September 2005 Archives by date

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Starting: Thu Sep 1 00:31:10 MST 2005
Ending: Fri Sep 30 22:09:48 MST 2005
Messages: 3874

• [Asterisk-Users] Recommendation for 8 lines analog card in Australia   Kib Eki
• [Asterisk-Users] Asterisk run problem, was working, rebooted server, now nothing   c_dragon at bellsouth.net
• [Asterisk-Users] Asterisk@Home: How to changed AMP User Login andPassword   Chad Brown
• [Asterisk-Users] TE cards with ISDN BRI?   Aaron Picht
• [Asterisk-Users] TE cards with ISDN BRI?   Armin Schindler
• [Asterisk-Users] Re: equipment configuration help   asterisk groups
• [Asterisk-Users] /etc/init.d/asterisk barfing   Steve Hanselman
• [Asterisk-Users] How to require a keypress on answer?   Tony Mountifield
• [Asterisk-Users] /etc/init.d/asterisk barfing   Julian Lyndon-Smith
• [Asterisk-Users] Snom 360 and hints   Alessio Focardi
• [Asterisk-Users] Micronet 5050s FXO gateway and hookflash transfers.   John Melody
• [Asterisk-Users] Mulig_SPAM: More than one outgoing call   Tue Topholm
• [Asterisk-Users] TDM400P problems   Alex Ongena
• [Asterisk-Users] Re: equipment configuration help   asterisk groups
• [Asterisk-Users] Why ZAPATA inserting pause before last digit, during dialing? GRRRRR....   Goran Dj.
• [Asterisk-Users] How to execute StopPlayTones when a SIP phone is answered   Chris Coulthurst
• [Asterisk-Users] Re: Strange problem with Bristuff   Stefan Tichy
• [Asterisk-Users] Help setting up trunk on AAH   Zeeshan Zakaria
• [Asterisk-Users] Snom 360 and hints   BJ Weschke
• [Asterisk-Users] TDM400P problems   Andrew Kohlsmith
• [Asterisk-Users] Sipura 1001 Adapter with two lines using one RG11 jack   Zeeshan Zakaria
• [Asterisk-Users] What this little red cross mean in AAH   Zeeshan Zakaria
• [Asterisk-Users] Snom 360 and hints   Jeff Brownlee
• [Asterisk-Users] Polycom 301 second line registration,   Jeremy Melanson
• [Asterisk-Users] Mobilephone users get echo of them self when calling in to our asterisk server.   Arne Morten Johansen
• [Asterisk-Users] Snom 360 and hints   altus
• [Asterisk-Users] Grandstream GXP-2000 Poor sound Quality   Aaron W
• [Asterisk-Users] What this little red cross mean in AAH   Tim Litwiller
• [Asterisk-Users] Softphone vmail indicator and TDM400P woes   Rich Adamson
• [Asterisk-Users] /etc/init.d/asterisk barfing   Steve Hanselman
• [Asterisk-Users] Sipura 1001 Adapter with two lines using one RG11 jack   Chris Mason (Lists)
• [Asterisk-Users] TDM400P problems   Alex Ongena
• [Asterisk-Users] Snom 360 and hints   BJ Weschke
• [Asterisk-Users] Snom 360 and hints   Nils Ohlmeier
• [Asterisk-Users] Re: Strange problem with Bristuff   tonini.massimo at tiscali.it
• [Asterisk-Users] Recommendation for 8 lines analog card in Australia   Darren Younger
• [Asterisk-Users] oh323 or h323   Steve Ducat
• [Asterisk-Users] Queues.conf OPTIONALURL within the Queues cmd   Dan
• [Asterisk-Users] Snom 360 and hints   Jeff Brownlee
• [Asterisk-Users] Connecting Asterisk to a Toshiba Strata system   Araba, Michael
• [Asterisk-Users] Re: Polycom 301 second line registration   Noah Miller
• [Asterisk-Users] Re: equipment configuration help   Erick Perez
• [Asterisk-Users] Overhead Paging Systems...   kurth at ozonecomputer.com
• [Asterisk-Users] Snom 360 hold problem   Michael George
• [Asterisk-Users] HELP - Queue Transfer   Patrick Adair
• [Asterisk-Users] looking for Russia and Israel Dids   Mehdi chouikh
• [Asterisk-Users] oh323 or h323   Mehdi chouikh
• [Asterisk-Users] /etc/init.d/asterisk barfing   Rich Adamson
• [Asterisk-Users] Overhead Paging Systems...   Chris Mason (Lists)
• [Asterisk-Users] zapata nationalprefix-problem [Virus checked]   DRi at b-w-computer.de
• [Asterisk-Users] /etc/init.d/asterisk barfing   Julian Lyndon-Smith
• [Asterisk-Users] One way echo canceling?   Matt Fredrickson
• [Asterisk-Users] TDM04b and echo   Matt Fredrickson
• [Asterisk-Users] Oracle Realtime Driver and CDR Logger   Kevin P. Fleming
• [Asterisk-Users] TE cards with ISDN BRI?   Kevin P. Fleming
• [Asterisk-Users] Overhead Paging Systems...   Chris Coulthurst
• [Asterisk-Users] oh323 or h323   Leandro Tenorio
• [Asterisk-Users] RE: Asterisk with Meridian1 option11 in the UK   Chands
• [Asterisk-Users] Enable anonymous SIP incoming calls   Gustavo García
• [Asterisk-Users] detecting extensions in use   Eric "Skippy" Hope
• [Asterisk-Users] Astaro SIP Proxy   Samy Antoun
• [Asterisk-Users] Loop error when compiling CVS version of 1.2-Beta   Geoff Karl
• [Asterisk-Users] Outbound Authentication   Graham Kiff
• [Asterisk-Users] Grandstream GXP-2000 Poor sound Quality   Jesus Mogollon
• [Asterisk-Users] Overhead Paging Systems...   William Boehlke
• [Asterisk-Users] Oracle Realtime Driver and CDR Logger   Kanuri, Seshu (Company IT)
• [Asterisk-Users] One way echo canceling?   Doug Lytle
• [Asterisk-Users] ztcfg problem   Giordano Grandis
• [Asterisk-Users] dialing extension, which context is searched   Iqbal
• [Asterisk-Users] Overhead Paging Systems...[More Info]   kurth at ozonecomputer.com
• [Asterisk-Users] Sangoma card problem with EWSD Exchange   Mike M
• [Asterisk-Users] *66 with Sipura devices.   Matt
• [Asterisk-Users] E-Tech ADWV01   Rene Kluwen
• [Asterisk-Users] How to resolve SMS/WAP/MMS/VoIP gateways on a shoestring?   Henry Junior
• [Asterisk-Users] Problem with include   Il Neofita
• [Asterisk-Users] Buying DIDs   Joshua Abbott
• [Asterisk-Users] Asterisk@Home: How to changed AMP User Login andPassword   Chands
• [Asterisk-Users] Grandstream GXP-2000 Poor sound Quality   Aaron W
• [Asterisk-Users] TOS bit and DSCP   Ronald Hartmann
• [Asterisk-Users] Asterisk run problem, was working, rebooted server, now nothing   c_dragon at bellsouth.net
• [Asterisk-Users] sip jitter buffer in 1.2?   Damon Estep
• [Asterisk-Users] Loop error when compiling CVS version of 1.2-Beta   Julian Lyndon-Smith
• [Asterisk-Users] Fax trouble with HP 3330mfp (again)   Remco Barende
• [Asterisk-Users] TDM04b and echo   Ariel Batista
• [Asterisk-Users] Problem with include   Kevin P. Fleming
• [Asterisk-Users] Grandstream GXP-2000 Poor sound Quality   Peter Svensson
• [Asterisk-Users] Speed Questiosn   Joshua Abbott
• [Asterisk-Users] dialparties.agi is returning no extensions to dial   Robert G. Ristroph
• [Asterisk-Users] Two devices behind nat   Chris Wilson
• [Asterisk-Users] Low handset microphone volume with Sipura SPA-841   Juan Jose Comellas
• [Asterisk-Users] TOS bit and DSCP   Rosario Pingaro
• [Asterisk-Users] Overhead Paging Systems...   Paul
• [Asterisk-Users] ipvolution t1 cards   Trey Scarborough
• [Asterisk-Users] sip jitter buffer in 1.2?   Matt
• [Asterisk-Users] How to speed-up INCOMING-RINGING-ENDED detection on X101P/zapata?   Goran Dj.
• [Asterisk-Users] ztcfg problem   Tzafrir Cohen
• [Asterisk-Users] /etc/init.d/asterisk barfing   Tzafrir Cohen
• [Asterisk-Users] /etc/init.d/asterisk barfing   Tzafrir Cohen
• [Asterisk-Users] One way echo canceling?   Matt Fredrickson
• [Asterisk-Users] ipvolution t1 cards   Andrew Kohlsmith
• [Asterisk-Users] AGI nor System working after a dial - Should it work?   Patrick Tracanelli
• [Asterisk-Users] IAX2 how to disable VAD ?   Julien
• [Asterisk-Users] ipvolution t1 cards   Wiley Siler
• [Asterisk-Users] Hardware dimensioning issues   William Boehlke
• [Asterisk-Users] Contact Directory on Polycom IP-501 phones   Jesse Keating
• [Asterisk-Users] Realtime IAX   Dana Olson
• [Asterisk-Users] Speed Questiosn   Carlos Alperin
• [Asterisk-Users] Contact Directory on Polycom IP-501 phones   Jeremy Melanson
• [Asterisk-Users] Contact Directory on Polycom IP-501 phones   Jesse Keating
• [Asterisk-Users] Hardware dimensioning issues   Juan Luis Moyano
• [Asterisk-Users] Best costs effective solution...   housi mueller
• [Asterisk-Users] Skipping problems on outgoing calls (using uLaw with an internal * server through Voxee)   Robert Geller
• [Asterisk-Users] /etc/init.d/asterisk barfing   Rich Adamson
• [Asterisk-Users] Automon filenames   Anton Krall
• [Asterisk-Users] Contact Directory on Polycom IP-501 phones   Jesse Keating
• [Asterisk-Users] Best costs effective solution...   Anton Krall
• [Asterisk-Users] How to shorten ringing stop detection onX101Pclone?   Goran Dj.
• [Asterisk-Users] ztcfg problem   Michel Koenen
• [Asterisk-Users] One way echo canceling?   Doug Lytle
• [Asterisk-Users] Question about Asterisk connections   Tielin Xu
• [Asterisk-Users] Problem with include   Il Neofita
• [Asterisk-Users] Problem with include   Kevin P. Fleming
• [Asterisk-Users] Recommendation for 8 lines analog card inAustralia   Paul Hales
• [Asterisk-Users] OT: SCALE 4x -- Call For Papers   Ilan Rabinovitch
• [Asterisk-Users] How to set CLIR when using call files ?   Michel Koenen
• [Asterisk-Users] Skipping problems on outgoing calls (using uLaw with an internal * server through Voxee)   Robert Geller
• [Asterisk-Users] Hardware dimensioning issues   Juan Luis Moyano
• [Asterisk-Users] Help on second dial   David Lung
• [Asterisk-Users] RE: Hardware dimensioning issues To: <juanmoyano@southecon.com.ar>   M O
• [Asterisk-Users] TE110p and E1   Stephen
• [Asterisk-Users] Hardware dimensioning issues   Carlos Alperin
• [Asterisk-Users] extra ring after answer on sip calls   Damon Estep
• [Asterisk-Users] RE: Hardware dimensioning issues To: <juanmoyano@southecon.com.ar>   Waldo Rubinstein
• [Asterisk-Users] Hardware dimensioning issues   Waldo Rubinstein
• [Asterisk-Users] Polycom Reboot Script   Anton Krall
• [Asterisk-Users] GXP-2000 presence   Anton Krall
• [Asterisk-Users] Fwd: asterisk in Taiwan   Lance Grover
• [Asterisk-Users] TXFAX() status   Moises Silva
• [Asterisk-Users] Grandstream GXP-2000 Poor sound Quality   Joe McConnaughey
• [Asterisk-Users] Re: equipment configuration help   Erick Perez
• [Asterisk-Users] Need Local HELP!!!   Steve Gladden
• [Asterisk-Users] TE406P seg fault on Stable   Matt Florell
• [Asterisk-Users] Recommendation for 8 lines analog card in Australia   C F
• [Asterisk-Users] Any one in Toronto / Canada can help me!   Talkvoip Telecom Canada
• [Asterisk-Users] Re: Asterisk-Users Digest, Vol 14, Issue 1   Nguyen Trung Tin
• [Asterisk-Users] How to require a keypress on answer?   C F
• [Asterisk-Users] Connecting Asterisk to a Toshiba Strata system   Leo Ann Boon
• [Asterisk-Users] Snom 360 and hints   altus
• [Asterisk-Users] SIPGetHeader   Samy Antoun
• [Asterisk-Users] Re: Strange problem with Bristuff   Thomas Petersen
• [Asterisk-Users] Snom 360 problem   altus
• [Asterisk-Users] Snom 360 and hints   Remco Barende
• [Asterisk-Users] Setting wcte11xp card to use IRQ   Lee Archer
• [Asterisk-Users] Fax problem, missing/compressed lines   Johan van Tongeren
• [Asterisk-Users] Italy FastWeb problem: ISDN line crashes every time cisco router turns off   Giorgio Incantalupo
• [Asterisk-Users] Italy FastWeb problem: ISDN line crashes every time cisco router turns off   Sergio Chersovani
• [Asterisk-Users] Snom 360 and hints   altus
• [Asterisk-Users] Re: How to require a keypress on answer?   Tony Mountifield
• [Asterisk-Users] Re: equipment configuration help   asterisk groups
• [Asterisk-Users] Snom 360 and hints   Nils Ohlmeier
• [Asterisk-Users] Snom 360 problem   Nils Ohlmeier
• [Asterisk-Users] Snom 360 hold problem   Nils Ohlmeier
• [Asterisk-Users] ASTCC-adding more than one trunk to one route   Faramarz Amidafshar
• [Asterisk-Users] Looking for better "Follow Me"   Hauke Zuehl
• [Asterisk-Users] Call drops   Sahil Gupta
• [Asterisk-Users] oh323 or h323   Steve Ducat
• [Asterisk-Users] Looking for better "Follow Me"   Michiel van Baak
• [Asterisk-Users] Looking for better "Follow Me"   Hauke Zuehl
• [Asterisk-Users] Looking for better "Follow Me"   Michiel van Baak
• [Asterisk-Users] Looking for better "Follow Me"   Sergio Chersovani
• [Asterisk-Users] Looking for better "Follow Me"   John covici
• [Asterisk-Users] Snom 360 and hints   Alessio Focardi
• [Asterisk-Users] Realtime IAX   Chris A. Icide
• [Asterisk-Users] chan_capi hfcpci mISDN linux 2.6.12 not working   Konrads Smelkovs
• [Asterisk-Users] chan_capi hfcpci mISDN linux 2.6.12 not working   Armin Schindler
• [Asterisk-Users] sip SUBSCRIPTION bug in 1.0.9   Philipp von Klitzing
• [Asterisk-Users] Snom 360 and hints   BJ Weschke
• [Asterisk-Users] Comparing mISDN and CAPI for ISDN BRI   Dias Badekas
• [Asterisk-Users] ISDN BRI setting up   Dias Badekas
• [Asterisk-Users] Any one in Toronto / Canada can help me!   Michael Stahl
• [Asterisk-Users] Any one in Toronto / Canada can help me!   Michael Stahl
• [Asterisk-Users] How to shorten ringing stop detection onX101Pclone?   f6hqz-m at hamwlan.net
• [Asterisk-Users] Re: Strange problem with Bristuff   Stefan Tichy
• [Asterisk-Users] oh323 or h323   Leandro Tenorio
• [Asterisk-Users] TDM400P problems   Alex Ongena
• [Asterisk-Users] oh323 or h323   Steve Ducat
• [Asterisk-Users] DTMF and "breaking through" voice prompts   Sherwood McGowan
• [Asterisk-Users] VoipBuster with astersisk?   Rudolf Ladyzhenskii
• [Asterisk-Users] Why is that: Sep 2 08:25:03 NOTICE[1403]: -- Registration for '1096377@192.168.0.100' timed out, trying again   Zeeshan Zakaria
• [Asterisk-Users] DTMF and "breaking through" voice prompts   Damon Estep
• [Asterisk-Users] DTMF and "breaking through" voice prompts   Andreas Sikkema
• [Asterisk-Users] Re: Oracle Realtime Driver and CDR Logger   Dias Badekas
• [Asterisk-Users] DTMF and "breaking through" voice prompts   Sherwood McGowan
• [Asterisk-Users] Looking for better "Follow Me"   Matthew Boehm
• [Asterisk-Users] monitoring VM via speaker and grabbing connection   Martin
• [Asterisk-Users] Semi-OT: An idea for New Orleans temporary communications infrastructure   Jeremy Melanson
• [Asterisk-Users] Semi-OT: An idea for New Orleans temporarycommunications infrastructure   Dean Collins
• [Asterisk-Users] FW: defunct email kill list   Dean Collins
• [Asterisk-Users] Zapata help needed howto configure nationalprefix for a single card   DRi at b-w-computer.de
• [Asterisk-Users] Semi-OT: An idea for New Orleans temporary communications infrastructure   Damon Estep
• [Asterisk-Users] Semi-OT: An idea for New Orleanstemporarycommunications infrastructure   Damon Estep
• [Asterisk-Users] G711u sound quality decrease with upgrade from 1.0.7 to CVS-HEAD?   Matt
• [Asterisk-Users] Re: Oracle Realtime Driver and CDR Logger   Kevin P. Fleming
• [Asterisk-Users] Re: G711u sound quality decrease with upgrade from 1.0.7 to CVS-HEAD?   Matt
• [Asterisk-Users] dialparties.agi is returning no extensions to dial   Robert G. Ristroph
• [Asterisk-Users] Comparing mISDN and CAPI for ISDN BRI   Armin Schindler
• [Asterisk-Users] Re: G711u sound quality decrease with upgrade from 1.0.7 to CVS-HEAD?   Matt
• [Asterisk-Users] how to execute something after Dial() ?   Simone Cittadini
• [Asterisk-Users] TDM400P problems   Martin
• [Asterisk-Users] Realtime IAX   Dana Olson
• [Asterisk-Users] Unable to create RTP session   Kamran Ahmad
• [Asterisk-Users] TDM400 w/ FXS S110M pinout on RJ11 connector?   Werner Johansson
• [Asterisk-Users] Fax problem, missing/compressed lines   Lee Howard
• [Asterisk-Users] Re: how to execute something after Dial() ?   Tony Mountifield
• [Asterisk-Users] TDM400P problems   Jonathan k. Creasy
• [Asterisk-Users] No application 'AgentsLogin'   ChB
• [Asterisk-Users] TDM400P problems   Andrew Kohlsmith
• [Asterisk-Users] oh323 or h323   Leandro Tenorio
• [Asterisk-Users] No application 'AgentsLogin'   Kevin P. Fleming
• [Asterisk-Users] AG-468 4xFXS - my personal review   Joseph
• [Asterisk-Users] Asterisk for Voicemail Server   Sean Cook
• [Asterisk-Users] Recommendations for a low cost GSM phone   Colin Anderson
• [Asterisk-Users] TDM400P problems   Martin
• [Asterisk-Users] CallerID Num and Name setting to Asterisk.. Problem   Doug Lytle
• [Asterisk-Users] TDM400P problems   Andrew Kohlsmith
• [Asterisk-Users] Grandstream GXP-2000 Poor sound Quality   Alvaro Parres
• [Asterisk-Users] TDM400P problems   Martin
• [Asterisk-Users] Linux-HA Heartbeat2 and Asterisk   Geoff Karl
• [Asterisk-Users] TDM400P problems   Martin
• [Asterisk-Users] PRI Identity Crisis   Damon Estep
• [Asterisk-Users] CallerID and CDR   Sherwood McGowan
• [Asterisk-Users] No application 'AgentsLogin'   ChB
• [Asterisk-Users] chan_sccp new release   Sergio Chersovani
• [Asterisk-Users] How to locate Toll Free Ownership   calvis
• [Asterisk-Users] Realtime IAX   Chris A. Icide
• [Asterisk-Users] Receptionist   Alain Major/Simard
• [Asterisk-Users] How to locate Toll Free Ownership   gw at adcomcorp.com
• [Asterisk-Users] FW: defunct email kill list   Kevin Walsh
• [Asterisk-Users] Semi-OT: An idea for New Orleanstemporarycommunications infrastructure   Jason p
• [Asterisk-Users] Receptionist   Joseph
• [Asterisk-Users] FW: defunct email kill list   razza
• [Asterisk-Users] chan_oh323.conf (inAccess version)   nor amie aris
• [Asterisk-Users] Re: equipment configuration help   Erick Perez
• [Asterisk-Users] Receptionist   Colin Anderson
• [Asterisk-Users] Notification of new voicemail by various methods   Robert G. Ristroph
• [Asterisk-Users] Call Return   Paradise Dove
• [Asterisk-Users] Notification of new voicemail by various met hods   Colin Anderson
• [Asterisk-Users] Distortion/crackling/skipping problems on outgoing calls -- please help!!!   Robert Geller
• [Asterisk-Users] Notification of new voicemail by various methods   John covici
• [Asterisk-Users] PRI Identity Crisis   Matthew Boehm
• [Asterisk-Users] Receptionist   Gavin Spurgeon
• [Asterisk-Users] oh323 or h323   Steve Ducat
• [Asterisk-Users] Receptionist   Colin Anderson
• [Asterisk-Users] Dlink dph-140s/ACT P104SLD   D. J. Williams
• [Asterisk-Users] Asterisk for Voicemail Server   Mark Phillips
• [Asterisk-Users] Notification of new voicemail by various methods   Johann
• [Asterisk-Users] Call Return   Soner Tari
• [Asterisk-Users] Notification of new voicemail by various methods   John covici
• [Asterisk-Users] SER+ASTERISK voicemail   harry gaillac
• [Asterisk-Users] Notification of new voicemail by various methods   Johann
• [Asterisk-Users] Asterisk and Eyebeam   Chris Mason (Lists)
• [Asterisk-Users] Receptionist   canuck15
• [Asterisk-Users] TE406P seg fault on Stable   steve at daviesfam.org
• [Asterisk-Users] TE406P seg fault on Stable   Matt Florell
• [Asterisk-Users] TE406P seg fault on Stable   Kevin P. Fleming
• [Asterisk-Users] Receptionist   Paul Hales
• [Asterisk-Users] Need * Setup Help   Joshua Abbott
• [Asterisk-Users] TE406P seg fault on Stable   Matt Florell
• [Asterisk-Users] Web-voicemail doesn't play files nor display default pictures   Claudio Canseco
• [Asterisk-Users] X101P ringing too long !   Goran Dj.
