterisk-Users] Asterisk spanDSP / Faxing problem   picciuX
• [Asterisk-Users] RE: IAX Incoming/Outgoing   picciuX
• [Asterisk-Users] canreinvite=yes   ram
• [Asterisk-Users] canreinvite=yes   ram
• [Asterisk-Users] Prepaid / postpaid solution   ram
• [Asterisk-Users] Re: a2billing without IVR   ram
• [Asterisk-Users] Oneway voice   ram
• [Asterisk-Users] Not getting mails from Mar 2   ram
• [Asterisk-Users] Prepaid / postpaid solution   ram
• [Asterisk-Users] Building a small Office EPABX with VoIP GW with Asterisk   ram
• [Asterisk-Users] anyway to a2billing without IVR   ram
• [Asterisk-Users] SER & Asterisk with DID incoming and out going   ram
• [Asterisk-Users] SER & Asterisk with DID incoming and out going   ram
• [Asterisk-Users] g729 and latency measures   ram
• [Asterisk-Users] Aterisk with Realtime   ram
• [Asterisk-Users] Aterisk with Realtime   ram
• [Asterisk-Users] Aterisk with Realtime   ram
• [Asterisk-Users] Aterisk with Realtime   ram
• [Asterisk-Users] Asterisk and gateway   ram
• [Asterisk-Users] Asterisk and gateway   ram
• [Asterisk-Users] Asterisk and gateway   ram
• [Asterisk-Users] inbound routing help   ram
• [Asterisk-Users] Incoming Call keeps ringing when the second call arrives   deniz rende
• [Asterisk-Users] fax receive using TDM400P, with Tzafir, Anton, Cosmin, Colin...   yrving rivas
• [Asterisk-Users] help required configuring card   rnacharya
• [Asterisk-Users] Fw: help required configuring card   rnacharya
• [Asterisk-Users] G.729 codec licencing   rnacharya
• [Asterisk-Users] connecting Avaya Partnet with asterisk , TE205P   rnacharya
• [Asterisk-Users] Is extension.conf documentation wrong?   rossi.tek
• [Asterisk-Users] embedded hardware for Asterisk?   sam
• [Asterisk-Users] Building Asterisk embedded device   sam
• [Asterisk-Users] Building Asterisk embedded device   sam
• [Asterisk-Users] Building Asterisk embedded device   sam
• [Asterisk-Users] Building Asterisk embedded device   sam
• [Asterisk-Users] asterisk@home V's Asterisk   scott
• [Asterisk-Users] Hardware Levels   scott
• [Asterisk-Users] Implementing MOH while trunks gets connected...   sdcharly at gmail.com
• [Asterisk-Users] TDM11B Hang up detection not working in France ?   sdcharly at gmail.com
• [Asterisk-Users] Anyone using the GSMgateway from CyberTelecom ?   sdcharly at gmail.com
• [Asterisk-Users] Anyone using the GSMgateway from CyberTelecom ?   sdcharly at gmail.com
• [Asterisk-Users] Anyone using the GSMgateway from CyberTelecom ?   sdcharly at gmail.com
• [Asterisk-Users] G729 codec   sdcharly at gmail.com
• [Asterisk-Users] Calls from PSTN , answering, When transfered get Hungup 'Zap/1-1'   sdcharly at gmail.com
• [Asterisk-Users] asterisk hang   stevanus
• [Asterisk-Users] asterisk hang   stevanus
• [Asterisk-Users] Avoiding initial deadlock on iax?   stevanus
• [Asterisk-Users] A room full of Cisco 7960s behind NAT   steve
• [Asterisk-Users] re: asterisk and iptables   steve
• [Asterisk-Users] Re: script to Restart Zaptel   steve
• [Asterisk-Users] I have my asterisk machine behind a Linux, Nat ...   steve
• [Asterisk-Users] re: Sound issues on SIP-SIP calls   steve