• [Asterisk-Users] random beeps in MeetMe   Ben Brown
• [Asterisk-Users] CVS-HEAD Inband Ringing?   Matt
• [Asterisk-Users] STUN on PAP2-NA 2.0.12(LS)   Tomas Florian
• [Asterisk-Users] random beeps in MeetMe   Tomas Florian
• [Asterisk-Users] Sipura 3000 setup   Ronald Wiplinger
• [Asterisk-Users] Sipura 3000 setup   D. J. Williams
• [Asterisk-Users] Polycom Reboot Script   Greg Boehnlein
• [Asterisk-Users] Realtime Voicemail & mailcmd=   niles at atheos.net
• [Asterisk-Users] Linux-HA Heartbeat2 and Asterisk   Matt Riddell
• [Asterisk-Users] AG-468 4xFXS - my personal review   Matt Riddell
• [Asterisk-Users] AG-468 4xFXS - my personal review   Joseph
• [Asterisk-Users] IVR Prompts   Joshua Abbott
• [Asterisk-Users] IVR Prompts   William
• [Asterisk-Users] AG-468 4xFXS - my personal review   Steve Underwood
• [Asterisk-Users] chan_capi [0.4.0|-cm-0.5.4] and Asterisk 1.2.0-beta1 - early B3 not early enough sometimes   Gary Hawkins
• [Asterisk-Users] chan_capi [0.4.0|-cm-0.5.4] and Asterisk 1.2.0-beta1 - early B3 not early enough sometimes   Armin Schindler
• [Asterisk-Users] Multiple ASTCC Cards Configuration   chawki hammoud
• [Asterisk-Users] IVR Prompts   Mark Phillips
• [Asterisk-Users] Distortion/crackling/skipping problems on outgoing calls -- please help!!!   Rich Adamson
• [Asterisk-Users] How to tell reason for hangup or busy in SIP or IAX   Roger Schreiter
• [Asterisk-Users] Sipura 3000 setup   Rich Adamson
• [Asterisk-Users] Debug info from txfax - howto?   Roger Schreiter
• [Asterisk-Users] Current status on _outgoing_ Swedish/Dutch DTMF CLIP for TDM400 FXS interfaces?   Werner Johansson
• [Asterisk-Users] random beeps in MeetMe   Ben Brown
• [Asterisk-Users] newbie install problem. And I already searched everywhere!   Jeroen Baten
• [Asterisk-Users] equipment and network advice   Mag Gam
• [Asterisk-Users] stale nonce?   Rich Adamson
• [Asterisk-Users] DNS SRV and new Asterisk install   Chris Roberts
• [Asterisk-Users] Dialing internal extensions with AAH 1.5   Chris Shipman
• [Asterisk-Users] newbie install problem. And I already searchedeverywhere!   Werner Johansson
• [Asterisk-Users] Multiple ASTCC Cards Configuration   Insider KT
• [Asterisk-Users] Multiple ASTCC Cards Configuration   Darren Wiebe
• [Asterisk-Users] Distortion/crackling/skipping problems on outgoing calls -- please help!!!   Robert Geller
• [Asterisk-Users] Distortion/crackling/skipping problems on outgoing calls -- please help!!!   Brian Capouch
• [Asterisk-Users] newbie install problem. And I already searchedeverywhere!   Jeroen Baten
• [Asterisk-Users] newbie install problem. And I already searchedeverywhere!   Jeroen Baten
• [Asterisk-Users] Linux-HA Heartbeat2 and Asterisk   Geoff Karl
• [Asterisk-Users] *81, block CID, using ATA   Jim Sturtevant
• [Asterisk-Users] *81, block CID, using ATA   Jim Sturtevant
• [Asterisk-Users] unicall deploy   acriollo
• [Asterisk-Users] newbie install problem. And I already searchedeverywhere!   Dave Cotton
• [Asterisk-Users] chan_iax2.c:7672 iax2_poke_noanswer   Chris Mason (Lists)
• [Asterisk-Users] Distortion/crackling/skipping problems on outgoing calls -- please help!!!   Robert Geller
• [Asterisk-Users] Distortion/crackling/skipping problems on outgoing calls -- please help!!!   Brian Capouch
• [Asterisk-Users] I connected my quicknet phonejack to the wall phone outlet and .......   cmisip
• [Asterisk-Users] Distortion/crackling/skipping problems on outgoing calls -- please help!!!   Robert Geller
• [Asterisk-Users] Distortion/crackling/skipping problems on outgoing calls -- please help!!!   Brian Capouch
• [Asterisk-Users] Argentina - zapata.conf switchtype for Argentina   Leandro Rzezak
• [Asterisk-Users] How Separate a few channels from the others on a PRI   Derrick Stensrud
• [Asterisk-Users] Distortion/crackling/skipping problems on outgoing calls -- please help!!!   Rich Adamson
• [Asterisk-Users] How To Separate incoming channels from the others on a PRI   Derrick Stensrud
• [Asterisk-Users] newbie install problem. And I alreadysearchedeverywhere!   brett at websmyths.com
• [Asterisk-Users] Re: equipment configuration help   astgroups
• [Asterisk-Users] Polycom 301 second line registration,   andres
• [Asterisk-Users] Distortion/crackling/skipping problems on outgoing calls -- please help!!!   Rich Adamson
• [Asterisk-Users] Any suggestions for an IP phone?   andres
• [Asterisk-Users] MWI - message waiting indication   harry gaillac
• [Asterisk-Users] Mplayer as replacement to mgp123 in MP3Player cmd   cmisip
• [Asterisk-Users] Sipura spa841 problems   Anton Krall
• [Asterisk-Users] Distortion/crackling/skipping problems on outgoing calls -- please help!!!   Robert Geller
• [Asterisk-Users] Distortion/crackling/skipping problems on outgoing calls -- please help!!!   Robert Geller
• [Asterisk-Users] Argentina - zapata.conf switchtype for Argentina   Guillermo Freige
• [Asterisk-Users] unicall deploy   Guillermo Freige
• [Asterisk-Users] Distortion/crackling/skipping problems on outgoing calls -- please help!!!   Robert Geller
• [Asterisk-Users] Distortion/crackling/skipping problems on outgoing calls -- please help!!!   Robert Geller
• [Asterisk-Users] Asterisk Community Participant; Katrina Refugee   JR Richardson
• [Asterisk-Users] Asterisk Community Participant; Katrina Refugee   Don Fanning
• [Asterisk-Users] newbie install problem. And I alreadysearchedeverywhere!   Jeroen Baten
• [Asterisk-Users] newbie install problem. And I alreadysearchedeverywhere!   Jeroen Baten
• [Asterisk-Users] Asterisk Community Participant; Katrina Refugee   harry gaillac
• [Asterisk-Users] newbie install problem. And I alreadysearchedeverywhere!   Stijn Jonker
• [Asterisk-Users] IPSwichBoard designers wanted   Thorben Jensen
• [Asterisk-Users] FW: Asterisk@home - requesting help on oh323, ISDN BRI and iConnectHere DID   AbdelRahman Tarzi
• [Asterisk-Users] Asterisk Community Participant; Katrina Refugee   Skeeve Stevens
• [Asterisk-Users] Open G.729 / G.723.1 update, fixed memory leak   Daniel Pocock
• [Asterisk-Users] Asterisk Community Participant; Katrina Refugee   Senad J
• [Asterisk-Users] Asterisk Community Participant; Katrina Refugee   f6hqz-m at hamwlan.net
• [Asterisk-Users] dial rule / prefix with #   Ben Dinnerville
• [Asterisk-Users] HELP - How Do I Separate incoming channels from the others on a PRI   Derrick Stensrud
• [Asterisk-Users] Any hardphones with SIP API?   Josip Gracin
• [Asterisk-Users] donating VOIP gear to the relief efforts.   Glenn A. Thompson
• [Asterisk-Users] FW: Asterisk@home - requesting help on oh323, ISDN BRI and iConnectHere DID   Jason Becker
• [Asterisk-Users] Asterisk SMS via IAX2?   Henry Junior
• [Asterisk-Users] Asterisk Community Participant; Katrina Refugee   Rich Adamson
• [Asterisk-Users] HELP - How Do I Separate incoming channels from the others on a PRI   C F
• [Asterisk-Users] Setting wcte11xp card to use IRQ   C F
• [Asterisk-Users] Asterisk Community Participant; Katrina Refugee [OT]   Norbert Kamenicky
• [Asterisk-Users] HELP - How Do I Separate incoming channels from the others on a PRI   Adam Goryachev
• [Asterisk-Users] Distortion/crackling/skipping problems on outgoing calls -- please help!!!   Rich Adamson
• [Asterisk-Users] newbie install problem. And I alreadysearchedeverywhere!   Jeroen Baten
• [Asterisk-Users] unicall deploy   Denis Galvão - iSolve
• [Asterisk-Users] chan_sip.c:946 __sip_xmit   Bartosz Jozwiak
• [Asterisk-Users] Nokia 32 Terminal   andrutto
• [Asterisk-Users] Dlink dph-140s/ACT P104SLD   D. J. Williams
• [Asterisk-Users] newbie install problem. And I alreadysearchedeverywhere!   Stijn Jonker
• [Asterisk-Users] newbie install problem. And I alreadysearchedeverywhere!   Armin Schindler
• [Asterisk-Users] OT: Sipura SPA 200 Caller ID Problem   Hadar Pedhazur
• [Asterisk-Users] SIP, NAT and MySQL support (sipfriends)   Jarek Jarzebowski
• [Asterisk-Users] Option 1 in IVR menu   Adrian A
• [Asterisk-Users] sipura spc.exe ?   Rich Adamson
• [Asterisk-Users] Messagenet.it   FaberK
• [Asterisk-Users] Option 1 in IVR menu   Adam Goryachev
• FW: [Asterisk-Users] OH323 with Asterisk@home - seems incomplete   AbdelRahman Tarzi
• [Asterisk-Users] Asterisk and Eyebeam   Juraj Bednar
• [Asterisk-Users] Nokia 32 Terminal   AbdelRahman Tarzi
• [Asterisk-Users] Updated Chan Unistim?   Andres
• [Asterisk-Users] FW: Asterisk@home - requesting help on oh323, ISDN BRI and iConnectHere DID   AbdelRahman Tarzi
• [Asterisk-Users] Asterisk Real-Time Voicemail Configuration   Matt
• [Asterisk-Users] Re: ztdummy and Linux 2.6.13-rc7   Doug Lytle
• [Asterisk-Users] Re: Asterisk Real-Time Voicemail Configuration   Matt
• [Asterisk-Users] Distortion/crackling/skipping problems on outgoing calls -- please help!!!   Robert Geller
• [Asterisk-Users] Re: ztdummy and Linux 2.6.13-rc7   Rob Thomas
• [Asterisk-Users] Snom hints/lights   Paul Hales
• [Asterisk-Users] kernel panic   Michael Welter
• [Asterisk-Users] Re: ztdummy and Linux 2.6.13-rc7   Doug Lytle
• [Asterisk-Users] Unable to hear.   Jennifer Hales
• [Asterisk-Users] Asterisk Real-Time Voicemail Configuration   Matthew Boehm
• [Asterisk-Users] Re: Asterisk Real-Time Voicemail Configuration   Matthew Boehm
• [Asterisk-Users] Argentina - zapata.conf switchtype for Argentina   Carlos Alperin
• [Asterisk-Users] Distortion/crackling/skipping problems on outgoing calls -- please help!!!   Rich Adamson
• [Asterisk-Users] Automon filenames   Anton Krall
• [Asterisk-Users] sending fax   Chris Shipman
• [Asterisk-Users] A few questions before final proposal...   Kurth Bemis
• [Asterisk-Users] A few questions before final proposal...   Adam Goryachev
• [Asterisk-Users] A few questions before final proposal...   Kurth Bemis
• [Asterisk-Users] help on 2 X-Lite: call failed: 404 not found   lee tance
• [Asterisk-Users] hints and polycom IP 300 phones   Adam Goryachev
• [Asterisk-Users] Problem with Asterisk app command Read...   Leo Burd
• [Asterisk-Users] A few questions before final proposal...   Johan van Tongeren
• [Asterisk-Users] sending fax   Johan van Tongeren
• [Asterisk-Users] RE: RE: Nokia 32 Terminal   andrutto
• [Asterisk-Users] sending fax   Johan van Tongeren
• [Asterisk-Users] Problem with Asterisk app command Read...   Johan van Tongeren
• [Asterisk-Users] pridump   andrutto
• [Asterisk-Users] Asterisk and SCCP unofficial site   Stefan Gofferje
• [Asterisk-Users] Billing - Disable accounts when balance gets 0 value   itn at itn.com.br
• [Asterisk-Users] (no subject)   itn at itn.com.br
• [Asterisk-Users] newbie install problem. And I alreadysearchedeverywhere!   Jeroen Baten
• [Asterisk-Users] WG: Timeout when Dialing - HELP   Pascal Speck
• [Asterisk-Users] newbie install problem. And I alreadysearchedeverywhere!   Jeroen Baten
• [Asterisk-Users] Billing - Disable accounts when balance gets 0 value   Simone Cittadini
• [Asterisk-Users] ReInvite not working   Ishay
• [Asterisk-Users] Re: ztdummy and Linux 2.6.13-rc7   Tony Mountifield
• [Asterisk-Users] DTMF issue on IVR   larry lin
• [Asterisk-Users] No DID on ZAP   Darren Wright
• [Asterisk-Users] DTMF issue on IVR   maka
• [Asterisk-Users] hints and polycom IP 300 phones   harry gaillac
• Tr: [Asterisk-Users] MWI - message waiting indication   harry gaillac
• [Asterisk-Users] GotoIf sample...   ryan nalupa
• [Asterisk-Users] queue transfers always get EXITWITHKEY   lenz
• [Asterisk-Users] chan_capi hfcpci mISDN linux 2.6.12 not working   Konrads Smelkovs
• SV: [Asterisk-Users] sending fax   Arne Morten Johansen
• [Asterisk-Users] Re: ztdummy and Linux 2.6.13-rc7   Doug Lytle
• [Asterisk-Users] Re: ztdummy and Linux 2.6.13-rc7   Rob Thomas
• [Asterisk-Users] Nokia 32 Terminal   Sergio Chersovani
• [Asterisk-Users] kernel panic   Tzafrir Cohen
• [Asterisk-Users] chan_capi hfcpci mISDN linux 2.6.12 not working   Konrads Smelkovs
• [Asterisk-Users] compiling asterisk   Dante Renda
• [Asterisk-Users] asterisk@home and zaphfc dial out not working   bman at host.sk
• [Asterisk-Users] Asterisk clustering with SIP proxy?   Roy Sigurd Karlsbakk
• [Asterisk-Users] kernel panic   Elio Rojano
• [Asterisk-Users] asterisk CAPI dial-in issues   Konrads Smelkovs
• [Asterisk-Users] TDM Card FXO Question   Robert Webb
• [Asterisk-Users] Re: ztdummy and Linux 2.6.13-rc7   Tony Mountifield
• [Asterisk-Users] A good HW   Jan Buchal
• [Asterisk-Users] asterisk CAPI dial-in issues   Armin Schindler
• [Asterisk-Users] Re: Asterisk Real-Time Voicemail Configuration   Matt
• [Asterisk-Users] asterisk CAPI dial-in issues   Sergio Chersovani
• [Asterisk-Users] Re:initiate call with asterisk -> authentication error telnetting to Manager API   Eric
• [Asterisk-Users] Re: initiate call with asterisk -> authentication error telnetting to Manager API   Tony Mountifield
• [Asterisk-Users] Re: initiate call with asterisk -> authentication error telnetting to Manager API   Eric
• [Asterisk-Users] asterisk CAPI dial-in issues   Armin Schindler
• [Asterisk-Users] TDM Card FXO Question   Rich Adamson
• [Asterisk-Users] A few questions before final proposal...   Mark Phillips
• [Asterisk-Users] Re: Asterisk Real-Time Voicemail Configuration   Matthew Boehm
• [Asterisk-Users] asterisk CAPI dial-in issues   Konrads Smelkovs
• [Asterisk-Users] Re: Asterisk-Users Digest, Vol 14, Issue 22   Nguyen Trung Tin
• [Asterisk-Users] sending fax   Chris Shipman
• [Asterisk-Users] Re: Asterisk Real-Time Voicemail Configuration   Matt
• [Asterisk-Users] unicall deploy   acriollo
• [Asterisk-Users] GotoIf sample...   anderson at skyweb.co.ke
• [Asterisk-Users] ooh323c h323_convertAsteriskCapToH323Cap Don't know how to deal with mode 0x40 (slin)   Konrads Smelkovs
• [Asterisk-Users] No DID on ZAP   Alexander Lopez
• [Asterisk-Users] chan_unical-MFC/R2 CPU usage problem   Hadi Jadallah
• [Asterisk-Users] more accounts   FaberK
• [Asterisk-Users] User authentication and privileges   Mark Elkins
• [Asterisk-Users] Asterisk Follow ME   Vladyslav
• [Asterisk-Users] Re: initiate call with asterisk -> authentication error telnetting to Manager API   Tony Mountifield
• [Asterisk-Users] BT100 and BETA 1.0.7.11   Bartosz Jozwiak
• [Asterisk-Users] "Provisioned, Down, Active", but D-channel seems to be fine   steve at daviesfam.org
• [Asterisk-Users] putty and winscp   Dean Collins
• [Asterisk-Users] asterisk CAPI dial-in issues   Konrads Smelkovs
• [Asterisk-Users] "Provisioned, Down, Active", but D-channel seems to be fine   Kevin P. Fleming
• [Asterisk-Users] "Provisioned, Down, Active", but D-channel seems to be fine   steve at daviesfam.org
• [Asterisk-Users] "Provisioned, Down, Active", but D-channel seems to be fine   Kevin P. Fleming
• [Asterisk-Users] Argentina - zapata.conf switchtype for Argentina   Guillermo Freige
• [Asterisk-Users] asterisk CAPI dial-in issues   Armin Schindler
• [Asterisk-Users] unicall deploy   Guillermo Freige
• [Asterisk-Users] asterisk CAPI dial-in issues   Armin Schindler
• [Asterisk-Users] BT100 and BETA 1.0.7.11   Bartosz Jozwiak
• [Asterisk-Users] BT100 and BETA 1.0.7.11   Bartosz Jozwiak
• [Asterisk-Users] Asterisk won't listen on another port   Aisling
• [Asterisk-Users] Unexpected results with "While" and "EndWhile" applications   John Todd
• [Asterisk-Users] Re: equipment configuration help   Erick Perez
• [Asterisk-Users] res_features.so (Call Features Resource) not loading   Asterisk Sales
• [Asterisk-Users] unicall deploy   acriollo
• [Asterisk-Users] sending fax   Harald Klein
• [Asterisk-Users] sending fax   Il Neofita
• [Asterisk-Users] Asterisk architecture   housi mueller
• [Asterisk-Users] Cisco 7960 upgrades   Sascha Ferley
• [Asterisk-Users] Asterisk architecture   Il Neofita
• [Asterisk-Users] res_features.so (Call Features Resource) not loading   Michiel van Baak
• [Asterisk-Users] Assessing network quality   Chris Mason (Lists)
• [Asterisk-Users] Zaptel issue   Asterisk
• [Asterisk-Users] Re: Re: Nokia 32 Terminal   andrutto
• [Asterisk-Users] Cisco 7960 upgrades   Matthew Boehm
• [Asterisk-Users] Cisco 7960 upgrades   Harald Klein
• [Asterisk-Users] Asterisk as a GSM-Gateway? Possible or not??   Stefan-Michael. Guenther (in-put GbR)
• [Asterisk-Users] Zaptel issue   Andrew Kohlsmith
• [Asterisk-Users] Zaptel issue   steve at daviesfam.org
• [Asterisk-Users] Channalized T1 and PRI with Asterisk   Ben Brown
• [Asterisk-Users] No DID on ZAP   Matt Riddell
• [Asterisk-Users] Asterisk architecture   housi mueller
• [Asterisk-Users] Asterisk architecture   Il Neofita
• [Asterisk-Users] Assessing network quality   Matt Riddell
• [Asterisk-Users] Asterisk overheating on VIA Epia M Series motherboard   Angus Comber
• [Asterisk-Users] Assessing network quality   Andres
• [Asterisk-Users] Channalized T1 and PRI with Asterisk   Michael D Schelin
• [Asterisk-Users] Asterisk won't listen on another port   Andres
• [Asterisk-Users] Agentlogin transfer calls   Neil Bullock
• [Asterisk-Users] Channalized T1 and PRI with Asterisk   Ben Brown
• [Asterisk-Users] TDM11B pinout   Gary Smith
• [Asterisk-Users] Asterisk architecture   housi mueller
• [Asterisk-Users] TDM400 and Phone does not 'ring'   Gary Smith
• [Asterisk-Users] Channalized T1 and PRI with Asterisk   Michael D Schelin
• [Asterisk-Users] Assessing network quality   Rich Adamson
• [Asterisk-Users] Channalized T1 and PRI with Asterisk   Peter Svensson
• [Asterisk-Users] TDM11B pinout   Robert Webb
• [Asterisk-Users] Cisco 7960 upgrades   Sascha Ferley
• [Asterisk-Users] Channalized T1 and PRI with Asterisk   Kevin P. Fleming
• [Asterisk-Users] Re: Re: Nokia 32 Terminal   AbdelRahman Tarzi
• [Asterisk-Users] Re: Re: Nokia 32 Terminal   AbdelRahman Tarzi
• [Asterisk-Users] unicall and cvs head   Anton Krall
• [Asterisk-Users] RE: RE: Nokia 32 Terminal   AbdelRahman Tarzi
• [Asterisk-Users] USING TWO ACCOUNTS WITH BROADVOICE   Bartosz Wegrzyn - asterisk
• [Asterisk-Users] Asterisk overheating on VIA Epia M Series motherboard   C F
• [Asterisk-Users] Heartbeat with Broadvoice   Bartosz Wegrzyn - asterisk
• [Asterisk-Users] Asterisk overheating on VIA Epia M Series motherboard   The VoIP Connection
• [Asterisk-Users] Channalized T1 and PRI with Asterisk   Ben Brown
• [Asterisk-Users] Asterisk overheating on VIA Epia M Series motherboard   Paul
• [Asterisk-Users] Asterisk for Voicemail Server   Jonathan k. Creasy
• [Asterisk-Users] Asterisk overheating on VIA Epia M Series motherboard   Paul
• [Asterisk-Users] USING TWO ACCOUNTS WITH BROADVOICE   Rich Adamson
• [Asterisk-Users] Asterisk Community Participant; Katrina Refugee   Mike M
• [Asterisk-Users] Cisco 7960 upgrades   Matt Riddell
• [Asterisk-Users] Unexpected results with "While" and "EndWhile" applications   C F
• [Asterisk-Users] User authentication and privileges   C F
• [Asterisk-Users] GotoIf sample...   C F
• [Asterisk-Users] Channalized T1 and PRI with Asterisk   Ben Brown
• [Asterisk-Users] A good HW   Tomas Florian
• [Asterisk-Users] Unexpected results with "While" and "EndWhile" applications   John Todd
• [Asterisk-Users] atxfer featuremap   Sander
• [Asterisk-Users] Automon filenames   usman at user.iphonica.net
• [Asterisk-Users] A good HW   Jan Buchal
• [Asterisk-Users] chan_unical-MFC/R2 CPU usage problem   Tobias Wolf
• [Asterisk-Users] Wierd Problem   Gulzar Hussain
• [Asterisk-Users] Automon filenames   Anton Krall
• [Asterisk-Users] SIP Callgroups   Wolfgang Pichler
• [Asterisk-Users] Asterisk won't listen on another port   Umair Bari
• [Asterisk-Users] USING TWO ACCOUNTS WITH BROADVOICE   Umair Bari
• [Asterisk-Users] Going crazy with FAX :-(   Michele "O-Zone" Pinassi
• [Asterisk-Users] Channalized T1 and PRI with Asterisk   Peter Svensson
• [Asterisk-Users] TDM 400p   Johan van Tongeren
• [Asterisk-Users] Asterisk won't listen on another port   Aisling
• [Asterisk-Users] Assessing network quality   Chris Mason (Lists)
• [Asterisk-Users] TDM 400p   Andrew Kohlsmith
• [Asterisk-Users] Asterisk Cluster   Matt King
• [Asterisk-Users] Asterisk Cluster   BJ Weschke
• [Asterisk-Users] Asterisk Cluster   Patrick
• [Asterisk-Users] Asterisk overheating on VIA Epia M Seriesmotherboard   Angus Comber
• [Asterisk-Users] Re: Asterisk Cluster   Matt King
• [Asterisk-Users] Going crazy with FAX :-(   Steve Underwood
• [Asterisk-Users] Asterisk overheating on VIA Epia MSeriesmotherboard   razza
• [Asterisk-Users] Re: Asterisk Cluster   BJ Weschke
• [Asterisk-Users] Transfering calls   Christian Schoepplein
• [Asterisk-Users] Asterisk as SIP/H.323 Signalling Gateway   BJ Weschke
• [Asterisk-Users] Business telephones   Steve Langstaff
• [Asterisk-Users] Help evacuees from LA, MS, AL locate lived ones   Michael Graves
• [Asterisk-Users] GotoIf sample...   Terry Wilson
• [Asterisk-Users] Business telephones   Martin
• [Asterisk-Users] Business telephones   Martin
• [Asterisk-Users] TE406P audio drops   Matt Florell
• [Asterisk-Users] TE406P audio drops   Kevin P. Fleming
• [Asterisk-Users] GotoIf sample...   Bruno De Luca
• [Asterisk-Users] TE406P audio drops   Andrew Kohlsmith
• [Asterisk-Users] Can get IAX connection but no SIP connection?   Tim P
• [Asterisk-Users] Assessing network quality   Chris Mason (Lists)
• [Asterisk-Users] civil emergency comms: Asterisk + HAM   Mike M
• [Asterisk-Users] TE406P audio drops   Shane Burrell
• [Asterisk-Users] Application rxfax missing ?   Arne Morten Johansen
• [Asterisk-Users] "all lines are busy"   Christian Buchter
• [Asterisk-Users] Asterisk overheating on VIA Epia MSeriesmotherboard   Technical Support
• [Asterisk-Users] Routing Time, Cisco vs. Asterisk   Mohammed Smadi
• [Asterisk-Users] Asterisk Community Participant; Katrina Refugee   Karl S. Katzke
• [Asterisk-Users] Asterisk overheating on VIA Epia MSeriesmoth erboard   Nathan C. Smith
• [Asterisk-Users] Asterisk overheating on VIA Epia M Seriesmotherboard   Karl S. Katzke
• [Asterisk-Users] TDM11B pinout   steve at daviesfam.org
• [Asterisk-Users] Channalized T1 and PRI with Asterisk   Michael D Schelin
• [Asterisk-Users] /dev/zap* is not showing up (gentoo, portage, asterisk 1.0.8)   Jachin Rupe
• [Asterisk-Users] PHP and ASterisk Manager   Anton Krall
• [Asterisk-Users] Asterisk BT100 Password Issue   Aisling
• [Asterisk-Users] Asterisk overheating on VIA Epia MSeriesmotherboard   Technical Support
• [Asterisk-Users] Asterisk BT100 Password Issue   Aisling
• [Asterisk-Users] Weird SIP behaviour   Nir Simionovich
• [Asterisk-Users] IP PBX Market Share and Growth   Kanuri, Seshu (Company IT)
• [Asterisk-Users] Threeway calling uses up two FXO lines   hugolivude
• [Asterisk-Users] PHP and ASterisk Manager   Matthew Boehm
• [Asterisk-Users] Can get IAX connection but no SIP connection?   Konrads Smelkovs
• [Asterisk-Users] PHP and ASterisk Manager   Jeff Buchbinder
• [Asterisk-Users] Loging agents in   José Pablo Ezequiel Fernández
• [Asterisk-Users] Threeway calling uses up two FXO lines   Michael Graves
• [Asterisk-Users] PHP and ASterisk Manager   Sherwood McGowan
• [Asterisk-Users] PHP and ASterisk Manager   Anton Krall
• [Asterisk-Users] Wireless router with built-in VOIP(FXS) ports for Ansterisk   Karunakar Chemudugunta
• [Asterisk-Users] PHP and ASterisk Manager   Anton Krall
• [Asterisk-Users] PHP and ASterisk Manager   Anton Krall
• [Asterisk-Users] Asterisk Community Participant; Katrina Refugee   Mike M
• [Asterisk-Users] Transfering to voicemail problem with 1.2beta   Jeremy Koski
• [Asterisk-Users] Wireless router with built-in VOIP(FXS) ports forAnsterisk   Sherwood McGowan
• [Asterisk-Users] Wireless router with built-in VOIP(FXS) ports forAnsterisk   Garrett Smith
• [Asterisk-Users] Polycom ip301 hangs at Running "sip.ld"   Jonathan k. Creasy
• [Asterisk-Users] Asterisk Community Participant; Katrina Refugee   Austin Denyer
• [Asterisk-Users] Wireless router with built-in VOIP(FXS) ports forAnsterisk   Damon Estep
• [Asterisk-Users] Asterisk scenario   housi mueller
• [Asterisk-Users] Application rxfax missing ?   BJ Weschke
• [Asterisk-Users] Sipura Devices and Asterisk?   Sherwood McGowan
• [Asterisk-Users] Can get IAX connection but no SIP connection?   Tim P
• [Asterisk-Users] Sipura Devices and Asterisk?   Benjamin Amezcua
• [Asterisk-Users] "all lines are busy"   Christian Buchter
• [Asterisk-Users] Polycom ip301 hangs at Running "sip.ld"   Doug Lytle
• [Asterisk-Users] Asterisk overheating on VIA Epia MSeriesmoth erboard   Remco Barende
• [Asterisk-Users] unicall and cvs head   Denis Galvão - iSolve
• [Asterisk-Users] asterisk handling of old voicemail messages   Damon Estep
• [Asterisk-Users] Utility to find length of wav49 file   Malcolm Taylor
• [Asterisk-Users] AstriCon Update: Please Register ASAP - Free Phones   Steven Sokol
• [Asterisk-Users] Sipura Devices and Asterisk?   Jonathan k. Creasy
• [Asterisk-Users] Asterisk overheating on VIA Epia MSeriesmotherboard   Jonathan k. Creasy
• [Asterisk-Users] Sipura Devices and Asterisk?   Sherwood McGowan
• [Asterisk-Users] Asterisk overheating on VIA Epia MSeriesmoth erboard   Francesco Peeters
• [Asterisk-Users] Polycom ip301 hangs at Running "sip.ld"   Jonathan k. Creasy
• [Asterisk-Users] Queue AgentCallBackLogin   João Paulo Antunes
• [Asterisk-Users] Sipura Devices and Asterisk?   Benjamin Amezcua
• [Asterisk-Users] Sipura Devices and Asterisk?   Karun Chemudugunta
• [Asterisk-Users] Sipura Devices and Asterisk?   Sander
• [Asterisk-Users] one extension goes straight to voicemail, others don't   taran
• [Asterisk-Users] Queue AgentCallBackLogin   Sander
• [Asterisk-Users] Sipura Devices and Asterisk?   Jonathan k. Creasy
• [Asterisk-Users] Routing depending on sip response code?   Sherwood McGowan
• [Asterisk-Users] Asterisk overheating on VIA Epia M Seriesmotherboard   Darrick Hartman
• [Asterisk-Users] Good Polycom Dealer?   Kenny Kant
• [Asterisk-Users] Good Polycom Dealer?   Corey S. McFadden
• [Asterisk-Users] Good Polycom Dealer?   Garrett Smith
• [Asterisk-Users] Asterisk overheating on VIA Epia MSeriesmotherboard   Rob Thomas
• [Asterisk-Users] Good Polycom Dealer?   Kristian Kielhofner
• [Asterisk-Users] Asterisk Community Participant; Katrina Refugee   Jay Milk
• [Asterisk-Users] Good Polycom Dealer?   Tarpo, Louie
• [Asterisk-Users] Speaking of Polycom phones...updated ROM: ouch!   Doug
• [Asterisk-Users] Asterisk as SIP/H.323 Signalling Gateway   Joerg Furrer
• [Asterisk-Users] /dev/zap* is not showing up (gentoo, portage, asterisk 1.0.8   FELIX E SKOWRONEK
• [Asterisk-Users] Good Polycom Dealer?   Michael Welter
• [Asterisk-Users] Cisco 7960 upgrades   Sascha Ferley
• [Asterisk-Users] Speaking of Polycom phones...updated ROM: ouch!   Jesse Keating
• [Asterisk-Users] /dev/zap* is not showing up (gentoo, portage, asterisk 1.0.8   Jachin Rupe
• [Asterisk-Users] Occasional quiet voicemails   Anthony Rodgers
• [Asterisk-Users] /dev/zap* is not showing up (gentoo, portage, asterisk 1.0.8   Rob Thomas
• [Asterisk-Users] /dev/zap* is not showing up (gentoo, portage, asterisk 1.0.8   FELIX E SKOWRONEK
• [Asterisk-Users] Routing depending on sip response code?   Moises Silva
• [Asterisk-Users] callerid...   Moises Silva
• [Asterisk-Users] one extension goes straight to voicemail, others don't   El Flynn
• [Asterisk-Users] Asterisk scenario   El Flynn
• [Asterisk-Users] Unable to hear.   Moises Silva
• [Asterisk-Users] Good Polycom Dealer?   Tarpo, Louie
• [Asterisk-Users] SIP Callgroups   El Flynn
• [Asterisk-Users] Utility to find length of wav49 file   El Flynn
• [Asterisk-Users] CTI and Asterisk   Stefano Blasco
• [Asterisk-Users] Going crazy with FAX :-(   El Flynn
• [Asterisk-Users] Asterisk scenario   housi mueller
• [Asterisk-Users] Asterisk scenario   Carlos Antunes
• [Asterisk-Users] Cisco 7960 upgrades   Cory Andrews
• [Asterisk-Users] Some problems (SendDTMF, Wait, Parked Calls)   Rubens Sanchez
• [Asterisk-Users] Cisco 7960 upgrades   Rich Adamson
• [Asterisk-Users] Occasional quiet voicemails   Rich Adamson
• [Asterisk-Users] Good Polycom Dealer?   Darrick Hartman
• [Asterisk-Users] PRI in and out   Rod Bacon
• [Asterisk-Users] PRI in and out   El Flynn
• [Asterisk-Users] USING TWO ACCOUNTS WITH BROADVOICE   Bartosz Wegrzyn - asterisk
• [Asterisk-Users] Which Linux distribution?   YT Lim
• [Asterisk-Users] Compiling Zaptel   Mike Roberts
• [Asterisk-Users] PRI in and out   C F
• [Asterisk-Users] PRI in and out   C F
• [Asterisk-Users] PRI in and out   Rod Bacon
• [Asterisk-Users] Working example of ALERT_INFO with Cisco ATAs?   Brian Capouch
• [Asterisk-Users] IAX2 Problems causing server to hang   Alex Kauffmann
• [Asterisk-Users] Working example of ALERT_INFO with Cisco ATAs?   Alexander Lopez
• [Asterisk-Users] queues   altus
• [Asterisk-Users] Which Linux distribution?   Rubens Sanchez
• [Asterisk-Users] asterisk, SIP, Re-INVITEs and different contexts   Irakli Natsvlishvili
• [Asterisk-Users] PHP and ASterisk Manager   Christoph Eicke
• [Asterisk-Users] queues   altus
• [Asterisk-Users] Some info about Cisco's 79xx, and Sipura's phones   Irakli Natsvlishvili
• [Asterisk-Users] Utility to find length of wav49 file   Malcolm Taylor
• [Asterisk-Users] zaptel init script   Christian Richter
• [Asterisk-Users] PRI in and out   Rich Adamson
• [Asterisk-Users] Hosted PBX (vPBX) and Call/PickUP Groups   René Mayorga
• [Asterisk-Users] PRI in and out   altus
• [Asterisk-Users] asterisk, SIP, Re-INVITEs and different contexts   Irakli Natsvlishvili
• [Asterisk-Users] asterisk, SIP, Re-INVITEs and different contexts   Olle E. Johansson
• [Asterisk-Users] Max concurrent faxes with txfax/spandsp?   Roger Schreiter
• [Asterisk-Users] zaptel init script   Tzafrir Cohen
• [Asterisk-Users] zaptel init script   Christian Richter
• [Asterisk-Users] Which Linux distribution?   John Daragon
• [Asterisk-Users] Ethernet / TcpIp phones   Alex
• [Asterisk-Users] How to connect many analog lines to Asterisk?   Josip Gracin
• [Asterisk-Users] How to connect many analog lines to Asterisk?   gincantalupo
• [Asterisk-Users] Ethernet / TcpIp phones   gincantalupo
• [Asterisk-Users] Ethernet / TcpIp phones   Christoph Eicke
• [Asterisk-Users] presence settings and Eyebeam   Vahan Yerkanian
• [Asterisk-Users] CTI and Asterisk   Giovanni Miano
• [Asterisk-Users] Which Linux distribution?   Tzafrir Cohen
• [Asterisk-Users] presence settings and Eyebeam   Olle E. Johansson
• [Asterisk-Users] Working example of ALERT_INFO with Cisco ATAs?   Olle E. Johansson
• [Asterisk-Users] ISDN PBX integration   Shahar Livne
• [Asterisk-Users] channels VHF/ HF radio in asterisk   makevuy
• [Asterisk-Users] Packet Cable   Chris Mason (Lists)
• [Asterisk-Users] "-- PROGRESS with cause code 34 received"?   Roy Sigurd Karlsbakk
• [Asterisk-Users] Which Linux distribution?   John Daragon
• [Asterisk-Users] presence settings and Eyebeam   Vahan Yerkanian
• [Asterisk-Users] Some problems (SendDTMF, Wait, Parked Calls)   Flobi
• [Asterisk-Users] Hosted PBX (vPBX) and Call/PickUP Groups   Flobi
• [Asterisk-Users] Desincripcion de la lista de Asterisk   Will Velez
• [Asterisk-Users] Desincripcion de la lista de Asterisk   Flobi
• [Asterisk-Users] channels VHF/ HF radio in asterisk   Mark Phillips
• [Asterisk-Users] Packet Cable   Mark Phillips
• [Asterisk-Users] Eeven Stranger - Asterisk BT100 Password Issue   Aisling
• [Asterisk-Users] Which Linux distribution?   Mark Phillips
• [Asterisk-Users] Eeven Stranger - Asterisk BT100 Password Issue   Flobi
• [Asterisk-Users] 2 X100P and SIP outbound routing   Paul Goodyear
• [Asterisk-Users] Eeven Stranger - Asterisk BT100 Password Issue   Aisling
• [Asterisk-Users] IAX PBX responds to IAX registration with expires time=0   Maciek
• [Asterisk-Users] Speex codec - Out of buffer space   David Hajek
• [Asterisk-Users] Polycom 300 with latest 1.5.3 firmware not registering   Jorge Alayon
• [Asterisk-Users] Asterisk overheating on VIA Epia M Seriesmotherboard   Dustin Wildes
• RES: [Asterisk-Users] Billing - Disable accounts when balance gets0 value   itn at itn.com.br
• [Asterisk-Users] Extensions - Realtime   Flobi
• [Asterisk-Users] channels VHF/ HF radio in asterisk   makevuy
• [Asterisk-Users] How to connect many analog lines to Asterisk?   Darren Wright
• [Asterisk-Users] Re: ISDN PBX integration   Shahar Livne
• [Asterisk-Users] channels VHF/ HF radio in asterisk   Jonathan k. Creasy
• [Asterisk-Users] Which Linux distribution?   garth at bitco.co.za
• [Asterisk-Users] CONNECT ACK timeout in libpri   gshaw at cistel.com
• [Asterisk-Users] Extensions - Realtime   Robert Bedell
• [Asterisk-Users] PHP and ASterisk Manager   Anton Krall
• [Asterisk-Users] Asterisk crashed?   Arik Funke
• [Asterisk-Users] Extensions - Realtime   Matthew Boehm
• [Asterisk-Users] TDM400 and Phone does not 'ring'   Wilson Pickett
• [Asterisk-Users] PBX Replacement   Sean Cook
• [Asterisk-Users] How to connect many analog lines to Asterisk?   C F
• [Asterisk-Users] PRI in and out   C F
• [Asterisk-Users] Extensions - Realtime   Flobi
• [Asterisk-Users] ArtDio IPF-2000 unable to send audio to Cisco 7940 until placed on hold and resumed   Michael Coburn
• [Asterisk-Users] How to connect many analog lines to Asterisk?   Josip Gracin
• [Asterisk-Users] Occasional quiet voicemails   Anthony Rodgers
• [Asterisk-Users] Re: Polycom 300 with latest 1.5.3 firmware not registering   Noah Miller
• [Asterisk-Users] Extensions - Realtime   Flobi
• [Asterisk-Users] Lock Extension   Stephen
• [Asterisk-Users] Working example of ALERT_INFO with Cisco ATAs?   Brian Capouch
• [Asterisk-Users] Second Line does not Connect - HELP - misdn,sip   Pascal Speck
• [Asterisk-Users] Speex codec - Out of buffer space   Rich Adamson
• [Asterisk-Users] ztcfg Kills My Dial Tone   Shaw Terwilliger
• [Asterisk-Users] Extensions - Realtime   Flobi
• [Asterisk-Users] Extensions - Realtime   Flobi
• [Asterisk-Users] How to connect many analog lines to Asterisk?   Francesco Peeters
• [Asterisk-Users] How to connect many analog lines to Asterisk?   Jonathan k. Creasy
• [Asterisk-Users] Extensions - Realtime   Matthew Boehm
• [Asterisk-Users] Extensions - Realtime   Matthew Boehm
• [Asterisk-Users] How to connect many analog lines to Asterisk?   Josip Gracin
• [Asterisk-Users] How to connect many analog lines to Asterisk?   Jonathan k. Creasy
• [Asterisk-Users] Extensions - Realtime   Flobi
• [Asterisk-Users] Asterisk crashed?   Flobi
• [Asterisk-Users] asterisk.org blocked - rejecting connections   Flobi
• [Asterisk-Users] "-- PROGRESS with cause code 34 received"?   Matt Fredrickson
• [Asterisk-Users] Speaking of Polycom phones...updated ROM: ouch!   Doug
• [Asterisk-Users] TDM card and voicemail volume   tmassey at obscorp.com
• [Asterisk-Users] Speaking of Polycom phones...updated ROM: ouch!   Jesse Keating
• [Asterisk-Users] asterisk.org blocked - rejecting connections   Martin
• [Asterisk-Users] Occasional quiet voicemails   Rich Adamson
• [Asterisk-Users] Which Linux distribution?   canuck15
• [Asterisk-Users] TDM400P not detecting hangup and not hanging up.   Faris Raouf
• [Asterisk-Users] asterisk.org blocked - rejecting connections   Martin
• [Asterisk-Users] Occasional quiet voicemails   Martin
• [Asterisk-Users] Motherboard and processor recommendations   Soner Tari
• [Asterisk-Users] Hosted PBX (vPBX) and Call/PickUP Groups   René Mayorga
• [Asterisk-Users] Asterisk with Vonage problems   Adrian A
• [Asterisk-Users] Hosted PBX (vPBX) and Call/PickUP Groups   C F
• [Asterisk-Users] sip - aastra 9133i   Martin
• [Asterisk-Users] Speaking of Polycom phones...updated ROM: ouch!   Doug
• [Asterisk-Users] TDM card and voicemail volume   Rich Adamson
• [Asterisk-Users] Several SIP clients behind router register with the same IP, messing up call routing, any ideas?   Roman Zhovtulya
• [Asterisk-Users] Several SIP clients behind router register with the same IP, messing up call routing, any ideas?   Flobi
• [Asterisk-Users] Utility to find length of wav49 file   brett at websmyths.com
• [Asterisk-Users] externpass in voicemail   Anton Krall
• [Asterisk-Users] Desincripcion de la lista de Asterisk   Carlos Alperin
• [Asterisk-Users] TDM400P not detecting hangup and not hanging up.   canuck15
• [Asterisk-Users] TDM04B + Voicemail poor Quality   tmassey at obscorp.com
• [Asterisk-Users] asterisk-statv2 showing blank screens   William Bohannan
• [Asterisk-Users] IAXy - no dailtone   Jimmy
• [Asterisk-Users] Remote Provisioning for the PA1688 phones.   Maps
• [Asterisk-Users] IVR Documentation an Sample.   PJ Santos
• [Asterisk-Users] g729 test   Michael Welter
• [Asterisk-Users]   Juan Salas
• [Asterisk-Users] g729 test   Andrew Kohlsmith
• [Asterisk-Users] IVR Documentation and Samples.   PJ Santos
• [Asterisk-Users] te110p   Pietro U
• [Asterisk-Users] RDNIS   Jonathan k. Creasy
• [Asterisk-Users] Want to use a remotely location POTS phone   Patrick Campbell
• [Asterisk-Users] Several SIP clients behind router register withthe same IP, messing up call routing, any ideas?   Roman Zhovtulya
• [Asterisk-Users] Want to use a remotely location POTS phone   Andrew Kohlsmith
• [Asterisk-Users] Want to use a remotely location POTS phone   AbdelRahman Tarzi
• [Asterisk-Users] Motherboard and processor recommendations   Tzafrir Cohen
• [Asterisk-Users] Need Help - Losing first few seconds of call when using Broadvoice   Nikunj Kothari
• [Asterisk-Users] Motherboard and processor recommendations   canuck15
• [Asterisk-Users] Problem with PRI channels, restarted after every call.   Boris Bakchiev
• [Asterisk-Users] Hack for Canadian weather   Colin Anderson
• [Asterisk-Users] Re: ztcfg Kills My Dial Tone [solved]   Shaw Terwilliger
• [Asterisk-Users] TDM400P not detecting hangup and not hanging up.   stevanus
• [Asterisk-Users] TDM400P problems   Martin
• [Asterisk-Users] Business telephones   Martin
• [Asterisk-Users] asterisk frequently dead   stevanus
• [Asterisk-Users] asterisk frequently dead   Andrew Kohlsmith
• [Asterisk-Users] Not can call to PSTN   diego.ramirez at acodin.com.co
• [Asterisk-Users] asterisk frequently dead   stevanus
• [Asterisk-Users] Sipura-2002 Can not make outgoing calls, incoming calls works OK   Joseph
• [Asterisk-Users] RDNIS   Gary Reuter
• [Asterisk-Users] OT: Differences between test equipment   tmassey at obscorp.com
• [Asterisk-Users] asterisk, SIP, Re-INVITEs and different contexts   irakli.natsvlishvili at thinkingvoice.com
• [Asterisk-Users] I should never be called!   Jason Kim
• [Asterisk-Users] IAXy - no dailtone   Francesco Peeters
• [Asterisk-Users] 410P upgrade to 411P?   Rod Bacon
• [Asterisk-Users] 410P upgrade to 411P?   Matt Florell
• [Asterisk-Users] (no subject)   prashant yadav
• [Asterisk-Users] Not can call to PSTN   Tzafrir Cohen
• [Asterisk-Users] (OT) Dialplan Standards for Business/Offices   KRTorio
• [Asterisk-Users] Transfer calls from cellphone   Arnar Birgisson
• [Asterisk-Users] Working example of ALERT_INFO with Cisco ATAs?   Olle E. Johansson
• [Asterisk-Users] Setting up multiple trunk groups with different internal ring groups   Matt Love
• [Asterisk-Users] How to increase delay before incoming call answer with tdm400p   taf taffey
• [Asterisk-Users] Motherboard and processor recommendations   Soner Tari
• [Asterisk-Users] How to increase delay before incoming call answer with tdm400p   Olle E. Johansson
• [Asterisk-Users] Motherboard and processor recommendations   Dave Cotton
• [Asterisk-Users] who use astlinux with booting from DOM?   oncemore
• [Asterisk-Users] Contexts are not being created - Asterisk BT100 Password Issue   Aisling
• [Asterisk-Users] How to increase delay before incoming call answer with tdm400p   taf taffey
• [Asterisk-Users] 2 X100P and SIP inbound routing   Paul Goodyear
• [Asterisk-Users] pri gateway   Baris Simsek
• [Asterisk-Users] Immediate response dial   Dias Badekas
• [Asterisk-Users] Extension a   Il Neofita
• [Asterisk-Users] How to increase delay before incoming call answer with tdm400p   Andrew Kohlsmith
• [Asterisk-Users] MAX PRI for single server (was: Not enough lines available for Asterisk implemetation)   asterisk groups
• [Asterisk-Users] How to increase delay before incoming call answer with tdm400p   taf taffey
• [Asterisk-Users] sending fax....i'm in trouble !   Michele "O-Zone" Pinassi
• [Asterisk-Users] pri gateway   altus
• [Asterisk-Users] Additional: Several SIP clients behind router registerwiththe same IP, messing up call routing, any ideas?   Roman Zhovtulya
• [Asterisk-Users] Sip clients through proxy   Kanishka Somaratne
• [Asterisk-Users] pri gateway   Baris Simsek
• [Asterisk-Users] Yuxin hardphones feedback   Nicolas Schmerber
• [Asterisk-Users] Want to use a remotely location POTS phone   Patrick Campbell
• [Asterisk-Users] Distinctive ringing on Cisco 79xx   Bruce Komito
• [Asterisk-Users] Immediate response dial   Dave Cotton
• [Asterisk-Users] Hangup problem   Marek Zachara
• [Asterisk-Users]   Flobi
• [Asterisk-Users] Motherboard and processor recommendations   Matt Florell
• [Asterisk-Users] Hangup problem   Tzafrir Cohen
• [Asterisk-Users] Insert Subject Here   Matthew Boehm
• [Asterisk-Users] OT: Differences between test equipment   Rich Adamson
• [Asterisk-Users] How to increase delay before incoming call answer with tdm400p   Flobi
• [Asterisk-Users] cvs head and seqno 102 (Critical Response) messages for Cisco 7960   Chris Stenton
• [Asterisk-Users] Transfer calls from cellphone   BJ Weschke
• [Asterisk-Users] Motherboard and processor recommendations   Chris
• [Asterisk-Users] Want to use a remotely location POTS phone   Rich Adamson
• [Asterisk-Users] Not enough lines available for Asterisk implemetation   Wayne Gemmell
• [Asterisk-Users] power over ethernet hub/switch   gincantalupo
• [Asterisk-Users] Asterisk & Euro-ISDN   Javier Vázquez
• [Asterisk-Users] power over ethernet hub/switch   Sean Milheim
• [Asterisk-Users] IVR Documentation and Samples.   Juan Salas
• [Asterisk-Users] All Circuits are busy   Pietro U
• [Asterisk-Users] MAX PRI for single server (was: Not enough lines available for Asterisk implemetation)   Simone Cittadini
• [Asterisk-Users] 2 X100P and SIP inbound routing   Paul Goodyear
• [Asterisk-Users] All Circuits are busy   Dave Cotton
• [Asterisk-Users] MAX PRI for single server (was: Not enough lines available for Asterisk implemetation)   Andrew Kohlsmith
• [Asterisk-Users] channels VHF/ HF radio in asterisk   Carlos Alperin
• [Asterisk-Users] power over ethernet hub/switch   Carlos Alperin
• [Asterisk-Users] power over ethernet hub/switch   Geoff Manning
• [Asterisk-Users] Multiple Instances of Asterisk (no contexts)   Geoff Karl
• [Asterisk-Users] MAX PRI for single server (was: Not enough linesavailable for Asterisk implemetation)   Chris Modesitt
• [Asterisk-Users] asterisk frequently dead   Matt
• [Asterisk-Users] Re: MAX PRI for single server (was: Not enough lines available for Asterisk implemetation)   Wayne Gemmell
• [Asterisk-Users] power over ethernet hub/switch   gincantalupo
• [Asterisk-Users] All Circuits are busy   Pietro U
• [Asterisk-Users] Multiple Instances of Asterisk (no contexts)   Matthew Boehm
• [Asterisk-Users] Problem: Got SIP response 481 "Call Leg/Transaction Does Not Exist"   Omar McKenzie
• [Asterisk-Users] Motherboard and processor recommendations   Matt Florell
• [Asterisk-Users] Slight OT: Multi WAN Router and SIP Calls   Geoff Manning
• [Asterisk-Users] Re: MAX PRI for single server (was: Not enough lines available for Asterisk implemetation)   Tony Mountifield
• [Asterisk-Users] Pass through of T.38   Roger Schreiter
• [Asterisk-Users] channels VHF/ HF radio in asterisk   Huddleston, Robert
• [Asterisk-Users] Re: MAX PRI for single server   Matthew Boehm
• [Asterisk-Users] Motherboard and processor recommendations   Martin
• [Asterisk-Users] Not enough lines available for Asterisk implemetation   John Daragon
• [Asterisk-Users] MAX PRI for single server   Jason Becker
• [Asterisk-Users] MAX PRI for single server (was: Not enoughlines available for Asterisk implemetation)   PistolPete
• [Asterisk-Users] Call goes through, but no audio   Doug
• [Asterisk-Users] MAX PRI for single server (was: Not enough lines available for Asterisk implemetation)   Wayne Gemmell
• [Asterisk-Users] All Circuits are busy   Dave Cotton
• [Asterisk-Users] Motherboard and processor recommendations   Chris
• [Asterisk-Users] Server Brand   Kenny Kant
• [Asterisk-Users] Txfax   Il Neofita
• [Asterisk-Users] Multiple Line Appearances / Why use this?   Kenny Kant
• [Asterisk-Users] Re: MAX PRI for single server   Andrew Kohlsmith
• [Asterisk-Users] Re: MAX PRI for single server   Tony Mountifield
• [Asterisk-Users] g729 test   Michael Welter
• [Asterisk-Users] Call goes through, but no audio   Sherwood McGowan
• [Asterisk-Users] How to cascade dial status back through IAX   Tony Mountifield
• [Asterisk-Users] All Circuits are busy   Pietro U
• [Asterisk-Users] Server Brand   Paul
• [Asterisk-Users] MAX PRI for single server   Matthew Boehm
• [Asterisk-Users] Pass through of T.38   Matthew Boehm
• [Asterisk-Users] channels VHF/ HF radio in asterisk   Carlos Alperin
• [Asterisk-Users] Spandsp   Tim P
• [Asterisk-Users] How do you change the festival voice   Kris Edwards
• [Asterisk-Users] voice over atlantic   asterisk groups
• [Asterisk-Users] Multiple Instances of Asterisk (no contexts)   Geoff Karl
• [Asterisk-Users] Transfer calls from cellphone   Arnar Birgisson
• [Asterisk-Users] g729 test   Andrew Kohlsmith
• [Asterisk-Users] MAX PRI for single server   Jason Becker
• [Asterisk-Users] play each person's voicemail   Gary S MacKay
• [Asterisk-Users] Want to use a remotely location POTS phone   Patrick Campbell
• [Asterisk-Users] Spandsp   diego.ramirez at acodin.com.co
• [Asterisk-Users] CVSHEAD callerid not working   John Hill
• [Asterisk-Users] Not can call to PSTN   diego.ramirez at acodin.com.co
• [Asterisk-Users] externpass in voicemail   Anton Krall
• [Asterisk-Users] (OT) Dialplan Standards for Business/Offices   Flobi
• [Asterisk-Users] Additional: Several SIP clients behind router registerwiththe same IP, messing up call routing, any ideas?   Flobi
• [Asterisk-Users] MAX PRI for single server   Matthew Boehm
• [Asterisk-Users] (no subject)   Mark Phillips
• [Asterisk-Users] Problem with IAXy   Waldo Rubinstein