• SV: [Asterisk-Users] re: Sound issues on SIP-SIP calls   steve
• [Asterisk-Users] How to disable event_log?   steve
• [Asterisk-Users] How hard to create Asterisk for Compact Flash?   stoffell
• [Asterisk-Users] hands-on experience with soft videophone   stoffell
• [Asterisk-Users] Re: DUNDi .... Halfway and CLUSTERING   stoffell
• [Asterisk-Users] ISDN Protocol Unknom Error with Junghanns OctoBRI   stoffell
• [Asterisk-Users] MeetMe freezes machine with Junghanns QuadBRI cards   stoffell
• [Asterisk-Users] MeetMe freezes machine with Junghanns QuadBRI cards   stoffell
• [Asterisk-Users] 3Com Phones   stoffell
• [Asterisk-Users] zaphfc on an 'actual' asterisk?   stoffell
• [Asterisk-Users] Buddy watch?   rivy strauss
• [Asterisk-Users] HOWTO volume per (7960) phone   support at sjobeck.com
• [Asterisk-Users] HOWTO initialize new kernel & kernel source without reboot   support at sjobeck.com
• [Asterisk-Users] RE: Asterisk-Users Digest, Vol 18, Issue 147   support at sjobeck.com
• [Asterisk-Users] OT: any one in Stockholm for quick piece of advice   support at sjobeck.com
• [Asterisk-Users] OT: reset LinkSys 941 to factory defaults & howto config' via TFTP   support at sjobeck.com
• [Asterisk-Users] What is asterisk   bmw suzuki
• [Asterisk-Users] 3Com Phones   tom
• [Asterisk-Users] Jittery meetme conference using Linksys 942 phones   tracinet
• [Asterisk-Users] SPA-941/2 Monitoring   tracinet
• [Asterisk-Users] Two asterisks on one machine   vivek at staff.ownmail.com
• [Asterisk-Users] Two asterisks on one machine   vivek at staff.ownmail.com
• [Asterisk-Users] Asterisk programmer needed   voip3 at nibble.net
• [Asterisk-Users] Asterisk programmer needed   voip3 at nibble.net
• [Asterisk-Users] Two dissimilar data only T1 lines possible in Asterisk?   voip3 at nibble.net
• [Asterisk-Users] Call Simulator   voipman
• [Asterisk-Users] Voicemail to Email   voipman
• [Asterisk-Users] type of incoming lines   mike webb
• [Asterisk-Users] getting your own phone number   mike webb
• [Asterisk-Users] pots -> asterisk -> tsu-600   mike webb
• [Asterisk-Users] tsu-600   mike webb
• [Asterisk-Users] tsu-600   mike webb
• [Asterisk-Users] asterisk turn key solution   mike webb
• [Asterisk-Users] Asterisk code help   santosh y
• [Asterisk-Users] my zap channel not ringing   yusuf
• [Asterisk-Users] Asterisk + SE Linux   yusuf
• [Asterisk-Users] ZT_CHANCONFIG failed on channel 2: , Guidance requested   yusuf
• [Asterisk-Users] How to transmit Video   yusuf
• [Asterisk-Users] carry forward uniqueid   yusuf
• [Asterisk-Users] Creating a voip network... use asterisk?   yusuf
• [Asterisk-Users] SIP Realtime Users   yusuf
• [Asterisk-Users] List of transcoding combinations   yusuf
• [Asterisk-Users] chan_h323 problem   yusuf
• [Asterisk-Users] my zap channel not ringing & source from   yusuf at ecntelecoms.com
• [Asterisk-Users] wake up calls   yusuf at ecntelecoms.com
• [Asterisk-Users] Two PBX   yusuf at ecntelecoms.com
• [Asterisk-Users] [offtopic] Asterisk <-IP-> Siemens HiPath 4000   vinicius zanc
• [Asterisk-Users] Avaya IP Office 412   zgor
• [Asterisk-Users] Avaya IP Office 412   zgor