• [Asterisk-Users] Asterisk overheating on VIA Epia MSeriesmoth erboard   Patrick
• [Asterisk-Users] Multiple Line Appearances / Why use this?   Flobi
• [Asterisk-Users] MAX PRI for single server   Carlos Antunes
• [Asterisk-Users] MAX PRI for single server   Matthew Boehm
• [Asterisk-Users] play each person's voicemail   Flobi
• [Asterisk-Users] civil emergency comms: Asterisk + HAM   Mike Hemstock
• [Asterisk-Users] Distinctive ringing on Cisco 79xx   Flobi
• [Asterisk-Users] civil emergency comms: Asterisk + HAM   Mark Phillips
• [Asterisk-Users] MAX PRI for single server   Jason Becker
• [Asterisk-Users] Distinctive ringing on Cisco 79xx   James Sizemore
• [Asterisk-Users] MAX PRI for single server   Carlos Antunes
• [Asterisk-Users] Conferences under asterisk   Alex Kobalto
• [Asterisk-Users] Distinctive ringing on Cisco 79xx   Flobi
• [Asterisk-Users] Motherboard and processor recommendations   canuck15
• [Asterisk-Users] TDM400P not detecting hangup and not hanging up   Faris Raouf
• [Asterisk-Users] Server Brand   canuck15
• [Asterisk-Users] voice over atlantic   David Hajek
• [Asterisk-Users] Additional: Several SIP clients behind routerregisterwiththe same IP, messing up call routing, any ideas?   Roman Zhovtulya
• [Asterisk-Users] voice over atlantic   Wiley Siler
• [Asterisk-Users] Siupra-2002 with astersik   Joseph
• [Asterisk-Users] SIP/2.0 487 Request Terminated problem on Cisco 7960   Chris Stenton
• [Asterisk-Users] Siupra-2002 with astersik   Sander
• [Asterisk-Users] MAX PRI for single server (was: Not enoughlinesavailable for Asterisk implemetation)   Damon Estep
• [Asterisk-Users] TE411P zapata.conf, monitoring echo cancellation and echo tail size   Eric Bishop
• [Asterisk-Users] voice over atlantic   David Hajek
• [Asterisk-Users] voice over atlantic   David Hajek
• [Asterisk-Users] Siupra-2002 with astersik   Joseph
• [Asterisk-Users] voice over atlantic   Wiley Siler
• [Asterisk-Users] voice over atlantic   David Hajek
• [Asterisk-Users] voice over atlantic   Wiley Siler
• [Asterisk-Users] How do you change the festival voice   Paul
• [Asterisk-Users] Txfax   Roger Schreiter
• [Asterisk-Users] Announcement: ASTPP-1.2-Beta   Darren Wiebe
• [Asterisk-Users] Re: Problem with IAXy   Waldo Rubinstein
• [Asterisk-Users] Which Linux distribution?   YT Lim
• [Asterisk-Users] FW: Adtran TA 616   Nick Colton
• [Asterisk-Users] How do you change the festival voice   Martin
• [Asterisk-Users] IAX Trunking Weirdness   Rod Bacon
• [Asterisk-Users] Txfax   Chris Shipman
• [Asterisk-Users] sending fax   Chris Shipman
• [Asterisk-Users] sending fax   Chris Shipman
• [Asterisk-Users] sip log messages every few seconds   Martin
• [Asterisk-Users] sip log messages every few seconds   Martin
• [Asterisk-Users] TE411P zapata.conf, monitoring echo cancellation and echo tail size   Matt Florell
• [Asterisk-Users] sending fax   Wiley Siler
• [Asterisk-Users] sending fax   Matthew Gibson
• [Asterisk-Users] sending fax   Il Neofita
• [Asterisk-Users] sip log messages every few seconds   Andres
• [Asterisk-Users] IAX Trunking Weirdness   Rod Bacon
• [Asterisk-Users] sending fax   Chris Shipman
• [Asterisk-Users] TDM PCI Master abort   Paul Harris
• [Asterisk-Users] PRI and Caller ID when immediate=yes   Boris Bakchiev
• [Asterisk-Users] Sangoma   Cory Andrews
• [Asterisk-Users] TE411P zapata.conf, monitoring echo cancellation and echo tail size   Cory Andrews
• [Asterisk-Users] Solution for 12 to 16 FXO to asterisk connection   Erick Perez
• [Asterisk-Users] Solution for 12 to 16 FXO to asterisk connection   Damon Estep
• [Asterisk-Users] (OT) Dialplan Standards for Business/Offices   KRTorio
• [Asterisk-Users] Sangoma   Nathan C. Smith
• [Asterisk-Users] Solution for 12 to 16 FXO to asterisk connection   Erick Perez
• [Asterisk-Users] TE411P zapata.conf, monitoring echo cancellation and echo tail size   Eric Bishop
• [Asterisk-Users] Question about setup Grandstream HandyTone 488 SIP with Astersik to Travel throught NAT.   Maps
• [Asterisk-Users] Solution for 12 to 16 FXO to asterisk connection   Sean Cook
• [Asterisk-Users] FW: Adtran TA 616   Dave Weis
• [Asterisk-Users] Solution for 12 to 16 FXO to asterisk connection   Ben Brown
• [Asterisk-Users] Solution for 12 to 16 FXO to asterisk connection   Darren Wright
• [Asterisk-Users] Solution for 12 to 16 FXO to asterisk connection   Damon Estep
• [Asterisk-Users] Solution for 12 to 16 FXO to asterisk connection   Ben Brown
• [Asterisk-Users] Montreal usergroup   Adrien Laurent
• [Asterisk-Users] 410P upgrade to 411P?   Rod Bacon
• [Asterisk-Users] T400P vs TE405P   Darren Wright
• [Asterisk-Users] MINNESOTA: TwinCities Asterisk Users Group - Saturday 9/10/2005   asterisk_help at iwishi.nu
• [Asterisk-Users] T1 DSP Card to T1 - TXFAX RXFAX Posible Solved   peterb
• [Asterisk-Users] pri gateway   altus
• [Asterisk-Users] can not make call with Unicall (MFC/R2)   Le Van Khoa
• [Asterisk-Users] Not enough lines available for Asterisk implemetation   steve at daviesfam.org
• [Asterisk-Users] sending fax   Tzafrir Cohen
• [Asterisk-Users] Motherboard and processor recommendations   Clive
• [Asterisk-Users] MAX PRI for single server   steve at daviesfam.org
• [Asterisk-Users] MAX PRI for single server   steve at daviesfam.org
• [Asterisk-Users] Using E1 without power off simence pbx   Asterisk Sales
• [Asterisk-Users] voice over atlantic   Tzafrir Cohen
• [Asterisk-Users] Motherboard and processor recommendations   Tzafrir Cohen
• [Asterisk-Users] Motherboard and processor recommendations   Domjan Attila
• [Asterisk-Users] the number of incoming calls in queue   Gary Li
• [Asterisk-Users] TDM400P not detecting hangup and not hanging up   Soner Tari
• [Asterisk-Users] Re: SIP/2.0 487 Request Terminated problem on Cisco 7960   Olle E. Johansson
• [Asterisk-Users] sending fax   Il Neofita
• [Asterisk-Users] Re: SIP/2.0 487 Request Terminated problem onCisco 7960   Chee Foong
• [Asterisk-Users] Re: MAX PRI for single server   Simone Cittadini
• [Asterisk-Users] Hosted PBX (vPBX) and Call/PickUP Groups   Chee Foong
• [Asterisk-Users] asterisk pri heavy load testing (was MAX PRI for single server)   Simone Cittadini
• [Asterisk-Users] Debian Sarge, Kernel 2.6.13 and AVM Fritz!PCI v2.0 card   Rainer Maier
• [Asterisk-Users] sip log messages every few seconds   Olle E. Johansson
• [Asterisk-Users] OT Humo[u]r IVR Menu sample   Dave Cotton
• [Asterisk-Users] Re: Debian Sarge, Kernel 2.6.13 and AVM Fritz!PCI v2.0 card   Tony Mountifield
• [Asterisk-Users] Huge Echo   Marek Zachara
• [Asterisk-Users] Debian Sarge, Kernel 2.6.13 and AVM Fritz!PCI v2.0 card   Rainer Maier
• [Asterisk-Users] Huge Echo   Andreas Sikkema
• [Asterisk-Users] Siupra-2002 with astersik   Matt
• [Asterisk-Users] BRI debug, national ISDN speech call problem   Steven Cherry
• [Asterisk-Users] T400P vs TE405P   Matt Florell
• [Asterisk-Users] remote SIP phones   Gary Smith
• [Asterisk-Users] Huge Echo   Marek Zachara
• [Asterisk-Users] TDM11B pinout   Gary Smith
• [Asterisk-Users] Huge Echo   Sander
• [Asterisk-Users] Huge Echo   Soner Tari
• [Asterisk-Users] pri gateway   Sander
• [Asterisk-Users] T400P vs TE405P   Andrew Kohlsmith
• [Asterisk-Users] Huge Echo   Marek Zachara
• [Asterisk-Users] civil emergency comms: Asterisk + HAM   Mike M
• [Asterisk-Users] voice over atlantic   asterisk groups
• [Asterisk-Users] Huge Echo   Marek Zachara
• [Asterisk-Users] voice over atlantic   asterisk groups
• [Asterisk-Users] spandsp txfax multi page problem   Bertalan Gergaly
• [Asterisk-Users] Doesn't finishes callerid spill   Gurminder Arora
• [Asterisk-Users] Huge Echo   Soner Tari
• [Asterisk-Users] Huge Echo   Marek Zachara
• [Asterisk-Users] civil emergency comms: Asterisk + HAM   Peter Bowyer
• [Asterisk-Users] civil emergency comms: Asterisk + HAM   Peter Bowyer
• [Asterisk-Users] Re: spandsp txfax multi page problem   Nenad Radosavljevic
• [Asterisk-Users] MAX PRI for single server   Matthew Boehm
• [Asterisk-Users] civil emergency comms: Asterisk + HAM   Mark Phillips
• [Asterisk-Users] New CUT()   John Hill
• [Asterisk-Users] Huge Echo   Soner Tari
• [Asterisk-Users] sending fax   Chris
• [Asterisk-Users] civil emergency comms: Asterisk + HAM   Mark Phillips
• [Asterisk-Users] MAX PRI for single server   Andrew Kohlsmith
• [Asterisk-Users] Montreal usergroup   Andre Courchesne - Consultant
• [Asterisk-Users] Want to use a remotely location POTS phone   Rich Adamson
• [Asterisk-Users] Txfax   Roger Schreiter
• [Asterisk-Users] Huge Echo   Marek Zachara
• [Asterisk-Users] Re: New CUT()   Tony Mountifield
• [Asterisk-Users] sending fax   Chris
• [Asterisk-Users] Huge Echo   Soner Tari
• [Asterisk-Users] Sangoma   Cory Andrews
• [Asterisk-Users] siemens pbx what i ask techinician?   Pablo Allietti
• [Asterisk-Users] VIP-050   andrutto
• [Asterisk-Users] Motherboard and processor recommendations   Walt Reed
• [Asterisk-Users] civil emergency comms: Asterisk + HAM   Mike M
• [Asterisk-Users] Sangoma   Damon Estep
• [Asterisk-Users] Motherboard and processor recommendations   Damon Estep
• [Asterisk-Users] Huge Echo   Marek Zachara
• [Asterisk-Users] FW: Adtran TA 616   Nick Colton
• [Asterisk-Users] Huge Echo   Jared Armstrong
• [Asterisk-Users] Sangoma   Dave Cotton
• [Asterisk-Users] Sangoma   Cory Andrews
• [Asterisk-Users] Motherboard and processor recommendations   Paul
• [Asterisk-Users] "Registered SIP '202' ... expires 1800". Why does it expire   Zeeshan Zakaria
• [Asterisk-Users] Detecting retries in call files   Steve Hanselman
• [Asterisk-Users] Solution for 12 to 16 FXO to asterisk connection   Tom Rymes
• [Asterisk-Users] civil emergency comms: Asterisk + HAM   Mike M
• [Asterisk-Users] OH323 for HEAD? 0.7.1 doesn't compile.   Tony Mountifield
• [Asterisk-Users] siemens pbx what i ask techinician?   Sander
• [Asterisk-Users] Storing extension prefs. in MySQL   Arnar Birgisson
• [Asterisk-Users] Huge Echo   Marek Zachara
• [Asterisk-Users] Changing User-Agent: Asterisk PBX   ChB
• [Asterisk-Users] Storing extension prefs. in MySQL   Andreas Sikkema
• [Asterisk-Users] reload   Urban
• [Asterisk-Users] Motherboard and processor recommendations   Tzafrir Cohen
• [Asterisk-Users] "Registered SIP '202' ... expires 1800". Why does it expire   Olle E. Johansson
• [Asterisk-Users] Siupra-2002 with astersik   Joseph
• [Asterisk-Users] woomera doesn't work (same OpenH323 problem as with chan_h323)   Tony Mountifield
• [Asterisk-Users] Asterisk connected to Concept XI520   Stefan Tichy
• [Asterisk-Users] Huge Echo   Mojo with Horan & Company, LLC
• [Asterisk-Users] "Registered SIP '202' ... expires 1800". Why does it expire   Tony Hoyle
• [Asterisk-Users] Re: siemens pbx what i ask techinician?   Sander
• [Asterisk-Users] Huge Echo   Marek Zachara
• [Asterisk-Users] RE:NewCUT()   John Hill
• [Asterisk-Users] "Registered SIP '202' ... expires 1800". Why does it expire   Marek Zachara
• [Asterisk-Users] Re: siemens pbx what i ask techinician?   Pablo Allietti
• [Asterisk-Users] Huge Echo   Mojo with Horan & Company, LLC
• [Asterisk-Users] MAX PRI for single server   steve at daviesfam.org
• [Asterisk-Users] FW: Problem: Got SIP response 481 "Call Leg/Transaction Does Not Exist"   Omar McKenzie
• [Asterisk-Users] Sangoma   steve at daviesfam.org
• [Asterisk-Users] Storing extension prefs. in MySQL   Chris A. Icide
• [Asterisk-Users] Huge Echo   steve at daviesfam.org
• [Asterisk-Users] Huge Echo   steve at daviesfam.org
• [Asterisk-Users] adding DNIS digits   Darren Wright
• [Asterisk-Users] Re: siemens pbx what i ask techinician?   Pablo Allietti
• [Asterisk-Users] Error on BSD app_rxfax.so   Joshua Abbott
• [Asterisk-Users] "Registered SIP '202' ... expires 1800". Why does it expire   reticent
• [Asterisk-Users] FW: Problem: Got SIP response 481 "Call Leg/Transaction Does Not Exist"   Patrick
• [Asterisk-Users] musiconhold errors in 1.2.0-beta1   Daniel Corbe
• [Asterisk-Users] Setting Account Code?   Matt
• [Asterisk-Users] Huge Echo   Marek Zachara
• [Asterisk-Users] SIP registration issues   Martin
• [Asterisk-Users] Re: Setting Account Code?   Matt
• [Asterisk-Users] Huge Echo   Marek Zachara
• [Asterisk-Users] Huge Echo   Adam Robins
• [Asterisk-Users] Huge Echo   Rich Adamson
• [Asterisk-Users] Adding users in AMP   Zeeshan Zakaria
• [Asterisk-Users] Huge Echo   Marek Zachara
• [Asterisk-Users] Re: siemens pbx what i ask techinician?   Sander
• [Asterisk-Users] Adding users in AMP   Chris
• [Asterisk-Users] Re: siemens pbx what i ask techinician?   Sander
• [Asterisk-Users] Huge Echo   Rich Adamson
• [Asterisk-Users] realtime and presence   Tod Detre
• [Asterisk-Users] Re: siemens pbx what i ask techinician?   Pablo Allietti
• [Asterisk-Users] Polycom 501 Multiple Line Instances   Matthew Boehm
• [Asterisk-Users] AMP 1.10.009 released!   Ryan Courtnage
• [Asterisk-Users] RTP ports in use grows and grows...   Bryan Field-Elliot
• [Asterisk-Users] AMP 1.10.009 released!   Kanuri, Seshu (Company IT)
• [Asterisk-Users] ASTCC speaks and cut RTP channel   Ricardo Poppi
• [Asterisk-Users] Polycom 501 Multiple Line Instances   Jesse Keating
• [Asterisk-Users] Special handling of IAX circuit-busy vs busy   dobrien at trilogy.com
• [Asterisk-Users] Motherboard and processor recommendations   Remco Barende
• [Asterisk-Users] pri gateway   Rob Lith
• [Asterisk-Users] Asterisk Extension Language   Richard Cook
• [Asterisk-Users] ALERT_INFO   Hugh L. Johnson
• [Asterisk-Users] did edmonton   Joseph
• [Asterisk-Users] voice over atlantic   David Hajek
• [Asterisk-Users] Wait for dialtone   AbdelRahman Tarzi
• [Asterisk-Users] Announcement: FOP 0.23 released   Nicolás Gudiño
• [Asterisk-Users] spandsp   José Pablo Ezequiel Fernández
• [Asterisk-Users] Huge Echo   canuck15
• [Asterisk-Users] "Registered SIP '202' ... expires 1800". Why does it expire   Bob Goddard
• [Asterisk-Users] Siupra-2002 with astersik   AbdelRahman Tarzi
• [Asterisk-Users] Motherboard and processor recommendations   canuck15
• [Asterisk-Users] vm notif   Jimmy Smith
• [Asterisk-Users] Re: NewCUT()   Tony Mountifield
• [Asterisk-Users] Txfax   Luki
• [Asterisk-Users] AMP 1.10.009 released!   canuck15
• [Asterisk-Users] Newman Telecom files   Carlos Alperin
• [Asterisk-Users] AMP 1.10.009 released!   Ariel Batista
• [Asterisk-Users] Txfax   Chris Shipman
• [Asterisk-Users] Newman Telecom files   Juan Jose Comellas
• [Asterisk-Users] Transferred calls dropping out of MeetMe   asterisk at gatheringpoint.net
• [Asterisk-Users] civil emergency comms: Asterisk + HAM   Jonathan k. Creasy
• [Asterisk-Users] Solution for 12 to 16 FXO to asterisk connection   Jonathan k. Creasy
• [Asterisk-Users] Queue "abandon" count increments incorrectly?   Chad Scott
• [Asterisk-Users] Txfax   Il Neofita
• [Asterisk-Users] Motherboard and processor recommendations   Andrew Furey
• [Asterisk-Users] Txfax   Chris
• [Asterisk-Users] call volume   Jonathan k. Creasy
• [Asterisk-Users] call volume   Rob Lith
• [Asterisk-Users] call volume   Paul
• [Asterisk-Users] civil emergency comms: Asterisk + HAM   f6hqz-m at hamwlan.net
• [Asterisk-Users] GotoIf Syntax to match first digits   ChB
• [Asterisk-Users] PRI echo   Jason Kim
• [Asterisk-Users] GotoIf Syntax to match first digits   ChB
• [Asterisk-Users] Changing User-Agent: Asterisk PBX   Curt Moore
• [Asterisk-Users] vm notif   Curt Moore
• [Asterisk-Users] call volume   steve at daviesfam.org
• [Asterisk-Users] Fritz, mISDN, Help   Konrads Smelkovs
• [Asterisk-Users] Fritz, mISDN, Help   John Fawcett
• [Asterisk-Users] GotoIf Syntax to match first digits   ChB
• [Asterisk-Users] Fritz, mISDN, Help   Konrads Smelkovs
• [Asterisk-Users] Need some HFC-S help   Vedran Dakic
• [Asterisk-Users] Fritz, mISDN, Help   Jon Dean
• [Asterisk-Users] Wait for dialtone   Rich Adamson
• [Asterisk-Users] civil emergency comms: Asterisk + HAM   Mark Phillips
• [Asterisk-Users] VoipBuster again   Rudolf Ladyzhenskii
• [Asterisk-Users] "Registered SIP '202' ... expires 1800". Why does it expire   Olle E. Johansson
• [Asterisk-Users] OH323 for HEAD? 0.7.1 doesn't compile.   Michael Manousos
• [Asterisk-Users] Required hardware   Anders Svensson
• [Asterisk-Users] AGI programming work required   Brad Hughes
• [Asterisk-Users] VoipBuster again   Sander
• [Asterisk-Users] Voicemail problem   Narcis GRATIANU
• [Asterisk-Users] call tests   harry gaillac
• [Asterisk-Users] Voicemail problem   Brad Hughes
• [Asterisk-Users] Changing User-Agent: Asterisk PBX   ChB
• [Asterisk-Users] Motherboard and processor recommendations   f6hqz-m at hamwlan.net
• [Asterisk-Users] Problems with TE205P   Tim Nyce
• [Asterisk-Users] AGI programming work required   Brad Hughes
• [Asterisk-Users] Distinctive Ring Problems   Graham Kiff
• [Asterisk-Users] AGI problem with library path   Innocent Evil
• [Asterisk-Users] False Zap answer problem (Again)   Soner Tari
• [Asterisk-Users] AGI programming work required   Alexander Lopez
• [Asterisk-Users] Fritz, mISDN, Help   John Fawcett
• [Asterisk-Users] False Zap answer problem (Again)   Soner Tari
• [Asterisk-Users] Fritz, mISDN, Help   Tzafrir Cohen
• [Asterisk-Users] Voicemail problem   ChB
• [Asterisk-Users] play each person's voicemail   GLNX
• [Asterisk-Users] Broadcasting via Asterisk   Ozan Blotter
• [Asterisk-Users] False Zap answer problem (Again)   Soner Tari
• [Asterisk-Users] Wait for dialtone   John Novack
• [Asterisk-Users] Compiling Zaptel   Kevin P. Fleming
• [Asterisk-Users] civil emergency comms: Asterisk + HAM   Mike M
• [Asterisk-Users] Configuring SIPURA 2002 to work wih Asterisk   Paul Conn
• [Asterisk-Users] civil emergency comms: Asterisk + HAM   f6hqz-m at hamwlan.net
• [Asterisk-Users] civil emergency comms: Asterisk + HAM   Mike M
• [Asterisk-Users] TE110P reset   JOAO CARLOS MOURA
• [Asterisk-Users] VoipBuster again   Rudolf Ladyzhenskii
• [Asterisk-Users] TE110P reset   Jason Walker
• [Asterisk-Users] civil emergency comms: Asterisk + HAM   Chris Travers
• [Asterisk-Users] TE110P reset   Andrew Kohlsmith
• [Asterisk-Users] Echo Issue   Ken Johnson
• [Asterisk-Users] TE411P zapata.conf, monitoring echo cancellation and echo tail size   Kevin P. Fleming
• [Asterisk-Users] ALERT_INFO   C F
• [Asterisk-Users] TE110P reset   Jason Walker
• [Asterisk-Users] MAX PRI for single server   Kevin P. Fleming
• [Asterisk-Users] MAX PRI for single server   Kevin P. Fleming
• [Asterisk-Users] Echo Issue   Rich Adamson
• [Asterisk-Users] Ignore incomingcall?   Shawn Porter
• [Asterisk-Users] civil emergency comms: Asterisk + HAM   Michael D Schelin
• [Asterisk-Users] VoipBuster again   Sander
• [Asterisk-Users] Configuring SIPURA 2002 to work wih Asterisk   Sander
• [Asterisk-Users] SIP Connection Problems   Dovid B. Asterisk Users
• [Asterisk-Users] SIP Connection Problems   Jason Walker
• [Asterisk-Users] SIP Connection Problems   Alexander Lopez
• [Asterisk-Users] Configuring SIPURA 2002 to work wih Asterisk   Anders Svensson
• [Asterisk-Users] Motherboard and processor recommendations   Konrads Smelkovs
• [Asterisk-Users] Fritz, mISDN, Help   Konrads Smelkovs
• [Asterisk-Users] Integrating with existing analog PBX   Martin Allen
• [Asterisk-Users] TE406p no interrupts   Jason Kim
• [Asterisk-Users] TE406p no interrupts   Alexander Lopez
• [Asterisk-Users] Integrating with existing analog PBX   John Daragon
• [Asterisk-Users] TE406p no interrupts   Boris Bakchiev
• [Asterisk-Users] TE406p no interrupts   Jason Kim
• [Asterisk-Users] OpenH323-Channel Q.931-Problems with Gatekeeper   Sebastian Mangelkramer
• [Asterisk-Users] TE406p no interrupts   Jason Kim
• [Asterisk-Users] TE406p no interrupts   Boris Bakchiev
• [Asterisk-Users] Make asterisk call out   Andreas Moroder
• [Asterisk-Users] Ignore incomingcall?   David Cook
• [Asterisk-Users] Make asterisk call out   Brian Roy
• [Asterisk-Users] VoipBuster again -- WORKS NOW!!!!   Rudolf Ladyzhenskii
• [Asterisk-Users] Fritz, mISDN, Help   Tzafrir Cohen
• [Asterisk-Users] Using RedirectAction with queues   Josip Gracin
• [Asterisk-Users] TE406p no interrupts   Jason Kim
• [Asterisk-Users] Special handling of IAX circuit-busy vs busy   Kevin P. Fleming
• [Asterisk-Users] Pass through of T.38   Kevin P. Fleming
• [Asterisk-Users] False Zap answer problem (Again)   Soner Tari
• [Asterisk-Users] civil emergency comms: Asterisk + HAM   Mike M
• [Asterisk-Users] rotate * log file?   Matt Riddell
• [Asterisk-Users] rotate * log file?   Paul
• [Asterisk-Users] rotate * log file?   Rich Adamson
• [Asterisk-Users] AGI + Ruby   joe heitzeberg
• [Asterisk-Users] rotate * log file?   Tzafrir Cohen
• [Asterisk-Users] civil emergency comms: Asterisk + HAM   Derek Whitten
• [Asterisk-Users] ruby-agi 0.0.2 released   Mohammad Khan
• [Asterisk-Users] cdr_addon_mysql.so pb   alexandre zhang
• [Asterisk-Users] cdr_addon_mysql.so pb   Jonathan k. Creasy
• [Asterisk-Users] cdr_addon_mysql.so pb   Matthew Boehm
• [Asterisk-Users] rotate * log file?   Rich Adamson
• [Asterisk-Users] Australian Dial tone TDM400P   Asterisk Sales
• 回复： RE: [Asterisk-Users] cdr_addon_mysql.so pb   alexandre zhang
• [Asterisk-Users] civil emergency comms: Asterisk + HAM   Chris Travers
• [Asterisk-Users] David Choo/eServices/eSpore is overseas   David Choo
• 回复： Re: [Asterisk-Users] cdr_addon_mysql.so pb   alexandre zhang
• [Asterisk-Users] Make asterisk call out   Samy Antoun
• [Asterisk-Users] civil emergency comms: Asterisk + HAM   Don Fanning
• [Asterisk-Users] H323 with asterisk-ooh323c   Adam Rybak
• 回复： Re: [Asterisk-Users] cdr_addon_mysql.so pb   Matthew Boehm
• [Asterisk-Users] cdr_addon_mysql.so pb   JOAO CARLOS MOURA
• [Asterisk-Users] TE110P reset   JOAO CARLOS MOURA
• [Asterisk-Users] civil emergency comms: Asterisk + HAM   Michael D Schelin
• [Asterisk-Users] Call Waiting Tracking?   Nathan E. Pralle
• [Asterisk-Users] Make asterisk call out   Darren Wiebe
• [Asterisk-Users] Presence Fully Supported?   Trevor Peirce
• [Asterisk-Users] TE406p no interrupts   Boris Bakchiev
• [Asterisk-Users] Presence Fully Supported?   Paul Hales
• [Asterisk-Users] first character in line 11 missing   Ronald Wiplinger
• [Asterisk-Users] "Registered SIP '202' ... expires 1800". Why does it expire   Tony Hoyle
• [Asterisk-Users] civil emergency comms: Asterisk + HAM   Steve Totaro
• [Asterisk-Users] TE110P reset   Steve Totaro
• [Asterisk-Users] civil emergency comms: Asterisk + HAM   C. Hatton Humphrey
• [Asterisk-Users] TE406p no interrupts   Jason Kim
• [Asterisk-Users] "Registered SIP '202' ... expires 1800". Why doesit expire   Race Vanderdecken
• [Asterisk-Users] TE406p no interrupts   Boris Bakchiev
• [Asterisk-Users] Hosted PBX (vPBX) and Call/PickUP Groups   Matt Riddell
• [Asterisk-Users] Hosted PBX (vPBX) and Call/PickUP Groups   Paul
• [Asterisk-Users] Asterisk on AMD64   Joseph
• [Asterisk-Users] Syslog file size   YT Lim
• [Asterisk-Users] Syslog file size   Brad Hughes
• [Asterisk-Users] Syslog file size   YT Lim
• [Asterisk-Users] extensions.conf for VOXEE using SIP!!   Jerry James
• [Asterisk-Users] civil emergency comms: Asterisk + HAM   Don Fanning
• [Asterisk-Users] civil emergency comms: Asterisk + HAM   Don Fanning
• [Asterisk-Users] Anyone using Telasip, Caller ID presentation outbound??   Chris Coulthurst
• [Asterisk-Users] civil emergency comms: Asterisk + HAM   Don Fanning
• [Asterisk-Users] Asterisk and AMP installed now what?   Tommy Denton
• [Asterisk-Users] Calling from one port on a SIPURA 2002 to the other port.   Paul Conn
• [Asterisk-Users] Syslog file size   Samy Antoun
• [Asterisk-Users] Asterisk and AMP installed now what?   Chris
• [Asterisk-Users] TE406p no interrupts   Jason Kim
• [Asterisk-Users] Asterisk and AMP installed now what?   Samy Antoun
• [Asterisk-Users] Asterisk on AMD64   Dave Cotton
• [Asterisk-Users] Hang up not hanging up (New Zealand Indications??)   Simon
• [Asterisk-Users] Hang up not hanging up (New Zealand Indications??)   Dave Cotton
• [Asterisk-Users] How to remove the voice mail greeting...   Simon
• [Asterisk-Users] Sip phone will not connect   Tommy Denton
• [Asterisk-Users] How to remove the voice mail greeting...   Christian Wengel
• [Asterisk-Users] Asterisk on AMD64   Domjan Attila
• [Asterisk-Users] TDM400 and Phone does not 'ring'   Alex Ongena
• [Asterisk-Users] Hang up not hanging up (New Zealand Indications??)   Simon
• [Asterisk-Users] Hang up not hanging up (New Zealand Indications??)   Dave Cotton
• [Asterisk-Users] tdm400p wattage   gincantalupo
• [Asterisk-Users] ChanSpy with asterisk 1.0.9   Mark Roeten
• [Asterisk-Users] TDM400 and Phone does not 'ring'   Dave Cotton
• [Asterisk-Users] asteriskathome and cisco 2600   manggoro at telematikajava.com
• [Asterisk-Users] Syslog file size   Tzafrir Cohen
• [Asterisk-Users] chan_zap.c:8050 pri_dchannel: Ring requested on unconfigured channel 255/255 span 2   Alex Ongena
• [Asterisk-Users] Slightly OT: VoIPbuster   Rudolf Ladyzhenskii
• [Asterisk-Users] Zap Channel   VoIP Newbie
• [Asterisk-Users] Zap Channel   Alex Ongena
• [Asterisk-Users] Zap Channel   Tzafrir Cohen
• [Asterisk-Users] Configure asterisk to dial user and notify if new voicemail   Eric
• [Asterisk-Users] DTMF Diagnostics ???   clint_in_sydney
• [Asterisk-Users] Asterisk Registration as Client to OpenSER   Ozan Blotter
• [Asterisk-Users] Hotel Setup?   Kurth Bemis
• [Asterisk-Users] RxFAX   Lee Archer
• [Asterisk-Users] Hotel Setup?   BJ Weschke
• [Asterisk-Users] Asterisk Registration as Client to OpenSER   Iqbal
• [Asterisk-Users] Asterisk Registration as Client to OpenSER   Soner Tari
• [Asterisk-Users] Syslog file size   Paul
• [Asterisk-Users] Asterisk Registration as Client to OpenSER   Soner Tari
• [Asterisk-Users] Montreal asterisk usergroup meeting today 6pm   Adrien Laurent
• [Asterisk-Users] Asterisk on AMD64   Joseph
• [Asterisk-Users] Hotel Setup?   James Fogg
• [Asterisk-Users] wctdm module won't load after kernel upgrade   Dionisis Koumouras
• [Asterisk-Users] wctdm module won't load after kernel upgrade   Dave Cotton
• [Asterisk-Users] Voicemail Not Recognizing user and password?   Sherwood McGowan
• [Asterisk-Users] XML to monitor queues on Cisco display ?   Jean-Louis curty
• [Asterisk-Users] How to increase delay before incoming call answer with tdm400p   taf taffey
• [Asterisk-Users] Hotel Setup?   kurth at ozonecomputer.com
• [Asterisk-Users] Hotel Setup?   Sherwood McGowan
• [Asterisk-Users] Asterisk on AMD64   Massimo De Nadal
• [Asterisk-Users] Can't pickup inbound calls with TDM400P Fxo   taf taffey
• [Asterisk-Users] Hang up not hanging up (New Zealand Indications?? )   Marek Zachara
• [Asterisk-Users] RxFAX   Rich Adamson
• [Asterisk-Users] Hotel Setup?   kurth at ozonecomputer.com
• [Asterisk-Users] Hotel Setup?   Jonathan k. Creasy
• [Asterisk-Users] can not make call with Unicall (MFC/R2)   Denis Galvão - iSolve
• [Asterisk-Users] Asterisk on AMD64   Dave Cotton
• [Asterisk-Users] civil emergency comms: Asterisk + HAM   Mike M
• [Asterisk-Users] AGI + Ruby   Kanuri, Seshu (Company IT)
• [Asterisk-Users] Problems dialing out TDM400P - latest CVS-HEAD   Ted Serreyn
• [Asterisk-Users] Callerid UK patches (from Lusyn)   Paul Goodyear
• [Asterisk-Users] civil emergency comms: Asterisk + HAM   Mike M
• [Asterisk-Users] AGI problem with library path   Olivier Perrin
• [Asterisk-Users] Other Voicemail systems   Sherwood McGowan
• [Asterisk-Users] Asterisk on AMD64   Michael Welter
• [Asterisk-Users] Other Voicemail systems   Steve Blair
• [Asterisk-Users] Canspy listening to SIP channels   Iqbal
• [Asterisk-Users] Asterisk on AMD64   Dave Cotton
• [Asterisk-Users] Other Voicemail systems   Sherwood McGowan
• [Asterisk-Users] Callerid fails in any release after beta1 fails   John Hill
• [Asterisk-Users] [Fwd: SwissVoice IP10S not able to dial calls with protocol SIP]   makevuy
• [Asterisk-Users] Zap Channel   VoIP Newbie
• [Asterisk-Users] Other Voicemail systems   Paul
• [Asterisk-Users] Re: Asterisk-Users Digest, Vol 14, Issue 70   Nguyen Trung Tin
• [Asterisk-Users] Hotel Setup?   Tom Rymes
• [Asterisk-Users] Hotel Setup?   Michael Welter
• [Asterisk-Users] chan_zap.c:8050 pri_dchannel: Ring requested on unconfigured channel 255/255 span 2   Alex Ongena
• [Asterisk-Users] Other Voicemail systems   Sherwood McGowan
• [Asterisk-Users] callfiles: set variables ?   Bruno.Voigt at ic3s.de
• [Asterisk-Users] Zap Channel   Alex Ongena
• [Asterisk-Users] Callerid UK patches (from Lusyn)   Tony Hoyle
• [Asterisk-Users] Hotel Setup?   Jonathan k. Creasy
• [Asterisk-Users] Hotel Setup?   Chris
• AW: [Asterisk-Users] Debian Sarge, Kernel 2.6.13 and AVM Fritz!PCI v2.0 card   Rainer Maier
• [Asterisk-Users] Asterisk on AMD64   Emanuele Pucciarelli
• [Asterisk-Users] civil emergency comms: Asterisk + HAM   Mark Phillips
• [Asterisk-Users] optimizing for via C3   Tzafrir Cohen
• [Asterisk-Users] Pass through of T.38   Carlos Alperin
• [Asterisk-Users] Skype purchased by Ebay 2.6 Billion   Cory Andrews
• [Asterisk-Users] Asterisk on AMD64   canuck15
• [Asterisk-Users] Adding DNIS digits   Darren Wright
• [Asterisk-Users] Skype purchased by Ebay 2.6 Billion   Dean Collins
• [Asterisk-Users] Other Voicemail systems   Sherwood McGowan
• [Asterisk-Users] Other Voicemail systems   Rich Adamson
• [Asterisk-Users] Skype purchased by Ebay 2.6 Billion   Kanuri, Seshu (Company IT)
• [Asterisk-Users] Skype purchased by Ebay 2.6 Billion   Colin Anderson
• [Asterisk-Users] Adding DNIS digits   C F
• [Asterisk-Users] Skype purchased by Ebay 2.6 Billion   Nathan C. Smith
• [Asterisk-Users] Skype purchased by Ebay 2.6 Billion   Iqbal
• [Asterisk-Users] New York Asterisk User Group - Established   Henry Junior
• [Asterisk-Users] Skype purchased by Ebay 2.6 Billion   Kanuri, Seshu (Company IT)
• [Asterisk-Users] AGI + Ruby   joe heitzeberg
• [Asterisk-Users] Debian Sarge, Kernel 2.6.13 and AVM Fritz!PCI v2.0 card   Tzafrir Cohen
• [Asterisk-Users] ruby-agi 0.0.2 released   joe heitzeberg
• [Asterisk-Users] Skype purchased by Ebay 2.6 Billion   Lull, Rick
• [Asterisk-Users] callfiles: set variables ?   Mike Clark
• [Asterisk-Users] Adding DNIS digits   Jonathan k. Creasy
• [Asterisk-Users] optimizing for via C3   Konrads Smelkovs
• [Asterisk-Users] Skype purchased by Ebay 2.6 Billion   Colin Anderson
• [Asterisk-Users] optimizing for via C3   Tzafrir Cohen
• [Asterisk-Users] Skype purchased by Ebay 2.6 Billion   Kanuri, Seshu (Company IT)
• [Asterisk-Users] Skype purchased by Ebay 2.6 Billion   Steve Kennedy
• [Asterisk-Users] Adding DNIS digits   Darren Wright
• [Asterisk-Users] Adding DNIS digits   C F
• [Asterisk-Users] Skype purchased by Ebay 2.6 Billion   Iqbal
• [Asterisk-Users] Skype purchased by Ebay 2.6 Billion   Brian Capouch
• [Asterisk-Users] Skype purchased by Ebay 2.6 Billion   Paul
• [Asterisk-Users] Adding DNIS digits   Jonathan k. Creasy
• [Asterisk-Users] WaitExten?   Sherwood McGowan
• [Asterisk-Users] Help - Cisco router configuration with Asterisk   Paul Mahler
• [Asterisk-Users] Adding DNIS digits   Jonathan k. Creasy
• [Asterisk-Users] Skype purchased by Ebay 2.6 Billion   Kanuri, Seshu (Company IT)
• [Asterisk-Users] Adding DNIS digits   Darren Wright
• [Asterisk-Users] Skype purchased by Ebay 2.6 Billion   Jonathan k. Creasy
• [Asterisk-Users] Adding DNIS digits   C F
• [Asterisk-Users] CallerID Name in dialplan   Andy Vega
• [Asterisk-Users] Help - Cisco router configuration with Asterisk   Cory Andrews
• [Asterisk-Users] LiveVOIP - I win :)   Jay Milk
• [Asterisk-Users] PRI echo   Matt Fredrickson
• [Asterisk-Users] Skype purchased by Ebay 2.6 Billion   Jay Milk
• [Asterisk-Users] compile error with postgres and voicemail   Sebastian Kühner
• [Asterisk-Users] Adding DNIS digits   Darren Wright
• [Asterisk-Users] Debian Sarge, Kernel 2.6.13 and AVM Fritz!PCI v2.0 card   Rainer Maier
• [Asterisk-Users] Skype purchased by Ebay 2.6 Billion   Kanuri, Seshu (Company IT)
• [Asterisk-Users] OT: Online TTS engines?   Colin Anderson
• [Asterisk-Users] What have I misconfigured?   Jonathan k. Creasy
• [Asterisk-Users] CallerID Name in dialplan   Jonathan k. Creasy
• [Asterisk-Users] CallerID Name in dialplan   Colin Anderson
• [Asterisk-Users] Stupid tricks: preventable?   alan
• [Asterisk-Users] LiveVOIP - I win :)   Wiley Siler
• [Asterisk-Users] Skype purchased by Ebay 2.6 Billion   Race Vanderdecken
• [Asterisk-Users] AGI + Ruby   Dan Littlejohn
• [Asterisk-Users] Meetme Dial Out   Asterisk Supporter
• [Asterisk-Users] AGI + Ruby   joe heitzeberg
• [Asterisk-Users] Montreal asterisk usergroup meeting today 6pm   Kurth Bemis
• [Asterisk-Users] CallerID Name in dialplan   MF Hulber
• [Asterisk-Users] Skype purchased by Ebay 2.6 Billion   Steve Kennedy
• [Asterisk-Users] Hotel Setup?   Daryll Strauss
• [Asterisk-Users] Skype purchased by Ebay 2.6 Billion   Colin Anderson
• [Asterisk-Users] What have I misconfigured?   Tarpo, Louie
• [Asterisk-Users] CallerID Name in dialplan   Alexander Lopez
• [Asterisk-Users] ChanSpy   Jennifer Hales
• [Asterisk-Users] What have I misconfigured?   Jonathan k. Creasy
• [Asterisk-Users] CallerID Name in dialplan   Joe Greco
• [Asterisk-Users] early dial (grandstream bt100)   Conrad Wood
• [Asterisk-Users] Meetme Dial Out   Matt Florell
• [Asterisk-Users] monitor peak channel use   Damon Estep
• [Asterisk-Users] Whisper Mode   Alexander Lopez
• [Asterisk-Users] Firmware upgrade Aastra 480i CT   Chris Coulthurst
• [Asterisk-Users] monitor peak channel use   Alexander Lopez
• [Asterisk-Users] monitor peak channel use   Rod Bacon
• [Asterisk-Users] monitor peak channel use   Damon Estep
• [Asterisk-Users] What have I misconfigured?   Rod Bacon
• [Asterisk-Users] OT: Online TTS engines?   Jeffrey Bird
• [Asterisk-Users] What have I misconfigured?   Jonathan k. Creasy
• [Asterisk-Users] Skype purchased by Ebay 2.6 Billion   Matt
• [Asterisk-Users] What have I misconfigured?   Tarpo, Louie
• [Asterisk-Users] Subject: '#' dialplan pattern matching   Brian Gehrich
• [Asterisk-Users] Firmware upgrade Aastra 480i CT   Martin
• [Asterisk-Users] Skype purchased by Ebay 2.6 Billion   Chris HARIGA
• [Asterisk-Users] Calling from one port on a SIPURA 2002 to the other port.   Moody
• [Asterisk-Users] High system load and system freezes   Joerg Furrer
• [Asterisk-Users] ChanSpy   Alex Kauffmann
• [Asterisk-Users] PRI echo   Gabriel Gunderson
• [Asterisk-Users] CallerID Name in dialplan   Dylan VanHerpen
• [Asterisk-Users] Hotel Setup?   Steve Totaro
• [Asterisk-Users] Firmware upgrade Aastra 480i CT   canuck15
• [Asterisk-Users] Is "ChanIsAvail" thread safe?   hugolivude
• [Asterisk-Users] Is "ChanIsAvail" thread safe?   Carlos Antunes
• [Asterisk-Users] Phonecall or something as robust   Joshua Abbott
• [Asterisk-Users] get dialstatus variable when returning No such context/extension   Geoff Karl
• [Asterisk-Users] OT: Online TTS engines?   Rod Bacon
• [Asterisk-Users] Zyxel Prestige 2000W Firmware - GOOD!   Rod Bacon
• [Asterisk-Users] 2 box single Asterisk   Asterisk Sales
• [Asterisk-Users] 2 box single Asterisk   Christoph Eicke
• [Asterisk-Users] NAT and SIP.conf update.   Derek Conniffe
• [Asterisk-Users] NAT and SIP.conf update.   Jens Vagelpohl
• [Asterisk-Users] NAT and SIP.conf update.   Peter Bowyer
• [Asterisk-Users] Coexistence of zaphfc and hisax?   Stefan Helbing
• [Asterisk-Users] Hang up not hanging up (New Zealand Indications??)   Matt Riddell
• [Asterisk-Users] Skype purchased by Ebay 2.6 Billion   Steve Kennedy
• [Asterisk-Users] Meetme Question   Accursio Avona
• [Asterisk-Users] Meetme Question   Francesco Peeters
• [Asterisk-Users] NAT and SIP.conf update.   razza
• [Asterisk-Users] NAT and SIP.conf update.   Jens Vagelpohl
• [Asterisk-Users] NAT and SIP.conf update.   Derek Conniffe
• [Asterisk-Users] Skype purchased by Ebay 2.6 Billion   Anders Svensson
• [Asterisk-Users] NAT and SIP.conf update.   razza
• [Asterisk-Users] Zaptel lines locking up   Rick
• [Asterisk-Users] OT: Online TTS engines?   Bob Goddard
• [Asterisk-Users] OT: Online TTS engines?   razza
• [Asterisk-Users] Zap Channel   VoIP Newbie
• [Asterisk-Users] Nat & Sip & Pain   Derek Conniffe
• [Asterisk-Users] Monitoring status of ISDN lines   Enzo Michelangeli
• [Asterisk-Users] Skype purchased by Ebay 2.6 Billion   Matt
• [Asterisk-Users] NAT and SIP.conf update.   Wilson Pickett
• [Asterisk-Users] compile error with postgres and voicemail   Sebastian Kühner
• [Asterisk-Users] show queue callcenter output?   Rajkumar S
• [Asterisk-Users] Nat & Sip & Pain   razza
• [Asterisk-Users] Integration between Asterisk and Siemens HiCom 150e over ISDN   Simon D
• [Asterisk-Users] Nat & Sip & Pain   chentschel at arnet.com.ar
• [Asterisk-Users] PLEASE HELP!! CALLERID FAILS!!   John Hill
• [Re: [Asterisk-Users] civil emergency comms: Asterisk + HAM]]   Derek Whitten
• [Asterisk-Users] Monitoring status of ISDN lines   Matteo Brancaleoni
• [Asterisk-Users] OT: Online TTS engines?   Colin Anderson
• [Asterisk-Users] Micro-cuts in MusicOnHold   Yoann Le Bihan
• [Asterisk-Users] SetCIDName question   DRi at b-w-computer.de
• [Asterisk-Users] Pass through of T.38   Matthew Boehm
• [Asterisk-Users] SetCIDName question   Derek Whitten
• [Asterisk-Users] Integration between Asterisk and Siemens HiCom150e over ISDN   Sander
• [Asterisk-Users] passing variables to h extension   Simone Cittadini
• [Asterisk-Users] Real-time Linux claims single-digit microsecond responsiveness   Carlos Antunes
• [Asterisk-Users] Bristuff version for use with 1.2.0beta1   Karsten Wemheuer
• [Asterisk-Users] SetCIDName question   DRi at b-w-computer.de
• [Asterisk-Users] asterisk callerid problems   Jimmy Smith
• [Asterisk-Users] problem with FXS module   Innocent Evil
• [Asterisk-Users] actionID on manager events   Michael George
• [Asterisk-Users] PRI echo   Matt Fredrickson
• [Asterisk-Users] TDMoE Configuration problems   Kevin Bockman
• [Asterisk-Users] NAT and SIP.conf update.   Michiel van Baak
• [Asterisk-Users] NAT and SIP.conf update.   canuck15
• [Asterisk-Users] Re: passing variables to h extension   Tony Mountifield
• [Asterisk-Users] rotate * log file?   Brian C. Fertig
• [Asterisk-Users] rotate * log file?   Michiel van Baak
• [Asterisk-Users] Nat & Sip & Pain   Derek Conniffe
• [Asterisk-Users] Meetme Question   Accursio Avona
• FW: [Asterisk-Users] Nat & Sip & Pain   Derek Conniffe
• [Asterisk-Users] NAT and SIP.conf update.   Derek Conniffe
• [Asterisk-Users] rotate * log file?   Kevin Bockman
• [Asterisk-Users] NAT and SIP.conf update.   Mark Phillips
• [Asterisk-Users] Meetme Question   Doug Lytle
• [Asterisk-Users] How to create IVR menu and transfer to another sip extensions.   PJ Santos
• [Asterisk-Users] passing variables to h extension   Gary Reuter
• [Asterisk-Users] What have I misconfigured?   Jonathan k. Creasy
• [Asterisk-Users] Can anyone explain why this is happening? Odd CUT Problem   Matt
• [Asterisk-Users] asterisk hangup detection on a pbx analog port]   Brian Kennedy
• [Asterisk-Users] NAT and SIP.conf update.   razza
• [Asterisk-Users] asterisk hangup detection on a pbx analog port]   BJ Weschke
• [Asterisk-Users] translate letters into digits   Armin Schindler
• [Asterisk-Users] Asterisk + NEC IPK 192 integration   Dustin Rue
• [Asterisk-Users] Using RedirectAction with queues   Morten Isaksen
• [Asterisk-Users] NAT and SIP.conf update.   Michiel van Baak
• [Asterisk-Users] translate letters into digits   Tom Hayden
• [Asterisk-Users] Polycom IP500 Mass Configurations   Cody Lerum
• [Asterisk-Users] Polycom IP500 Mass Configurations   Kevin P. Fleming
• [Asterisk-Users] AMP created extensions busy when dialed.   Vamsi Pottangi
• [Asterisk-Users] Fedora Core 4 not recognizing X100P cards   Shawn Porter
• [Asterisk-Users] translate letters into digits   Armin Schindler
• [Asterisk-Users] ZoomTel x5v Model 5565: is it any good?   Jean-Michel Hiver
• [Asterisk-Users] Not able to access asterisk from internet via ip-forwarding   Vamsi Pottangi
• [Asterisk-Users] Fedora Core 4 not recognizing X100P cards   Mojo with Horan & Company, LLC
• [Asterisk-Users] Not able to access asterisk from internet via ip-forwarding   Mojo with Horan & Company, LLC
• [Asterisk-Users] Not able to access asterisk from internet via ip-forwarding   Mojo with Horan & Company, LLC
• [Asterisk-Users] Re: actionID on manager events   Tony Mountifield
• [Asterisk-Users] Fedora Core 4 not recognizing X100P cards   Ing CIP Alejandro Celi Mariátegui
• [Asterisk-Users] Stumped on vMail problem, any ideas?   Kevin Hanson
• [Asterisk-Users] disable chan_skinny and chan_oss   Joseph
• [Asterisk-Users] Fedora Core 4 not recognizing X100P cards   Carlos Chavez
• [Asterisk-Users] Fedora Core 4 not recognizing X100P cards   Innocent Evil
• [Asterisk-Users] Dialplan Design Q   info at beprojects.com
• [Asterisk-Users] Integration Nortel x Asterisk   PJ Santos
• [Asterisk-Users] Oh323 and Asterisk with MERA   Sahil Gupta
• [Asterisk-Users] TDM400P stops answering   Andy Howell
• [Asterisk-Users] sometimes dtmf passed, sometimes not (cisco 7960 SIP)   Mat Stace, Colewood
• [Asterisk-Users] TDM400P stops answering   Michael Welter
• [Asterisk-Users] disable chan_skinny and chan_oss   Domjan Attila
• [Asterisk-Users] MTA V102   Juan Salas
• [Asterisk-Users] First PRI Installed - WOOT   Wiley Siler
• [Asterisk-Users] Cisco AS5400 Configuration as a SIP Peer - URGENT   Matt Roth
• [Asterisk-Users] How to IGNORE distinctive ring   Brad Jacobs
• [Asterisk-Users] TDM400P stops answering   Kevin P. Fleming
• [Asterisk-Users] callfiles: how to query current dial attempt nr in extensions.conf?   Bruno Voigt
• [Asterisk-Users] make * listen on a specific ethernet interface   asterisk at gatheringpoint.net
• [Asterisk-Users] callfile: How to invoke SetCallerPres ?   Bruno Voigt
• [Asterisk-Users] TDM400P stops answering   Reid Forrest
• [Asterisk-Users] How to IGNORE distinctive ring   Shawn Porter
• [Asterisk-Users] TDM400P stops answering   Leonardo Gomes Figueira
• [Asterisk-Users] OT: Online TTS engines?   Jeffrey Bird
• [Asterisk-Users] Fedora Core 4 not recognizing X100P cards   YT Lim
• [Asterisk-Users] CVS vs CVS-HEAD   Joseph
• [Asterisk-Users] make * listen on a specific ethernet interface   Carlos Alperin
• [Asterisk-Users] How to IGNORE distinctive ring   Brad Jacobs
• [Asterisk-Users] populating asterisk realtime tables from config files   Jonathan k. Creasy
• [Asterisk-Users] Fedora Core 4 not recognizing X100P cards   Mojo with Horan & Company, LLC
• [Asterisk-Users] Cisco AS5400 Configuration as a SIP Peer - URGENT   Leandro Tenorio
• [Asterisk-Users] populating asterisk realtime tables from configfiles   Jonathan k. Creasy
• [Asterisk-Users] Asterisk@home with Eyebeam   Dinesh
• [Asterisk-Users] Fedora Core 4 not recognizing X100P cards   Innocent Evil
• [Asterisk-Users] wctdm, issue w/outbound calls   Sean Hull
• [Asterisk-Users] OT: Online TTS engines?   Paul
• [Asterisk-Users] Digium Cards in Australia   rudolfl at optusnet.com.au
• [Asterisk-Users] Re: wctdm, issue w/outbound calls (additional notes)   Sean Hull
• [Asterisk-Users] Digium Cards in Australia   Callum McGillivray
• [Asterisk-Users] Call Wrapup time for agents.   Callum McGillivray
• [Asterisk-Users] PRI zap channels not cleared when no match in context for dialed number on inbound call   Damon Estep
• [Asterisk-Users] PRI zap channels not cleared when no match incontext for dialed number on inbound call   Alexander Lopez
• [Asterisk-Users] PRI zap channels not cleared when no matchincontext for dialed number on inbound call   Damon Estep
• [Asterisk-Users] PRI zap channels not cleared whennomatchincontext for dialed number on inbound call   Damon Estep
• [Asterisk-Users] PRI zap channels not cleared when nomatchincontext for dialed number on inbound call   Alexander Lopez
• [Asterisk-Users] Call Wrapup time for agents.   Mark Phillips
• [Asterisk-Users] PRI zap channels not cleared whennomatchincontext for dialed number on inbound call   Damon Estep
• [Asterisk-Users] Digium Cards in Australia   Boris Bakchiev
• [Asterisk-Users] Call Wrapup time for agents.   Alexander Lopez
• [Asterisk-Users] Call Wrapup time for agents.   Kevin P. Fleming
• [Asterisk-Users] Call Wrapup time for agents.   Alexander Lopez
• [Asterisk-Users] slight echo via sip provider   Damon Estep
• [Asterisk-Users] Limiting call minutes on a GSM SIM   Remco Barende
• [Asterisk-Users] Zap Clocking - Frame Slips - tdm400p wcfxozttest cpu spikes spandsp   Boris Bakchiev
• [Asterisk-Users] Limiting call minutes on a GSM SIM   trixter http://www.0xdecafbad.com
• [Asterisk-Users] Call Wrapup time for agents.   Callum McGillivray
• [Asterisk-Users] spandsp frame slip tolerance.   Boris Bakchiev
• [Asterisk-Users] Anyone knows how to receive a SIP call without registering gateway?   C. Savinovich
• [Asterisk-Users] slight echo via sip provider   Florian Overkamp
• [Asterisk-Users] Anyone knows how to receive a SIP call without registering gateway?   BJ Weschke
• [Asterisk-Users] Callerid fails in any release after beta1 fails   Richard Kashdan
• [Asterisk-Users] Zyxel Prestige 2000W Firmware - GOOD!   Paul Hales
• [Asterisk-Users] 2 box single Asterisk   David Phelan
• [Asterisk-Users] Digium Cards in Australia   Paul Hales
• [Asterisk-Users] Limiting call minutes on a GSM SIM   Remco Barende
• [Asterisk-Users] Limiting call minutes on a GSM SIM   trixter http://www.0xdecafbad.com
• [Asterisk-Users] pri release cause code mismatch   Tirpák Miklós
• [Asterisk-Users] callfile: How to invoke SetCallerPres ?   Steve Hanselman
• [Asterisk-Users] oh323 and Asterisk: Calls always hang up   Hauke Zuehl
• [Asterisk-Users] call restrictions   Erdem HAKİ
• [Asterisk-Users] SetCIDName question   DRi at b-w-computer.de
• [Asterisk-Users] call restrictions   Christoph Eicke
• [Asterisk-Users] Re: passing variables to h extension   Simone Cittadini
• [Asterisk-Users] Callerid fails in any release after beta1 fails   Doug Lytle
• [Asterisk-Users] Anyone knows how to receive a SIP call withoutregistering gateway?   Enzo Michelangeli
• [Asterisk-Users] sometimes dtmf passed, sometimes not (cisco 7960 SIP)   Mat Stace, Colewood Internet
• [Asterisk-Users] T.38 ATA   Nenad Radosavljevic
• [Asterisk-Users] GotoIf Syntax to match first digits   ChB
• [Asterisk-Users] NAT and SIP.conf update.   Dave Cotton
• [Asterisk-Users] Call Wrapup time for agents.   lenz
• [Asterisk-Users] GotoIf Syntax to match first digits   DRi at b-w-computer.de
• [Asterisk-Users] Sipura Registration time out, no incoming calls   Zeeshan
• [Asterisk-Users] populating asterisk realtime tables from configfiles   Matt
• [Asterisk-Users] Sipura Registration time out, no incoming calls   Matt
• [Asterisk-Users] T.38 ATA   Chris Mason (Lists)
• [Asterisk-Users] T.38 ATA   Moody
• [Asterisk-Users] T.38 ATA   Ivan Meic (Vox Mundi)
• [Asterisk-Users] T.38 ATA   Anders Svensson
• [Asterisk-Users] TDMoE Configuration problems   Leonardo Gomes Figueira
• [Asterisk-Users] PRI to PRI passthrough with DID intact   Steven
• [Asterisk-Users] STUN vs NAT Helper   Waldo Rubinstein
• [Asterisk-Users] Dial Application Return Codes - Help needed   Mark Edwards
• [Asterisk-Users] Meetme Question   Accursio Avona
• [Asterisk-Users] MAX PRI for single server (was: Not enoughlinesavailable for Asterisk implemetation)   Troy Settle
• [Asterisk-Users] T.38 ATA   Rosario Pingaro
• [Asterisk-Users] STUN vs NAT Helper   chentschel at arnet.com.ar
• [Asterisk-Users] GotoIf Syntax to match first digits   ChB
• [Asterisk-Users] Callerid fails in any release after beta1 fails   John Hill
• [Asterisk-Users] first character in line 11 missing   Paul Hewlett
• [Asterisk-Users] STUN vs NAT Helper   Derek Conniffe
• [Asterisk-Users] Asterisk@home with Eyebeam   steve at daviesfam.org
• [Asterisk-Users] (no subject)   Christoph Eicke
• [Asterisk-Users] (no subject)   Matt Ryanczak
• [Asterisk-Users] Realtime IAX   Dana Olson
• [Asterisk-Users] MAX PRI for single server (was:Not enoughlinesavailable for Asterisk implemetation)   Damon Estep
• [Asterisk-Users] timeout with queue   Wolfgang Lumpp
• [Asterisk-Users] Fedora Core 4 not recognizing X100P cards   Innocent Evil
• [Asterisk-Users] (no subject)   Pablo Allietti
• [Asterisk-Users] pri release cause code mismatch   Johann Steinwendtner
• [Asterisk-Users] SMS using a PRI channel   Roger Schreiter
• [Asterisk-Users] Asterisk 1.0.9 long term stability   Sig Lange
• [Asterisk-Users] IAX Registration with servers   Naren Koka
• [Asterisk-Users] IAX Registration with servers   Dave Cotton
• [Asterisk-Users] Callerid fails in any release after beta1 fails   Doug Lytle
• [Asterisk-Users] R1.502 of chan_zap.c kills callerid on a x101p   John Hill
• [Asterisk-Users] Re: (no subject)   Pablo Allietti
• [Asterisk-Users] TDM400P stops answering   Andy Howell
• [Asterisk-Users] TDM400P stops answering   Andy Howell
• [Asterisk-Users] Asterisk as a gateway. 'flash for transfers transparency?'   felipe hangen
• [Asterisk-Users] Fedora Core 4 not recognizing X100P cards   Tzafrir Cohen
• [Asterisk-Users] TE110P - Asterisk@Home Install Problems   Robert Wagner
• [Asterisk-Users] Dialplan Design Q   Moises Silva
• [Asterisk-Users] TDM400P stops answering   Kevin P. Fleming
• [Asterisk-Users] How to create IVR menu and transfer to another sip extensions.   Moises Silva
• [Asterisk-Users] T.38 ATA   Enzo Michelangeli
• [Asterisk-Users] Asterisk 1.0.9 long term stability   info at beprojects.com
• [Asterisk-Users] How to create IVR menu and transfer to another sip extensions.   PJ Santos
• [Asterisk-Users] Re: T.38 ATA   Nenad Radosavljevic
• [Asterisk-Users] How to create IVR menu and transfer to anothersip extensions.   Anthony Rodgers
• [Asterisk-Users] Zyxel Prestige 2000W Firmware - GOOD!   Mojo with Horan & Company, LLC
• [Asterisk-Users] Asterisk 1.0.9 long term stability <--thread hijack, why not reboot?   Colin Anderson
• [Asterisk-Users] RxFax problems.   Arne Morten Johansen
• [Asterisk-Users] Asterisk 1.0.9 long term stability   Carlos Alperin
• [Asterisk-Users] Anyone knows how to receive a SIP call withoutregistering gateway?   Damon Estep
• RES: [Asterisk-Users] How to create IVR menu and transfer to anothersip extensions.   Fábio Sakai
• [Asterisk-Users] Polycom randomly fails outbound calls,   Andres Paglayan
• [Asterisk-Users] Re: Asterisk 1.0.9 long term stability   Noah Miller
• [Asterisk-Users] How to create IVR menu and transfer to anothersip extensions.   Sander
• [Asterisk-Users] timeout with queue   Sander
• [Asterisk-Users] RxFax problems   Arne Morten Johansen
• [Asterisk-Users] Stupid tricks: preventable?   Colin Anderson
• [Asterisk-Users] Re: T.38 ATA   Rosario Pingaro
• [Asterisk-Users] Re: Polycom randomly fails outbound calls,   Noah Miller
• [Asterisk-Users] RxFax problems   Dave Cotton
• [Asterisk-Users] Re: Asterisk 1.0.9 long term stability   Colin Anderson
• [Asterisk-Users] Re: (no subject)   Sander
• [Asterisk-Users] Indications for Ireland   Sean Rima
• [Asterisk-Users] Asterisk 1.0.9 long term stability <--thread hijack, why not reboot?   Matt
• [Asterisk-Users] Asterisk Consulting Project ISO Hired Gun   Cory Andrews
• [Asterisk-Users] RxFax/TxFax - Compile Problem   David Sampson
• [Asterisk-Users] Re: Asterisk 1.0.9 long term stability   Tony Mountifield
• [Asterisk-Users] sox conversion has introduces background hiss for both 8k and 41K recordings to gsm   Jerry Geis
• [Asterisk-Users] Re: TE110P - Asterisk@Home Install Problems   Pablo Allietti
• [Asterisk-Users] Asterisk Consulting Project ISO Hired Gun   Alexander Lopez
• [Asterisk-Users] ASTCC issues   Michael K. Rodriguez
• [Asterisk-Users] Echo on SPA-3000 FXO   Paul Dugas
• [Asterisk-Users] How to create IVR menu and transfer to anothersip extensions.   PJ Santos
• [Asterisk-Users] RxFax/TxFax - Compile Problem   BJ Weschke
• [Asterisk-Users] Re: Polycom randomly fails outbound calls,   Andres Paglayan
• [Asterisk-Users] ${VM_CIDNUM} shows up but ${VM_CALLERID} & ${VM_CIDNAME} don't?   Doug
• [Asterisk-Users] Re: (no subject)   Pablo Allietti
• [Asterisk-Users] Asterisk 1.0.9 long term stability <--threadhijack, why not reboot?   canuck15
• [Asterisk-Users] Cisco AS5400 Configuration as a SIP Peer - URGENT   Matt Roth
• [Asterisk-Users] Re: Asterisk 1.0.9 long term stability   canuck15
• [Asterisk-Users] TDM400P stops answering   Chris Stone
• [Asterisk-Users] Echo on SPA-3000 FXO   pcman theMan
• [Asterisk-Users] # dialplan not working...   Scott
• [Asterisk-Users] Compile error on cdr_yada for asterisk on centos with Oracle   Han van Hulst
• [Asterisk-Users] Re: Asterisk 1.0.9 long term stability   info at beprojects.com
• [Asterisk-Users] Echo on SPA-3000 FXO   Chris Mason (Lists)
• [Asterisk-Users] g729 to asterisk to g729 voip provider   Erick Perez
• [Asterisk-Users] RE: Asterisk-Users Digest, Vol 14, Issue 86   support at sjobeck.com
• [Asterisk-Users] Re: Polycom randomly fails outbound calls,   Andres Paglayan
• [Asterisk-Users] Routes IPSEc And Asterisk.   Carlos Arnt
• [Asterisk-Users] PRI to PRI passthrough with DID intact   C F
• [Asterisk-Users] Interop with Cisco T1/PRI on the 2811 and PSTN   David Zhi Wu
• [Asterisk-Users] Re: Newbie needs advice   Paul
• [Asterisk-Users] Sipura Registration time out, no incoming calls   Zeeshan
• [Asterisk-Users] Weird SIP behavior or I need a shrink?   Federico Alves
• [Asterisk-Users] I connected my quicknet phonejack to the wall phone outlet and .......   cmisip
• [Asterisk-Users] How to uninstall   Technical Support
• [Asterisk-Users] Echo on SPA-3000 FXO   Neil Cherry
• [Asterisk-Users] Asterisk 1.0.9 long term stability<--threadhijack, why not reboot?   William Boehlke
• [Asterisk-Users] Distinctive Ring Tones   Howard Lowndes
• [Asterisk-Users] Voice recognition   Ed Dowding
• [Asterisk-Users] Asterisk 1.0.9 long term stability <--threadhijack, why not reboot?   John Novack
• [Asterisk-Users] Liquidation: Cisco; Polycom; D-Link; MediaTrix, Colubris - Highly Reduced Prices   Justin Newman
• [Asterisk-Users] Liquidation: Cisco; Polycom; D-Link; MediaTrix, Colubris - Highly Reduced Prices   Paul
• [Asterisk-Users] Voice recognition   Dean Collins
• [Asterisk-Users] Cannot hear teleco side error message   Xue Liangliang
• [Asterisk-Users] Starting From Scratch   Thczv F. Thczv
• [Asterisk-Users] RXFax   Chris
• [Asterisk-Users] Re: (no subject)   Sander
• [Asterisk-Users] Fedora Core 4 not recognizing X100P cards   Innocent Evil
• [Asterisk-Users] Starting From Scratch   Sander
• [Asterisk-Users] compile problems with yada   Han van Hulst
• [Asterisk-Users] Routes IPSEc And Asterisk.   Peter Svensson
• [Asterisk-Users] asterisk frequently dead   stevanus
• [Asterisk-Users] timeout with queue   Wolfgang Lumpp
• [Asterisk-Users] pri release cause code mismatch   Tirpák Miklós
• [Asterisk-Users] problem with FXS module   gincantalupo
• SV: [Asterisk-Users] RxFax problems   Arne Morten Johansen
• [Asterisk-Users] RXFax   Lee Archer
• [Asterisk-Users] Originate not understanding 2 vars in setvars   Joerg Lauer
• [Asterisk-Users] Indications for Ireland   Ronan Mullally
• ***SPAM*** [Asterisk-Users] actionID on manager events   Joerg Lauer
• [Asterisk-Users] Using RedirectAction with queues   Josip Gracin
• [Asterisk-Users] ${DIALSTATUS} problems   Mark Edwards
• AW: ***SPAM*** [Asterisk-Users] actionID on manager events   Anton Kostanjsek
• [Asterisk-Users] Re: Originate not understanding 2 vars in setvars   Tony Mountifield
• [Asterisk-Users] Re: ${DIALSTATUS} problems   Tony Mountifield
• [Asterisk-Users] Incoming / Outgoing call problems on TDM card.   essay essa
• [Asterisk-Users] PSTN calls are quiet   Paul Goodyear
• [Asterisk-Users] iax phone and asterisk server on different LANs   gincantalupo
• [Asterisk-Users] pri release cause code mismatch   Johann Steinwendtner
• [Asterisk-Users] PSTN calls are quiet   garth at bitco.co.za
• [Asterisk-Users] SIP rogue channel   Bart van Daal
• [Asterisk-Users] USB ISDN   Derek Conniffe
• [Asterisk-Users] PSTN calls are quiet   Paul Goodyear
• [Asterisk-Users] Echo on SPA-3000 FXO   Chris Mason (Lists)
• [Asterisk-Users] TxFAX don't work   Michele "O-Zone" Pinassi
• [Asterisk-Users] No sounds on Playback()   rpagquil at philonline.com
• [Asterisk-Users] Why isn't 3-way calling a standard feature?   Geoffrey Cleaves
• [Asterisk-Users] iax phone and asterisk server on different LANs   Emanuele Pucciarelli
• [Asterisk-Users] Looking for China DID   Steve Ducat
• [Asterisk-Users] TE110P - Asterisk@Home Install Problems - Televantage 3 T1   Robert Wagner
• [Asterisk-Users] MusicOnHold not working   Gurminder Arora
• [Asterisk-Users] MusicOnHold not working   Sherwood McGowan
• [Asterisk-Users] linux kernel tweaking for Asterisk   Brad Borgald
• [Asterisk-Users] Echo on SPA-3000 FXO   Paul Dugas
• [Asterisk-Users] Echo on SPA-3000 FXO   asterisk at txpe.net
• [Asterisk-Users] cdr server   Altus Snyman
• [Asterisk-Users] asterisk frequently dead   Moises Silva
• [Asterisk-Users] cdr server   Emanuele Pucciarelli
• [Asterisk-Users] Echo on SPA-3000 FXO   BJ Weschke
• [Asterisk-Users] cdr server   Ed Greenberg
• [Asterisk-Users] PSTN calls are quiet   garth at bitco.co.za
• [Asterisk-Users] Echo on SPA-3000 FXO   asterisk at txpe.net
• [Asterisk-Users] dialplan to try VOIP providers if they can't terminate call   Geoff Karl
• [Asterisk-Users] Asterisk don't start   monte83 at libero.it
• [Asterisk-Users] Grandstream HandyTone 386   Ugur GUNCER
• [Asterisk-Users] Asterisk don't start   Christoph Eicke
• [Asterisk-Users] Asterisk CDR information into Oracle DB   Han van Hulst
• [Asterisk-Users] USB ISDN (OT question)   Jörg Wolf
• [Asterisk-Users] Asterisk 1.0.9 long term stability <--thread hijack, why not reboot?   Andrew Kohlsmith
• [Asterisk-Users] Don't install asterisk-chan-capi   monte83 at libero.it
• [Asterisk-Users] Configuring GR303 trunks from Asterisk to a Taqua/TEKELEC T7000   Paul Conn
• [Asterisk-Users] Echo on SPA-3000 FXO   BJ Weschke
• [Asterisk-Users] Asterisk CDR information into Oracle DB   Han van Hulst
• [Asterisk-Users] Fax->Email for Hosted PBX   Michael Welter
• [Asterisk-Users] Echo on SPA-3000 FXO   Nathan C. Smith
• [Asterisk-Users] Getting email of voicemail to work   Matt Love
• [Asterisk-Users] Transfering from a device to a queue crashes Asterisk   David F. Bakker
• [Asterisk-Users] Asterisk 1.0.9 long term stability <--thread hijack, why not reboot?   Paul
• [Asterisk-Users] Getting email of voicemail to work   Colin Anderson
• [Asterisk-Users] Fax->Email for Hosted PBX   Alexander Lopez
• [Asterisk-Users] Siemens Hi-Path help   Pablo Allietti
• [Asterisk-Users] Comfort Noise Generation with Zap-IAX   Henry Jensen
• [Asterisk-Users] Asterisk 1.0.9 long term stability <--thread hijack, why not reboot?   Colin Anderson
• [Asterisk-Users] Seperate Incoming calls on TDM02?   C. Hatton Humphrey
• [Asterisk-Users] RE: Stupid tricks: preventable?   alan
• [Asterisk-Users] Asterisk 1.0.9 long term stability <--thread hijack, why not reboot?   Andrew Kohlsmith
• [Asterisk-Users] RxFax/TxFax - Compile Problem   David Sampson
• [Asterisk-Users] Fax->Email for Hosted PBX   Colin Anderson
• [Asterisk-Users] MusicOnHold not working   Alex Kobalto
• [Asterisk-Users] Asterisk CDR information into Oracle DB   Michiel van Baak
• [Asterisk-Users] Fax->Email for Hosted PBX   Tom Hayden
• [Asterisk-Users] Polycom oddities: Mixed up digits -> *8 Call Pickup   Doug
• [Asterisk-Users] Asterisk 1.0.9 long term stability <--thread hijack, why not reboot?   Tzafrir Cohen
• [Asterisk-Users] Don't install asterisk-chan-capi   Tzafrir Cohen
• [Asterisk-Users] Can not get realtime static voicemail.conf to work   Matt
• [Asterisk-Users] Oh323 and Asterisk with MERA   Kevin P. Fleming
• [Asterisk-Users] Seperate Incoming calls on TDM02?   Matt Fredrickson
• [Asterisk-Users] Asterisk 1.0.9 long term stability <--thread hijack, why not reboot?   Colin Anderson
• [Asterisk-Users] internet connection between Africa and Europe   Stéphane LAVRI
• [Asterisk-Users] internet connection between Africa and Europe   Sahil Gupta
• [Asterisk-Users] Seperate Incoming calls on TDM02?   Tom Rymes
• [Asterisk-Users] Phonecall or something as robust   Dustin Wildes
• [Asterisk-Users] Call Pickup between ZAP and SIP technologies   Angel R. Diaz
• [Asterisk-Users] If call fails, then try again with something else   Eric
• [Asterisk-Users] dialing sip before answering pstn line   bman at host.sk
• [Asterisk-Users] Can not get realtime static voicemail.conf to work   Matt Gibson
• [Asterisk-Users] internet connection between Africa and Europe   Jean-Michel Hiver
• [Asterisk-Users] Unable to call some numbers with I4L   Massimo Frisoni
• [Asterisk-Users] internet connection between Africa and Europe   Stefan de Konink
• [Asterisk-Users] internet connection between Africa and Europe   Anders Svensson
• [Asterisk-Users] Asterisk 1.0.9 long term stability <--thread hijack, why not reboot?   Paul
• [Asterisk-Users] Unable to call some numbers with I4L   Emanuele Pucciarelli
• [Asterisk-Users] RxFax/TxFax - Compile Problem   David Sampson
• [Asterisk-Users] Still having hangup problems in NZ   Simon
• [Asterisk-Users] Faxibility in NZ   Simon
• [Asterisk-Users] USB ISDN (OT question)   Derek Conniffe
• [Asterisk-Users] Indications for Ireland   Sean Rima
• [Asterisk-Users] Caller ID for auto outgoing calls   Jim Gottlieb
• [Asterisk-Users] Console/dsp and mplayer   Jerry Geis
• [Asterisk-Users] Asterisk 1.0.9 long term stability <--threadhijack, why not reboot?   Chris St Denis
• [Asterisk-Users] Caller ID for auto outgoing calls   Colin Anderson
• [Asterisk-Users] Seperate Incoming calls on TDM02?   Tom Rymes
• [Asterisk-Users] Asterisk 1.0.9 long term stability <--thread hijack, why not reboot?   Colin Anderson
• [Asterisk-Users] Asterisk CDRs   Prakash Rao Kanthi
• [Asterisk-Users] Asterisk CDRs   William Boehlke
• [Asterisk-Users] Asterisk 1.0.9 long term stability <--thread hijack, why not reboot?   Michael Loftis
• [Asterisk-Users] Asterisk CDRs   Prakash Rao Kanthi
• [Asterisk-Users] Asterisk CDRs   Nathan Pralle
• [Asterisk-Users] Asterisk 1.0.9 long term stability <--thread hijack, why not reboot?   Michael Loftis
• [Asterisk-Users] Asterisk CDRs   Prakash Rao Kanthi
• [Asterisk-Users] USB Phones for use with Asterisk   Callum McGillivray
• [Asterisk-Users] ZyXEL P662HW / SIP / Crashing   asterisk-users
• [Asterisk-Users] ZyXEL P662HW / SIP / Crashing   Julio Arruda
• [Asterisk-Users] Still having hangup problems in NZ   Matt Riddell
• [Asterisk-Users] Faxibility in NZ   Matt Riddell
• [Asterisk-Users] USB Phones for use with Asterisk   Matt Riddell
• [Asterisk-Users] SIP reinvite asterisk and NAT   Damon Estep
• [Asterisk-Users] USB Phones for use with Asterisk   Bill McCready (PCPhoneline.com)
• [Asterisk-Users] Echo on SPA-3000 FXO   Paul Dugas
• [Asterisk-Users] Asterisk and Zyxel Prestige 2000W_v2   Klaus Sonnenleiter
• [Asterisk-Users] triggering automatic dial-outs with Zap interface   M Sharif
• [Asterisk-Users] SIP reinvite asterisk and NAT   Mark Phillips
• [Asterisk-Users] Is digium supporting new te405p and te406p install?   Jason Kim
• [Asterisk-Users] SIP reinvite asterisk and NAT   Damon Estep
• [Asterisk-Users] Is digium supporting new te405p and te406p install?   Jason Walker
• [Asterisk-Users] Is digium supporting new te405p and te406p install?   Andrew Kohlsmith
• [Asterisk-Users] Sip recording   Lakmal
• [Asterisk-Users] Is digium supporting new te405p and te406p install?   Jason Kim
• [Asterisk-Users] Is digium supporting new te405p and te406pinstall?   Jason Walker
• [Asterisk-Users] QUESTION: RINGING CONTINUES DURING CALL   Alvin Austin
• [Asterisk-Users] Is digium supporting new te405p and te406pinstall?   Jason Kim
• [Asterisk-Users] SIP reinvite asterisk and NAT   Jason Walker
• [Asterisk-Users] Send SIP NOTIFY frequency   Adrian A
• [Asterisk-Users] Changing the sip port in sip.conf does not work   Federico Alves
• [Asterisk-Users] SIP reinvite asterisk and NAT   BJ Weschke
• [Asterisk-Users] Call recording between SIP phones   Lakmal
• [Asterisk-Users] Call recording between SIP phones   Kevin Bockman
• [Asterisk-Users] Help on RealTime Extensions on Oracle DB   Chris Deserva
• [Asterisk-Users] Transfering from a device to a queue crashesAsterisk   Jörg Wolf
• [Asterisk-Users] RxFax/TxFax - Compile Problem   Dave Cotton
• [Asterisk-Users] Asterisk and Zyxel Prestige 2000W_v2   Ugur GUNCER
• [Asterisk-Users] auto restart   Chee Foong
• [Asterisk-Users] Wildcard TE110P   amer karim
• [Asterisk-Users] Unable to create ZAP channel - All circuits are busy   Matt Love
• [Asterisk-Users] Call Forward - 7940 Asterisk - Help   c waddy
• [Asterisk-Users] MusicOnHold not working   Gurminder Arora
• use of automon sequence (was Re: [Asterisk-Users] Call recording between SIP phones)   John covici
• [Asterisk-Users] How to suppress Local/Zombie channels?   Anton Kostanjsek
• [Asterisk-Users] timeout with queue SOLVED   Wolfgang Lumpp
• [Asterisk-Users] USB Phones for use with Asterisk   c waddy
• [Asterisk-Users] Call Forward - 7940 Asterisk - Help   Stefan Gofferje
• [Asterisk-Users] Call Forward - 7940 Asterisk - Help   Derek Conniffe
• [Asterisk-Users] Call recording between SIP phones   Steve Totaro
• [Asterisk-Users] 7 digit dialing to e.164 format   Matt Schulte
• [Asterisk-Users] Is digium supporting new te405p and te406p install?   Andrew Kohlsmith
• [Asterisk-Users] Unable to call some numbers with I4L   Massimo Frisoni
• [Asterisk-Users] Asterisk don't start   monte83 at libero.it
• [Asterisk-Users] Help on RealTime Extensions on Oracle DB   Juan Salas
• [Asterisk-Users] RxFax/TxFax - Compile Problem   David Sampson
• [Asterisk-Users] Unable to call some numbers with I4L   Armin Schindler
• [Asterisk-Users] Call Forward - 7940 Asterisk - Help   Stefan Gofferje
• [Asterisk-Users] Call Forward - 7940 Asterisk - Help   c waddy
• [Asterisk-Users] broadvoice incoming caller ID is wierd when calling from voipjet   Paul
• [Asterisk-Users] Extension Restrictions   Waldo Rubinstein
• [Asterisk-Users] Seperate Incoming calls on TDM02?   C. Hatton Humphrey
• [Asterisk-Users] Asterisk don't start   monte83 at libero.it
• [Asterisk-Users] Asterisk don't start   monte83 at libero.it
• [Asterisk-Users] Caller Name: Asterisk reading too fast   J Thomas
• [Asterisk-Users] USB Phones for use with Asterisk   Kanuri, Seshu (Company IT)
• [Asterisk-Users] Caller Name: Asterisk reading too fast   Juan Jose Comellas
• [Asterisk-Users] SIP port assignment for user agents registering to Asterisk.   Steve Lane
• use of automon sequence (was Re: [Asterisk-Users] Call recording between SIP phones)   Kevin Bockman
• [Asterisk-Users] Seperate Incoming calls on TDM02?   C. Hatton Humphrey
• [Asterisk-Users] 7 digit dialing to e.164 format   Jonathan k. Creasy
• [Asterisk-Users] RxFax/TxFax - Compile Problem   Alexander Lopez
• [Asterisk-Users] queue_log on mysql   lenz
• [Asterisk-Users] alsa issue with asound.conf   Jerry Geis
• [Asterisk-Users] Asterisk don't start   monte83 at libero.it
• [Asterisk-Users] direct sip call pickup   Giordano Grandis
• [Asterisk-Users] Caller Name: Asterisk reading too fast   Jeremy Gault
• [Asterisk-Users] direct sip call pickup   Alexander Lopez
• [Asterisk-Users] mpg123 on x86_64 (Opteron MP)   asterisk at gatheringpoint.net
• [Asterisk-Users] Caller Name: Asterisk reading too fast   Alexander Lopez
• [Asterisk-Users] New version of idefisk softphone released.   Zoa
• [Asterisk-Users] Caller Name: Asterisk reading too fast   steve at daviesfam.org
• R: [Asterisk-Users] direct sip call pickup   Giordano Grandis
• R: [Asterisk-Users] direct sip call pickup   Iqbal
• use of automon sequence (was Re: [Asterisk-Users] Call recording between SIP phones)   John covici
• [Asterisk-Users] Asterisk as a gateway. 'flash for transfers transparency?'   felipe hangen
• [Asterisk-Users] RxFax/TxFax - Compile Problem   Dave Cotton
• [Asterisk-Users] queue_log on mysql   William Lloyd
• [Asterisk-Users] Easier way for end user to change main greeting?   Doug
• [Asterisk-Users] Re: T.38 ATA   VoIP Newbie
• R: [Asterisk-Users] direct sip call pickup   Mojo with Horan & Company, LLC
• [Asterisk-Users] RxFax/TxFax - Compile Problem   Alexander Lopez
• [Asterisk-Users] Sipura 2k voice quality   Matthew Harrell
• [Asterisk-Users] Zap failed   Ugur GUNCER
• [Asterisk-Users] wiki down?   Francesco Peeters
• [Asterisk-Users] didn't get a frame from channel   Andres Paglayan
• [Asterisk-Users] wiki down?   Wiley Siler
• [Asterisk-Users] Sipura 2k voice quality   Andres Paglayan
• [Asterisk-Users] Re: Polycom randomly fails outbound calls,   Andres Paglayan
• [Asterisk-Users] lastest spandsp-0.03pre1 don't compile   Raymond Chen
• [Asterisk-Users] USB Phones for use with Asterisk   Guillermo Salas M
• [Asterisk-Users] queue_log on mysql   lenz
• [Asterisk-Users] Re: Polycom randomly fails outbound calls,   Noah Miller
• R: [Asterisk-Users] direct sip call pickup   Joshua Colp - Asterlink
• [Asterisk-Users] Weird behaviour   andrutto
• [Asterisk-Users] asterisk mixing sound card with anybody?   Jerry Geis
• [Asterisk-Users] How to access * thru router when ip address is not known   Robert Rawlinson
• [Asterisk-Users] How to access * thru router when ip address is not known   mike.jennings at charter.net
• [Asterisk-Users] Re: g729 to asterisk to g729 voip provider   Erick Perez
• [Asterisk-Users] wiki down?   Francesco Peeters
• [Asterisk-Users] Re: g729 to asterisk to g729 voip provider   ChB
• [Asterisk-Users] wiki down?   ChB
• [Asterisk-Users] How to access * thru router when ip address is not known   Colin Anderson
• [Asterisk-Users] USB Phones for use with Asterisk   Derek Whitten
• [Asterisk-Users] How to access * thru router when ip address is not known   Colin Anderson
• [Asterisk-Users] Anyone using iPlan Networks in Argentina?   Ilan Rabinovitch
• [Asterisk-Users] Re: g729 to asterisk to g729 voip provider   Erick Perez
• [Asterisk-Users] Re: g729 to asterisk to g729 voip provider   Christian B
• [Asterisk-Users] How to access * thru router when ip address is not known   razza
• [Asterisk-Users] linux sip or iax phone that will autoanswer and route to console   Jerry Geis
• [Asterisk-Users] Orinoco Injectors   Darren Wright
• [Asterisk-Users] Seperate Incoming calls on TDM02?   Matt Love
• [Asterisk-Users] mpg123 on x86_64 (Opteron MP)   Joseph
• [Asterisk-Users] How to access * thru router when ip address is not known   Colin Anderson
• [Asterisk-Users] How to access * thru router when ip address is not known   Paul
• [Asterisk-Users] mpg123 on x86_64 (Opteron MP)   Paul
• [Asterisk-Users] TDM400P Dialing Out - "Cannot be completed as dialed"   Barry King
• [Asterisk-Users] How to IGNORE distinctive ring   Steven Premeau
• [Asterisk-Users] Grandstream   Joshua Abbott
• [Asterisk-Users] Grandstream   Rene Kluwen
• [Asterisk-Users] Re: g729 to asterisk to g729 voip provider   Enzo Michelangeli
• [Asterisk-Users] Double Ring   Matt
• Fwd: [Asterisk-Users] Seperate Incoming calls on TDM02?   C. Hatton Humphrey
• [Asterisk-Users] Re: Double Ring   Matt
• [Asterisk-Users] free IAX calling platform   Matthew Simpson
• [Asterisk-Users] Who is going to AstriCon (The Asterisk Conference)?   Steven Sokol
• [Asterisk-Users] Grandstream   Jason Walker
• [Asterisk-Users] Who is going to AstriCon (The Asterisk Conference)?   Matthew Simpson
• [Asterisk-Users] wav instead of gsm for vm-sounds?   Damon Estep
• [Asterisk-Users] Who is going to AstriCon (The Asterisk Conference)?   Brian Roy
• [Asterisk-Users] wav instead of gsm for vm-sounds?   Joshua Abbott
• [Asterisk-Users] wav instead of gsm for vm-sounds?   Damon Estep
• [Asterisk-Users] Who is going to AstriCon (The Asterisk Conference)?   Steven Sokol
• [Asterisk-Users] Re: Sipura 2k voice quality   Matthew Harrell
• [Asterisk-Users] wav instead of gsm for vm-sounds?   Kevin P. Fleming
• [Asterisk-Users] wav instead of gsm for vm-sounds?   Damon Estep
• [Asterisk-Users] wav instead of gsm for vm-sounds?   Kevin P. Fleming
• [Asterisk-Users] wav instead of gsm for vm-sounds?   Damon Estep
• [Asterisk-Users] wav instead of gsm for vm-sounds?   Kevin P. Fleming
• [Asterisk-Users] wav instead of gsm for vm-sounds?   Damon Estep
• [Asterisk-Users] TDM400P Dialing Out - "Cannot be completed as dialed"   MF Hulber
• [Asterisk-Users] wav instead of gsm for vm-sounds?   Damon Estep
• [Asterisk-Users] wav instead of gsm for vm-sounds?   Kevin P. Fleming
• [Asterisk-Users] TDM400P Dialing Out - "Cannot be completed as dialed"   John Novack (port)
• [Asterisk-Users] How to make Basic authenticatuion in Asterisk server.   shubhendu mishra
• [Asterisk-Users] Odd SpanDSP behavior   Chris
• [Asterisk-Users] Grandstream   Rob Lith
• [Asterisk-Users] Who is going to AstriCon (The Asterisk Conference)?   Zoa
• [Asterisk-Users] Re: wav instead of gsm for vm-sounds?   Tony Mountifield
• [Asterisk-Users] AgentCallbackLogin and calling outside   Rajkumar S
• [Asterisk-Users] DNS SRV supported phones   Justin Richards
• [Asterisk-Users] queue_log on mysql   William Lloyd
• [Asterisk-Users] Who is going to AstriCon (The Asterisk Conference)?   steve at daviesfam.org
• [Asterisk-Users] MGCP service from Free Télécom   Stewart Nelson
• [Asterisk-Users] Flash Operator Panel Help   Insider KT
• [Asterisk-Users] Re: wav instead of gsm for vm-sounds?   Kevin P. Fleming
• [Asterisk-Users] Re: wav instead of gsm for vm-sounds?   Damon Estep
• [Asterisk-Users] Re: wav instead of gsm for vm-sounds?   Kevin P. Fleming
• [Asterisk-Users] Flash Operator Panel Help   Nicolás Gudiño
• [Asterisk-Users] Re: wav instead of gsm for vm-sounds?   Damon Estep
• [Asterisk-Users] AgentCallbackLogin and calling outside   Damon Estep
• [Asterisk-Users] Re: Who is going to AstriCon (The Asterisk Conference)?   Steven Sokol
• [Asterisk-Users] Re: wav instead of gsm for vm-sounds?   Kevin P. Fleming
• [Asterisk-Users] New version of idefisk softphone released.   Mike Clark
• [Asterisk-Users] AgentCallbackLogin and calling outside   BJ Weschke
• [Asterisk-Users] Who is going to AstriCon (The Asterisk Conference)?   Steven Sokol
• [Asterisk-Users] AgentCallbackLogin and calling outside   Rajkumar S
• [Asterisk-Users] Who is going to AstriCon (The Asterisk Conference)?   Kevin P. Fleming
• [Asterisk-Users] Who is going to AstriCon (The Asterisk Conference)?   Kurth Bemis
• [Asterisk-Users] Asterisk realtime beta   Olivier Taylor
• [Asterisk-Users] Re: Who is going to AstriCon (The Asterisk Conference)?   Thorsten Lockert
• [Asterisk-Users] Who is going to AstriCon (The Asterisk Conference)?   Steven Sokol
• [Asterisk-Users] mpg123 on x86_64 (Opteron MP)   asterisk at gatheringpoint.net
• [Asterisk-Users] Asterisk realtime beta   Urban
• [Asterisk-Users] Re: Who is going to AstriCon (The Asterisk Conference)?   Matthew Simpson
• [Asterisk-Users] Caller ID for auto outgoing calls   Paul
• [Asterisk-Users] Who is going to AstriCon (The Asterisk Conference)?   Steven Sokol
• [Asterisk-Users] Re: wav instead of gsm for vm-sounds?   Tony Mountifield
• [Asterisk-Users] Who is going to AstriCon (The Asterisk Conference)?   Steven Sokol
• [Asterisk-Users] AstriCon 2006 Location   Kenny Kant
• [Asterisk-Users] AstriCon 2006 Location   Steven Sokol
• [Asterisk-Users] (no subject)   Insider KT
• [Asterisk-Users] AstriCon 2006 Location   Matthew Simpson
• [Asterisk-Users] AstriCon 2006 Location   Kevin Bockman
• [Asterisk-Users] Asterisk realtime beta   Matthew Boehm
• [Asterisk-Users] Re: Who is going to AstriCon (The Asterisk Conference)?   Bruce Ferrell
• [Asterisk-Users] capiFax causes segfault on asterisk   gw at adcomcorp.com
• [Asterisk-Users] capiFax causes segfault on asterisk   gw at adcomcorp.com
• [Asterisk-Users] Re: wav instead of gsm for vm-sounds?   Damon Estep
• [Asterisk-Users] internet connection between Africa and Europe   steve at daviesfam.org
• [Asterisk-Users] ASTCC issues   Darren Wiebe
• [Asterisk-Users] moh - turn off randomization?   Damon Estep
• [Asterisk-Users] AstriCon 2006 Location   asterisk
• [Asterisk-Users] MGCP service from Free Télécom   Dave Weis
• [Asterisk-Users] ZapHFC Channel unavailable   Rene Kluwen
• [Asterisk-Users] AstriCon 2006 Location   Steven Sokol
• RE: [Asterisk-Users] MGCP service from Free Télécom   Senad J
• [Asterisk-Users] Anybody using SIP Interaction Proxy 2.X and Asterisk CVS head?   Chris Modesitt
• [Asterisk-Users] checking voice mail from different phone   Joseph
• [Asterisk-Users] DNS SRV supported phones   Howard Lowndes
• [Asterisk-Users] moh - turn off randomization?   Kevin P. Fleming
• [Asterisk-Users] moh - turn off randomization?   Damon Estep
• [Asterisk-Users] unlocking cisco 7940 phone   Brian
• [Asterisk-Users] Who is going to AstriCon (The AsteriskConference)?   Craig Guy
• [Asterisk-Users] Who is going to AstriCon (The Asterisk Conference)?   Steven Sokol
• [Asterisk-Users] bounty partners and/or possible coder? queues.conf ackcall and pre-ack announce   Damon Estep
• [Asterisk-Users] checking voice mail from different phone   Damon Estep
• [Asterisk-Users] Who is going to AstriCon (The AsteriskConference)?   Caleb
• [Asterisk-Users] checking voice mail from different phone   Paul
• 回复： Re: [Asterisk-Users] queue_log on mysql   Gary Li
• [Asterisk-Users] unlocking cisco 7940 phone   Tom Vile
• [Asterisk-Users] How does one set-up incoming/outgoing SIP with no registration and only IP authentication?   Frank Tarczynski
• [Asterisk-Users] AstriCon 2006 Location   Reid Forrest
• [Asterisk-Users] Who is going to AstriCon (TheAsteriskConference)?   Dean Collins
• [Asterisk-Users] AstriCon 2006 Location   Steven Sokol
• [Asterisk-Users] Who is going to AstriCon (TheAsteriskConference)?   John covici
• [Asterisk-Users] Who is going to AstriCon (TheAsteriskConference)?   Jason Walker
• [Asterisk-Users] Still having hangup problems in NZ   Simon
• [Asterisk-Users] Complete NPA-NXX list for USA/Canada npanxx, ratecenters, etc (attached)   chris-list
• [Asterisk-Users] Asterisk realtime beta   Olivier Taylor
• [Asterisk-Users] DID from an analog phone   Christian Gansberger
• [Asterisk-Users] AstriCon 2006 Location   Matt Florell
• [Asterisk-Users] DID from an analog phone   C. Hatton Humphrey
• [Asterisk-Users] moh - turn off randomization?   Bob Goddard
• [Asterisk-Users] g729 passthrough?   Elton Machado
• [Asterisk-Users] moh - turn off randomization?   Kevin P. Fleming
• [Asterisk-Users] AstriCon 2006 Location   Kevin P. Fleming
• [Asterisk-Users] AstriCon 2006 Location   Senad J
• [Asterisk-Users] AstriCon 2006 Location   asterisk
• [Asterisk-Users] TFTP and DHCP...   Francois Meehan
• [Asterisk-Users] TFTP and DHCP...   Bob Goddard
• [Asterisk-Users] (no subject)   lucabencini at tin.it
• [Asterisk-Users] Asterisk realtime beta   Olivier Taylor
• [Asterisk-Users] Voicemail   lucabencini at tin.it
• [Asterisk-Users] Voicemail   Damon Estep
• [Asterisk-Users] AstriCon 2006 Location   Matthew Simpson
• [Asterisk-Users] AstriCon 2006 Location   Matthew Simpson
• [Asterisk-Users] voicemail context. macro, and directory   Damon Estep
• [Asterisk-Users] AstriCon 2006 Location   Brian C
• [Asterisk-Users] Who is going to AstriCon (The AsteriskConference)?   canuck15
• [Asterisk-Users] AstriCon 2006 Location   Paul
• [Asterisk-Users] Who is going to AstriCon (The AsteriskConference)?   Paul
• [Asterisk-Users] Asterisk Won't Process Call   Michael Stearne
• [Asterisk-Users] Asterisk Won't Process Call   steve at daviesfam.org
• [Asterisk-Users] AstriCon 2006 Location   Rich Adamson
• [Asterisk-Users] AstriCon 2006 Location   Brian Capouch
• [Asterisk-Users] AstriCon 2006 Location   Paul
• [Asterisk-Users] AstriCon 2006 Location   Senad J
• [Asterisk-Users] AstriCon 2006 Location   Brian Capouch
• [Asterisk-Users] AstriCon 2006 Location   John Novack
• [Asterisk-Users] AstriCon 2006 Location   Rich Adamson
• [Asterisk-Users] AstriCon 2006 Location   Rich Adamson
• [Asterisk-Users] Two POTS in, but only want one out?   Ken D'Ambrosio
• [Asterisk-Users] PRI to PRI passthrough with DID intact   Steven
• [Asterisk-Users] AstriCon 2006 Location   Reid Forrest
• [Asterisk-Users] Asterisk Won't Process Call   Klaus Sonnenleiter
• [Asterisk-Users] Two POTS in, but only want one out?   Rich Adamson
• [Asterisk-Users] limiting calls per day based on amount of time   Joseph
• [Asterisk-Users] Re: Asterisk-Users Digest, Vol 14, Issue 108   Tran Tony
• [Asterisk-Users] New User -> couple of dumb questions   Shawn Porter
• [Asterisk-Users] AstriCon 2006 Location   Mike Clark
• [Asterisk-Users] Integration between Asterisk and SiemensHiCom150e over ISDN   Simon D
• [Asterisk-Users] Integration between Asterisk andSiemensHiCom150e over ISDN   wendys at tiscali.de
• [Asterisk-Users] ChanSpy not loading   Hussain Umair
• [Asterisk-Users] Differ between "private" and "out of area"?   Goran Dj.
• [Asterisk-Users] SpanDSP   Doug Lytle
• [Asterisk-Users] Julien COURTEMANCHE/TELINTRANS/FR est absent(e).   julien.courtemanche at telintrans.fr
• [Asterisk-Users] sometimes CIDNUM shows, sometimes CIDNAME??? Why, why, why, why?   Goran Dj.
• [Asterisk-Users] New User -> couple of dumb questions   Rich Adamson
• [Asterisk-Users] sometimes CIDNUM shows, sometimes CIDNAME??? Why, why, why, why?   Shaun Ewing
• [Asterisk-Users] Voicemail   Matt
• [Asterisk-Users] sometimes CIDNUM shows, sometimes CIDNAME??? Why, why, why, why?   Goran Dj.
• [Asterisk-Users] AstriCon 2006 Location   Tad Heckaman
• [Asterisk-Users] AstriCon 2006 Location   Jonathan k. Creasy
• [Asterisk-Users] Voicemail   Rich Adamson
• [Asterisk-Users] Voicemail   Damon Estep
• [Asterisk-Users] Monitor and sox mix quality   gw at adcomcorp.com
• [Asterisk-Users] HW Question (TDM400)   rudolfl at optusnet.com.au
• [Asterisk-Users] Cisco Callmanager & Asterisk for Voicemail revisited   Shaun Ewing
• [Asterisk-Users] E1 configuration problem   manish kumar
• [Asterisk-Users] E1 configuration problem   Gurminder Arora
• [Asterisk-Users] limiting calls per day based on amount of time   Darren Wiebe
• [Asterisk-Users] ASTCC speaks and cut RTP channel   Darren Wiebe
• [Asterisk-Users] Cisco Callmanager & Asterisk for Voicemail revisited   Rich Adamson
• [Asterisk-Users] Monitor and sox mix quality   Jonathan Feally
• [Asterisk-Users] Monitor and sox mix quality   Elmar Haneke
• [Asterisk-Users] Asterisk ISDN: Problem Setting CallerID as DID Extension Numbers.   Voicomm User
• [Asterisk-Users] problems with remote access to PSTN   asterisk at frameweb.it
• [Asterisk-Users] Asterisk ISDN: Problem Setting CallerID as DID Extension Numbers.   Armin Schindler
• [Asterisk-Users] AstriCon 2006 Location   Giovanni Miano
• [Asterisk-Users] Asterisk realtime beta   Urban
• [Asterisk-Users] HW Question (TDM400)   Patrick
• [Asterisk-Users] Cisco Callmanager & Asterisk for Voicemail revisited   Doug Lytle
• [Asterisk-Users] ISDN BRI 2 pci cards and mISDN   Dias Badekas
• [Asterisk-Users] HW Question (TDM400)   Rudolf Ladyzhenskii
• [Asterisk-Users] Voipbuster in Australia -- delay problem   Rudolf Ladyzhenskii
• [Asterisk-Users] E1 configuration problem   Sergio Serrano
• [Asterisk-Users] Voipbuster in Australia -- delay problem   Paul
• [Asterisk-Users] Voipbuster in Australia -- delay problem   Rudolf Ladyzhenskii
• [Asterisk-Users] Cisco Callmanager & Asterisk for Voicemail revisited   Shaun Ewing
• [Asterisk-Users] How does one set-up incoming/outgoing SIP with no registration and only IP authentication?   Frank Tarczynski
• [Asterisk-Users] Voipbuster in Australia -- delay problem   pisacc at gmail.com
• [Asterisk-Users] Cisco Callmanager & Asterisk for Voicemail revisited   info at beprojects.com
• [Asterisk-Users] need a simply configuration for calling in/out to PSTN   Ahmad
• [Asterisk-Users] Cisco Callmanager & Asterisk for Voicemail revisited   pisacc at gmail.com
• [Asterisk-Users] Cisco Callmanager & Asterisk for Voicemail revisited   Shaun Ewing
• [Asterisk-Users] Unable to allocate channel structure   Crystal Stream, Incorporated
• [Asterisk-Users] Cisco Callmanager & Asterisk for Voicemail revisited   Pisac
• [Asterisk-Users] Cisco Callmanager & Asterisk for Voicemail revisited   Pisac
• [Asterisk-Users] Cisco Callmanager & Asterisk for Voicemail revisited   info at beprojects.com
• [Asterisk-Users] Complete NPA-NXX list for USA/Canada npanxx, ratecenters, etc (attached)   Kurth Bemis
• [Asterisk-Users] Prompt translation: can't find "please wait try ext" prompt filename   Alexandre Leclerc
• [Asterisk-Users] Asterisk in Spanish   Sebastian Milioto
• [Asterisk-Users] Complete NPA-NXX list for USA/Canada npanxx, ratecenters, etc (attached)   Joel Vandal
• [Asterisk-Users] clear SIP channel   Bart van Daal
• [Asterisk-Users] limiting calls per day based on amount of time   Tzafrir Cohen
• [Asterisk-Users] DNS SRV supported phones   Anish Basu
• [Asterisk-Users] Asterisk in Spanish   Shaun Ewing
• [Asterisk-Users] Complete NPA-NXX list for USA/Canada npanxx,   Joe Greco
• [Asterisk-Users] Prompt translation: can't find "please wait try ext" prompt filename   Shaun Ewing
• [Asterisk-Users] Voicemail   Matt
• [Asterisk-Users] AstriCon 2006 Location   Jonathan k. Creasy
• [Asterisk-Users] Dlink dph-140s/ACT P104SLD   D. J. Williams
• [Asterisk-Users] AstriCon 2006 Location   Jonathan k. Creasy
• [Asterisk-Users] problems with PRI   asterisk at frameweb.it
• [Asterisk-Users] AstriCon 2006 Location   Darren Younger
• [Asterisk-Users] Complete NPA-NXX list for USA/Canada npanxx,   Nathan Pralle
• [Asterisk-Users] AstriCon 2006 Location   W. Kevin Hunt
• [Asterisk-Users] TXFAX   Il Neofita
• [Asterisk-Users] Differ between "private" and "out of area"?   Matt
• [Asterisk-Users] AstriCon 2006 Location   Brian C. Fertig
• [Asterisk-Users] AstriCon 2006 Location   asterisk
• [Asterisk-Users] AstriCon 2006 Location   Paul
• [Asterisk-Users] Asterisk in Spanish   Moises Silva
• [Asterisk-Users] TFTP and DHCP...   Francois Meehan
• [Asterisk-Users] Asterisk Won't Process Call   Michael Stearne
• [Asterisk-Users] Asterisk Won't Process Call   steve at daviesfam.org
• [Asterisk-Users] Asterisk realtime beta   Matthew Boehm
• [Asterisk-Users] AstriCon 2006 Location   Jonathan k. Creasy
• [Asterisk-Users] AstriCon 2006 Location   Jonathan k. Creasy
• [Asterisk-Users] Voicemail   Chris Coulthurst
• [Asterisk-Users] FW: ADTRAN Virtual Classes: Ensuring QoS for VoIP & Total Access 900 Series   Dean Collins
• [Asterisk-Users] AstriCon 2006 Location   Jonathan k. Creasy
• [Asterisk-Users] Meetme Problem   kurt x
• [Asterisk-Users] chan_alsa.c blocking sound port - solution   Jerry Geis
• [Asterisk-Users] Prompt translation: can't find "please wait try ext" prompt filename   Alexandre Leclerc
• [Asterisk-Users] Unable to open space (format ulaw)?   Paul
• [Asterisk-Users] sip invite question   Matt Hess
• [Asterisk-Users] Complete NPA-NXX list for USA/Canada npanxx,   Joe Greco
• [Asterisk-Users] TXFAX   Damon Estep
• [Asterisk-Users] Round-robin with Queue   Jeremy Gault
• [Asterisk-Users] Differ between "private" and "out of area"?   Goran Dj
• [Asterisk-Users] AstriCon 2006 Location   Paul
• [Asterisk-Users] OT: Xoops Skype module   Dean Collins
• [Asterisk-Users] Asterisk ISDN: Problem Setting CallerID as DIDExtension Numbers.   gw at adcomcorp.com
• [Asterisk-Users] OT: Hardware Interrupts; Who is it?   Matthew Boehm
• [Asterisk-Users] Easier way for end user to change main greeting?   Scott
• [Asterisk-Users] Meetme Problem   Doug Lytle
• [Asterisk-Users] hints not working on CVS HEAD   Paradise Dove
• [Asterisk-Users] Anyone have the firmware for WRT54GP2?   Sherwood McGowan
• [Asterisk-Users] AstriCon 2006 Location   Alan Bunch
• [Asterisk-Users] How does one set-up incoming/outgoing SIP with no registration and only IP authentication?   Jason Williams
• [Asterisk-Users] Asterisk realtime beta   Olivier Taylor
• [Asterisk-Users] Differ between "private" and "out of area"?   Matt
• [Asterisk-Users] Asterisk in Spanish   Elio Rojano
• [Asterisk-Users] SIP audio port usage   Adrien Laurent
• [Asterisk-Users] i4l ring indication problem, again...   Omadon
• [Asterisk-Users] VM low volume - testers needed   Andrew Kohlsmith
• [Asterisk-Users] Meetme Problem   Rich Adamson
• [Asterisk-Users] VM low volume - testers needed   Rich Adamson
• [Asterisk-Users] Asterisk in Spanish   Sergio Serrano
• [Asterisk-Users] hints and the sNOM 360   Paul Hewlett
• [Asterisk-Users] Complete NPA-NXX list for USA/Canada npanxx,   Damon Estep
• [Asterisk-Users] OT: Hardware Interrupts; Who is it?   Tony Hoyle
• [Asterisk-Users] Sip and ISDN problem   Andrew Nowrot
• [Asterisk-Users] VM low volume - testers needed   Patrick
• [Asterisk-Users] SIP audio port usage   Sherwood McGowan
• [Asterisk-Users] Pinging ...   Alan Bunch
• [Asterisk-Users] Fwd: Asterisk in Spanish   Sebastian Milioto
• [Asterisk-Users] Differ between "private" and "out of area"?   Rich Adamson
• [Asterisk-Users] OT: Hardware Interrupts; Who is it?   Rich Adamson
• [Asterisk-Users] Asterisk realtime beta   Matthew Boehm
• [Asterisk-Users] ztdummy configuration help   kurt x
• [Asterisk-Users] HooDaHek Version 0.5 Release   Nathan E. Pralle
• [Asterisk-Users] Most desireable Linux distribution for Asterisk?   Asterisk Lists
• [Asterisk-Users] Re: SIP audio port usage   Stefan Tichy
• [Asterisk-Users] Re: ztdummy configuration help   Stefan Tichy
• [Asterisk-Users] Looking for firmware for Cisco 12sp+ and 30VIP   Stern, Craig
• [Asterisk-Users] kill a .call file   jltaylor
• [Asterisk-Users] IAX dialplan problem?   John Crowhurst
• [Asterisk-Users] Differ between "private" and "out of area"?   jltaylor
• [Asterisk-Users] Asterisk monitoring availability   Sig Lange
• [Asterisk-Users] kill a .call file   trixter http://www.0xdecafbad.com
• [Asterisk-Users] IAX dialplan problem?   Thameem Ansari
• [Asterisk-Users] Differ between "private" and "out of area"?   trixter http://www.0xdecafbad.com
• [Asterisk-Users] Most desireable Linux distribution for Asterisk?   Michiel van Baak
• [Asterisk-Users] OT: Hardware Interrupts; Who is it?   Matthew Boehm
• [Asterisk-Users] Asterisk monitoring availability   Sherwood McGowan
• [Asterisk-Users] AstriCon 2006 Location   Matt Fredrickson
• [Asterisk-Users] IAX dialplan problem?   John Crowhurst
• [Asterisk-Users] Most desireable Linux distribution for Asterisk?   José Pablo Ezequiel Fernández
• [Asterisk-Users] kill a .call file   jltaylor
• [Asterisk-Users] OT: Hardware Interrupts; Who is it?   Sig Lange
• [Asterisk-Users] Pinging ...   Alan Bunch
• [Asterisk-Users] Error with VOIP-INFO   Crystal Stream, Incorporated
• [Asterisk-Users] SIP audio port usage   Rich Adamson
• [Asterisk-Users] OT: Hardware Interrupts; Who is it?   Kevin Collins
• [Asterisk-Users] Most desireable Linux distribution for Aster isk?   Colin Anderson
• [Asterisk-Users] realtime audio for asterisk using jack   Esben Stien
• [Asterisk-Users] Re: Pinging ...   Noah Miller
• [Asterisk-Users] Asterisk Keep Crashing need Help please   Ka Lun Chan
• [Asterisk-Users] Differ between "private" and "out of area"?   Goran Dj
• [Asterisk-Users] Pinging ...   Ariel Batista
• [Asterisk-Users] Most desireable Linux distribution for Aster isk?   Ariel Batista
• [Asterisk-Users] Most desireable Linux distribution for Asterisk?   Nathan Pralle
• [Asterisk-Users] kill a .call file   Jeremy Gault
• [Asterisk-Users] kill a .call file   Remco Barende
• [Asterisk-Users] unlocking cisco 7940 phone   Brian
• [Asterisk-Users] Pinging ...   Alan Bunch
• [Asterisk-Users] OT: Hardware Interrupts; Who is it?   Rich Adamson
• [Asterisk-Users] T.38 & Canreinvite (yes, again)   info at beprojects.com
• [Asterisk-Users] Silence suppression /RTP VAD and Asterisk? Dropped calls on IP-500   david.j as
• [Asterisk-Users] Orinoco Injectors   Michael Loftis
• [Asterisk-Users] IAX dialplan problem?   Colin Anderson
• [Asterisk-Users] kill a .call file   Rene Kluwen
• [Asterisk-Users] T.38 & Canreinvite (yes, again)   Joshua Colp - Asterlink
• [Asterisk-Users] unlocking cisco 7940 phone   chouck
• [Asterisk-Users] Most desireable Linux distribution for Asterisk?   Paul
• [Asterisk-Users] OT: Hardware Interrupts; Who is it?   Matthew Boehm
• [Asterisk-Users] Silence suppression /RTP VAD and Asterisk? Dropped calls on IP-500   Jeremy Gault
• [Asterisk-Users] Dial time limit doesn't work when calling party transfers   Alejandro Ríos P.
• [Asterisk-Users] IAX dialplan problem?   John Crowhurst
• [Asterisk-Users] AstriCon 2006 Location   Asterisk
• [Asterisk-Users] Most desireable Linux distribution for Asterisk?   Matt Florell
• [Asterisk-Users] Point to Point with Fritz Card ...   Joao Correia
• [Asterisk-Users] AstriCon 2006 Location   Robert Hajime Lanning
• [Asterisk-Users] Differ between "private" and "out of area"?   Rich Adamson
• [Asterisk-Users] unlocking cisco 7940 phone   Michael Welter
• [Asterisk-Users] OT: Hardware Interrupts; Who is it?   Kevin Collins
• [Asterisk-Users] TDM400P Dialing Out - "Cannot be completed as dialed"   Barry King
• [Asterisk-Users] TDM400P Dialing Out - "Cannot be completed as dialed"   Barry King
• [Asterisk-Users] IAX dialplan problem?   Colin Anderson
• [Asterisk-Users] H.263 Format video   Areski K
• [Asterisk-Users] [Fwd: ASTCC speaks and cut RTP channel => Kind of solution...   Ricardo Poppi
• [Asterisk-Users] unlocking cisco 7940 phone   Rich Adamson
• [Asterisk-Users] pridialplan per call or per channel group?   Gary Reuter
• [Asterisk-Users] TDM400P Dialing Out - "Cannot be completed as dialed"   John Novack
• [Asterisk-Users] Asterisk Keep Crashing need Help please   Mark Edwards
• [Asterisk-Users] AstriCon 2006 Location   Shaun Ewing
• [Asterisk-Users] Looking for firmware for Cisco 12sp+ and 30VIP   Shaun Ewing
• [Asterisk-Users] hfc card unplug & plug not working?   bman at host.sk
• [Asterisk-Users] MSNs don't work for me... :(   Yoann Le Bihan
• [Asterisk-Users] Looking for firmware for Cisco 12sp+ and 30VIP   Goran Dj
• [Asterisk-Users] TDM400P Dialing Out - "Cannot be completed as dialed"   Barry King
• [Asterisk-Users] kill a .call file   Remco Barende
• [Asterisk-Users] Voicemail() application returning -1 on a hangup   Robert G. Ristroph
• [Asterisk-Users] Zap calls dropping just after answer   Jesse Keating
• [Asterisk-Users] Re: Most desireable Linux distribution for Asterisk?   asterisk
• [Asterisk-Users] Re: Welcome to the "Asterisk-Users" mailing list   chuck gelm
• [Asterisk-Users] unlocking cisco 7940 phone   Brian
• [Asterisk-Users] Call dropped 100% of time when incoming IAX routed to outgoing CAPI   Christopher Mylonas
• [Asterisk-Users] T.38 & Canreinvite (yes, again)   Carlos Alperin
• [Asterisk-Users] TDM400P Dialing Out - "Cannot be completed as dialed"   John Novack (port)
• [Asterisk-Users] Voicemail   Chris Coulthurst
• [Asterisk-Users] need example about sjphone with asterisk   julien bossart
• [Asterisk-Users] asterisk frequently dead   stevanus
• [Asterisk-Users] MWI indicator HINT on Snom thru IAX?   Colin Anderson
• [Asterisk-Users] Zap calls dropping just after answer   Jesse Keating
• [Asterisk-Users] TDM400P stops answering   Andy Howell
• [Asterisk-Users] Buy a digium hardware   Leopoldo Rodríguez H
• [Asterisk-Users] ztdummy configuration help   Lohit
• [Asterisk-Users] Point to Point with Fritz Card ...   Craig Guy
• [Asterisk-Users] TE410 stop responding   Chee Foong
• [Asterisk-Users] "Stopping retransmission on" messages   Chris Miller
• [Asterisk-Users] Re: Welcome to the "Asterisk-Users" mailing list   Francesco Peeters
• [Asterisk-Users] Resolving QOS problems   Chris Miller
• [Asterisk-Users] OT: Xoops Skype module   Tzafrir Cohen
• [Asterisk-Users] hfc card unplug & plug not working?   gincantalupo
• [Asterisk-Users] need example about sjphone with asterisk   Zoa
• [Asterisk-Users] AstriCon 2006 Location   Wayne Gemmell
• [Asterisk-Users] Cisco 7960 Locking Up   Mark Johnson
• [Asterisk-Users] problems with PRI   asterisk at frameweb.it
• [Asterisk-Users] Resolving QOS problems   Matt Riddell
• [Asterisk-Users] Hangup after voicemail not detected   Gurminder Arora
• [Asterisk-Users] "Stopping retransmission on" messages   Matt Riddell
• [Asterisk-Users] PTN calls into asterisk slow release   C W Nel
• [Asterisk-Users] sipp examples   Julian Lyndon-Smith
• [Asterisk-Users] Is there a clever way to page a group of extensions?   Patrick Lidstone (Personal e-mail)
• [Asterisk-Users] i4l ring indication problem, again...   Massimo Frisoni
• [Asterisk-Users] pri gateway   Baris Simsek
• [Asterisk-Users] HELP: Valiant E1 CB and UniCall   Paradise Dove
• [Asterisk-Users] Connect not signalled (SIP -> Zap)   Roger Schreiter
• [Asterisk-Users] Buy a digium hardware   Wiley Siler
• [Asterisk-Users] Re: Point to Point with Fritz Card ...   Dias Badekas
• [Asterisk-Users] asterisk-oh323: New versions 0.6.7 and 0.7.3   Michael Manousos
• [Asterisk-Users] General Config information   Paul Goodyear
• [Asterisk-Users] What hardware would you recommend?   Francesco Peeters
• [Asterisk-Users] Integrating * with Mitel SX2000 Lite   John Mylchreest
• [Asterisk-Users] SIP audio port usage   Adrien Laurent
• [Asterisk-Users] SIP audio port usage   Sherwood McGowan
• [Asterisk-Users] Red or Yellow alarm monitoring   Steven
• [Asterisk-Users] [Fwd: ASTCC speaks and cut RTP channel => Kind of solution...   Darren Wiebe
• [Asterisk-Users] SIP audio port usage   Sebastian Milioto
• [Asterisk-Users] Asterisk@Home Music on Hold   Armin Lediger
• [Asterisk-Users] SIP audio port usage   Sherwood McGowan
• [Asterisk-Users] using a voip cable modem   Calvin Lockhart
• [Asterisk-Users] SIP audio port usage   Rich Adamson
• [Asterisk-Users] Aterisk App ICES Question   Daniel Mikusa
• [Asterisk-Users] Cisco Ip phones   Sander
• [Asterisk-Users] one way voice   Mark D'Cruz
• [Asterisk-Users] one way voice   Tom Vile
• [Asterisk-Users] Cisco 7960 Locking Up   Rich Adamson
• [Asterisk-Users] HooDaHek 0.6 Released   Nathan Pralle
• [Asterisk-Users] HooDaHek 0.6 Released   Paul
• [Asterisk-Users] Cisco Ip phones   Rich Adamson
• [Asterisk-Users] STUN vs NAT Helper   Dan Adams
• [SPAM:***** SpamScore] Re: [SPAM:******** SpamScore] [Asterisk-Users] Call Transfer using SIP clients   hugolivude
• [Asterisk-Users] BackgroundDetect problem   Kevin Bockman
• [Asterisk-Users] HooDaHek 0.6 Released   Nathan Pralle
• [Asterisk-Users] HooDaHek 0.6 Released   Rene Kluwen
• [Asterisk-Users] Re: SMS using a PRI channel   Stefan Tichy
• [Asterisk-Users] how to distinguish the "ringing" and "connected" for zap channel   Liu Peter
• [Asterisk-Users] ODBC Voicemail WEB Retrieval   pbx at itsngroup.com
• [Asterisk-Users] Re: HooDaHek 0.6 Released   Steven
• [Asterisk-Users] agent channel busy - how to stop it?   1 2
• [Asterisk-Users] Re: how to distinguish the "ringing" and "connected"for zap channel   Alchaemist
• [Asterisk-Users] RxFax/TxFax - Compile Problem   Alexander Lopez
• [Asterisk-Users] HooDaHek 0.6 Released   Nathan Pralle
• [Asterisk-Users] RxFax/TxFax - Compile Problem   José Pablo Ezequiel Fernández
• [Asterisk-Users] RxFax/TxFax - Compile Problem   Doug Lytle
• [Asterisk-Users] HooDaHek 0.6 Released   Jonathan k. Creasy
• [Asterisk-Users] RxFax/TxFax - Compile Problem   Alexander Lopez
• [Asterisk-Users] RxFax/TxFax - Compile Problem   Michael Welter
• [Asterisk-Users] Cisco Ip phones   Florian Overkamp
• [Asterisk-Users] RxFax/TxFax - Compile Problem   Alexander Lopez
• [Asterisk-Users] HooDaHek 0.6 Released   trixter http://www.0xdecafbad.com
• [Asterisk-Users] T.38 & Canreinvite (yes, again)   list
• [Asterisk-Users] ODBC Voicemail WEB Retrieval   Dan Littlejohn
• [Asterisk-Users] RxFax/TxFax - Compile Problem   Doug Lytle
• [Asterisk-Users] Cisco Ip phones   Michiel van Baak
• [Asterisk-Users] HooDaHek 0.6 Released   Nathan Pralle
• [Asterisk-Users] [ANNOUNCE] chan_capi-cm-0.6 released   Armin Schindler
• [Asterisk-Users] ODBC Voicemail WEB Retrieval   Liu Peter
• [Asterisk-Users] Re: how to distinguish the "ringing" and "connected"for zap channel   Liu Peter
• [SPAM:***** SpamScore] Re: [SPAM:******** SpamScore] [Asterisk-Users] Call Transfer using SIP clients   hugolivude
• [Asterisk-Users] Cisco Ip phones   Anders Svensson
• [Asterisk-Users] pri gateway   tim panton
• [Asterisk-Users] AstriCon 2006 Location   Kristian Kielhofner
• [Asterisk-Users] Snom-320 badly garbled audio   Darren Ellis
• [Asterisk-Users] Re: how to distinguish the "ringing" and "connected"for zap channel   Liu Peter
• [Asterisk-Users] Re: HooDaHek 0.6 Released   Nathan Pralle
• [Asterisk-Users] RxFax/TxFax - Compile Problem   José Pablo Ezequiel Fernández
• [Asterisk-Users] Re: HooDaHek 0.6 Released   Steven
• [Asterisk-Users] RxFax/TxFax - Compile Problem   José Pablo Ezequiel Fernández
• [Asterisk-Users] Cisco Ip phones   Michiel van Baak
• [Asterisk-Users] Multiple PCI cards   Matthew Fredrickson
• [Asterisk-Users] Asterisk vertical service activation codes   hugolivude
• [Asterisk-Users] MySQL and Asterisk   Dan Journo
• [Asterisk-Users] ${DIALSTATUS} problems   Joan Bautista
• [Asterisk-Users] [Fwd: ASTCC speaks and cut RTP channel, => Kind of solution...   Ricardo Poppi
• [Asterisk-Users] Snom-320 badly garbled audio   Torsten Krueger
• [Asterisk-Users] Cisco Ip phones   Sander
• [Asterisk-Users] Multiple PCI cards   Joan Bautista
• [Asterisk-Users] Cisco Ip phones   Sander
• [Asterisk-Users] HooDaHek 0.6 Released   Shawn Rutledge
• [Asterisk-Users] HooDaHek 0.6 Released   Nathan Pralle
• [Asterisk-Users] Differ between "private" and "out of area"?   Goran Dj
• [Asterisk-Users] Re: HooDaHek 0.6 Released   Nathan Pralle
• [Asterisk-Users] MySQL and Asterisk   Nathan Pralle
• [Asterisk-Users] MySQL and Asterisk   Matthew Boehm
• [Asterisk-Users] Cisco Ip phones   Michiel van Baak
• [Asterisk-Users] Snom-320 badly garbled audio   Christian Stredicke
• [Asterisk-Users] fixlocalprefix error   Chad Brown
• [Asterisk-Users] MySQL and Asterisk   Dan Journo
• [Asterisk-Users] Re: MySQL and Asterisk   Steven
• [Asterisk-Users] OT: Hardware Interrupts; Who is it?   steve at daviesfam.org
• [Asterisk-Users] ${DIALSTATUS} problems   Liu Peter
• [Asterisk-Users] TE110P hybrid configuration for data and voice   Damon Brown
• [Asterisk-Users] SUCCESS - 512 Simultaneous Calls with Digital Recording   Matt Roth
• [Asterisk-Users] MOH failures (bad quality with interruptions)   Yoann Le Bihan
• [Asterisk-Users] Monitor and sox mix quality   Jonathan Feally
• [Asterisk-Users] Re: MOH failures (bad quality with interruptions)   Yoann Le Bihan
• [Asterisk-Users] sipuras 841 bad sound   Anton Krall
• [Asterisk-Users] cvs-head and unicall with r2mfc   Anton Krall
• [Asterisk-Users] [ANNOUNCE] chan_capi-cm-0.6 released   Patrick
• [Asterisk-Users] Handling SIP 404 event   Shaun Ewing
• [Asterisk-Users] SUCCESS - 512 Simultaneous Calls with Digital Recording   Patrick
• [Asterisk-Users] [ANNOUNCE] chan_capi-cm-0.6 released   Shaun Ewing
• [Asterisk-Users] [ANNOUNCE] chan_capi-cm-0.6 released   Shaun Ewing
• [Asterisk-Users] SUCCESS - 512 Simultaneous Calls with Digital Recording   Michael Welter
• [Asterisk-Users] SUCCESS - 512 Simultaneous Calls with Digital Recording   Matt Roth
• [Asterisk-Users] ODBC VM Playback from Web Page   pbx at itsngroup.com
• [Asterisk-Users] HooDaHek w/AST 1.2   Technical Support
• [Asterisk-Users] SUCCESS - 512 Simultaneous Calls with Digital Recording   Matt Florell
• [Asterisk-Users] DIDx   Crystal Stream, Incorporated
• [Asterisk-Users] Is there a clever way to page a group of extensions?   Luki
• [Asterisk-Users] Can I connect an IAXy to my Panasonic PBX?   Raúl Gómez Cabrera
• [Asterisk-Users] HooDaHek w/AST 1.2   Nathan E. Pralle
• [Asterisk-Users] sipuras 841 bad sound   Juan Jose Comellas
• [Asterisk-Users] MWI indicator HINT on Snom thru IAX?   Luki
• [Asterisk-Users] ${DIALSTATUS} problems   Mark Edwards
• [Asterisk-Users] ${DIALSTATUS} problems   trixter http://www.0xdecafbad.com
• [Asterisk-Users] Anyone using Asterisk to take credit card payments?   FELIX E SKOWRONEK
• [Asterisk-Users] MWI indicator HINT on Snom thru IAX?   Colin Anderson
• [Asterisk-Users] automon wav format problems   Anton Krall
• [Asterisk-Users] Phone lines   Jennifer Hales
• [Asterisk-Users] Re: MusicOnHold not working   Gurminder Arora
• [Asterisk-Users] Re: MySQL and Asterisk   Dan Journo
• [Asterisk-Users] iax2 trunking wackyness   Clive
• [Asterisk-Users] Asterisk PBX   kapil dhawan
• [Asterisk-Users] [ANNOUNCE] chan_capi-cm-0.6 released   Armin Schindler
• [Asterisk-Users] Asterisk PBX   Sahil Gupta
• [Asterisk-Users] SUCCESS - 512 Simultaneous Calls with Digital Recording   Zoa
• [Asterisk-Users] Cisco Ip phones   Florian Overkamp
• [Asterisk-Users] iax2 trunking wackyness   Matt Riddell
• [Asterisk-Users] Brand New IPSwitchBoard   Thorben Jensen
• [Asterisk-Users] SUCCESS - 512 Simultaneous Calls with Digital Recording   trixter http://www.0xdecafbad.com
• [Asterisk-Users] Re: HooDaHek 0.6 Released   Tony Mountifield
• [Asterisk-Users] SUCCESS - 512 Simultaneous Calls with Digital Recording   Zoa
• [Asterisk-Users] DID problem with calls from analog to ISDN   Christian Gansberger
• [Asterisk-Users] Call getting disconnected in queue   Rajkumar S
• [Asterisk-Users] SUCCESS - 512 Simultaneous Calls with Digital Recording   trixter http://www.0xdecafbad.com
• [Asterisk-Users] SUCCESS - 512 Simultaneous Calls with Digital Recording   Zoa
• [Asterisk-Users] Intermitant delays on call setup.   Morgan Gilroy
• [Asterisk-Users] How to retrieve voicemail from an IP phone?   Ryan Pagquil
• [Asterisk-Users] How to retrieve voicemail from an IP phone?   Ryan Pagquil
• [Asterisk-Users] How to retrieve voicemail from an IP phone?   Rudolf Ladyzhenskii
• [Asterisk-Users] How to retrieve voicemail from an IP phone?   Pisac
• [Asterisk-Users] iax2 trunking wackyness   Clive
• [Asterisk-Users] How to retrieve voicemail from an IP phone?   Rich Adamson
• [Asterisk-Users] IAX2 registration   Christoph Eicke
• [Asterisk-Users] First release of the Asteriskguru Operator Panel   Zoa
• [Asterisk-Users] Call Forward - 7940 Asterisk - Help   Tom Rymes
• [Asterisk-Users] Asterisk PBX   Tom Rymes
• [Asterisk-Users] Asterisk PBX   Francesco Peeters
• [Asterisk-Users] Asterisk PBX   Francesco Peeters
• [Asterisk-Users] Asterisk PBX   Nathan Pralle
• [Asterisk-Users] Asterisk PBX   Francesco Peeters
• [Asterisk-Users] iax2 trunking wackyness   Andrew Kohlsmith
• [Asterisk-Users] Need a test number in NE US that I can call   Paul
• [Asterisk-Users] How to retrieve voicemail from an IP phone?   Rich Adamson
• [Asterisk-Users] add 0 (zero) to incoming callerID - how?   Tomasz Chmielewski
• [Asterisk-Users] Packetization period for CODECs   kurt x
• [Asterisk-Users] permit syntax question   Min Qiu
• [Asterisk-Users] Using *0 to flash an external trunk on bridged channel   hugolivude
• [Asterisk-Users] Ask for config files of Nortell Meridian Op 11 & Asterisk for PRI   Alvaro Parres
• [Asterisk-Users] maximum concurrent ZAP channels .... max conf ports ...   Vamsi Pottangi
• [Asterisk-Users] HELP: E1 ChannelBank and UniCall   Paradise Dove
• [Asterisk-Users] Cellphones and Asterisk Bluetooth   chawki hammoud
• [Asterisk-Users] add 0 (zero) to incoming callerID - how?   Nathan Pralle
• [Asterisk-Users] iax2 trunking wackyness   Matt Riddell
• [Asterisk-Users] ISDN-forwarding to intern without cost?   Oliver Rath
• [Asterisk-Users] add 0 (zero) to incoming callerID - how?   Matt Riddell
• [Asterisk-Users] Does Asterisk know if the trunks are busy?   Steven
• [Asterisk-Users] Caller ID and Call Parking on an analog PSTN line?   Brian McEntire
• [Asterisk-Users] Caller ID and Call Parking on an analog PSTN line?   Kevin Bockman
• [Asterisk-Users] Caller ID and Call Parking on an analog PSTN line?   Nathan Pralle
• [Asterisk-Users] sipuras 841 bad sound   Jason Williams
• [Asterisk-Users] Caller ID and Call Parking on an analog PSTN line?   John Novack
• [Asterisk-Users] qualify=yes   Manny A. Wise
• [Asterisk-Users] Does Asterisk know if the trunks are busy?   Kevin Bockman
• [Asterisk-Users] Macro exists if an application returned -1   Armin Schindler
• [Asterisk-Users] Problem with Queues   Crystal Stream, Incorporated
• [Asterisk-Users] Caller ID and Call Parking on an analog PSTN line?   Brian McEntire
• [Asterisk-Users] Addendum to Problem with Queues question   Crystal Stream, Incorporated
• [Asterisk-Users] HOWTO: A simple AGI application to modify incomi ng CallerID on the fly using SQL Server and *not* UnixODBC   Colin Anderson
• [Asterisk-Users] ISDN-forwarding to intern without cost?   Joerg Lauer
• [Asterisk-Users] Macro exists if an application returned -1   Thorsten Lockert
• [Asterisk-Users] Help with asterisk-oh323 driver   Fernando Herrera
• [Asterisk-Users] Problem with Queues   Kevin Bockman
• [Asterisk-Users] ODBC Voicemail WEB Retrieval V1.1   pbx at itsngroup.com
• [Asterisk-Users] Addendum to Problem with Queues question   Kevin Bockman
• [Asterisk-Users] is possible connect?   Pablo Allietti
• [Asterisk-Users] sipuras 841 bad sound   Anton Krall
• [Asterisk-Users] Problem with Queues   Crystal Stream, Incorporated
• [Asterisk-Users] SUCCESS - 512 Simultaneous Calls with Digital Recording   Matt Roth
• [Asterisk-Users] Macro exists if an application returned -1   Armin Schindler
• [Asterisk-Users] presence settings and Eyebeam   Kevin Hanson
• [Asterisk-Users] SUCCESS - 512 Simultaneous Calls with Digital Recording   Matt Hess
• [Asterisk-Users] Weird Over Lapping Asterisk Calls via SIP Phones   asterisk at emptyhole.net
• [Asterisk-Users] How can i call to a cellphone here in Mexico?   Claudio Canseco
• [Asterisk-Users] iax2 trunking wackyness   Adam Robins
• [Asterisk-Users] add 0 (zero) to incoming callerID - how?   Michiel van Baak
• [Asterisk-Users] iax2 trunking wackyness   Andrew Kohlsmith
• [Asterisk-Users] oh323 driver and RFC2833   Fernando Herrera
• [Asterisk-Users] SUCCESS - 512 Simultaneous Calls with Digital Recording   info at beprojects.com
• [Asterisk-Users] SUCCESS - 512 Simultaneous Calls with Digital Recording   Zoa
• [Asterisk-Users] Get SIP to work over very limited network access   Benjamin Lawetz
• [Asterisk-Users] VM low volume - testers needed   Rich Adamson
• [Asterisk-Users] re: Problems with Queues   Crystal Stream, Incorporated
• [Asterisk-Users] How to retrieve voicemail from an IP phone?   Min Qiu
• [Asterisk-Users] problem with monitor meetme   Michal Misiak
• [Asterisk-Users] SUCCESS - 512 Simultaneous Calls with Digital Recording   Matt Roth
• [Asterisk-Users] Problems with sipura 1001's and 2002's   Phil Allred
• [Asterisk-Users] Re: Re: how to distinguish the "ringing" and"connected"for zap channel   Alchaemist
• [Asterisk-Users] IAX2 vs SIP Phones and adapters   canuck15
• [Asterisk-Users] Re: Re: how to distinguish the "ringing" and"connected"for zap channel   Alchaemist
• [Asterisk-Users] VM low volume - testers needed   Rich Adamson
• [Asterisk-Users] Re: [Asterisk-Dev] maximum concurrent ZAP channels .... max conf ports ...   Matthew Fredrickson
• [Asterisk-Users] SUCCESS - 512 Simultaneous Calls with Digital Recording   Zoa
• [Asterisk-Users] Problem with monitor application meetme   Michal Misiak
• [Asterisk-Users] Get SIP to work over very limited network access   Sebastian A. Espindola
• [Asterisk-Users] Asterisk and a SPA3000 behind NAT peer registration   Alchaemist
• [Asterisk-Users] How to retrieve voicemail from an IP phone?   John Crowhurst
• [Asterisk-Users] Problem with meetme monitor (recording)   Michal Misiak
• [Asterisk-Users] How can i call to a cellphone here in Mexico?   Alex Kauffmann
• [Asterisk-Users] Callprogress and TDM400 in Brasil   Ricardo Poppi
• [Asterisk-Users] Problem with meetme monitor (recording)   Matt Florell
• [Asterisk-Users] POP3 and TTS (Festival?)   Alchaemist
• [Asterisk-Users] How to retrieve voicemail from an IP phone?   Min Qiu
• [Asterisk-Users] How can i call to a cellphone here in Mexico?   Claudio Canseco
• [Asterisk-Users] Pinging ...   Steve Totaro
• [Asterisk-Users] Weird Over Lapping Asterisk Calls via SIP Phones   Chris Travers
• [Asterisk-Users] Asterisk Platform - Success Strories - iAreanet in the news.   Kanuri, Seshu (Company IT)
• [Asterisk-Users] Asterisk and a SPA3000 behind NAT peer registration   Rich Adamson
• [Asterisk-Users] How to retrieve voicemail from an IP phone?   Min Qiu
• [Asterisk-Users] WMI problem   Il Neofita
• [Asterisk-Users] How can i call to a cellphone here in Mexico?   Rene Kluwen
• [Asterisk-Users] Re: How can i call to a cellphone here in Mexico?   Alchaemist
• [Asterisk-Users] Re: How can i call to a cellphone here in Mexico?   Alchaemist
• [Asterisk-Users] Re: How can i call to a cellphone here in Mexico?   Claudio Canseco
• [Asterisk-Users] POP3 and TTS (Festival?)   Bartosz Jozwiak
• [Asterisk-Users] Re: How can i call to a cellphone here in Mexico?   Don Dawson
• [Asterisk-Users] Re: How can i call to a cellphone here in Mexico?   Claudio Canseco
• [Asterisk-Users] Soyo Phones Crashing   Ilan Rabinovitch
• [Asterisk-Users] Tux/Asterisk logo for Cisco phones   Mark Phillips
• [Asterisk-Users] automon wav format problems   Anton Krall
• [Asterisk-Users] cvs-head and unicall with r2mfc   Anton Krall
• [Asterisk-Users] SoundPoint IP Attendant Console   Bartosz Jozwiak
• [Asterisk-Users] Tux/Asterisk logo for Cisco phones   Cory Andrews
• [Asterisk-Users] How to retrieve voicemail from an IP phone?   Ryan Pagquil
• [Asterisk-Users] Tux/Asterisk logo for Cisco phones   David Mallwitz
• [Asterisk-Users] Re: [Asterisk-Dev] maximum concurrent ZAP channels .... max confports ...   Matthew Fredrickson
• [Asterisk-Users] Pinging ...   Alexander Lopez
• [Asterisk-Users] ftp.soft-switch.org down?   Anton Krall
• [Asterisk-Users] ftp.soft-switch.org down?   trixter http://www.0xdecafbad.com
• [Asterisk-Users] Cisco AS5XXX + CallerID Name   Max Braz
• [Asterisk-Users] SoundPoint IP Attendant Console   Cory Andrews
• [Asterisk-Users] SoundPoint IP Attendant Console   Kevin P. Fleming
• [Asterisk-Users] I got "403", "Forbidden"... please help   Ryan Pagquil
• [Asterisk-Users] SUCCESS - 512 Simultaneous Calls with Digital Recording   Matt Florell
• [Asterisk-Users] ftp.soft-switch.org down?   Anton Krall
• [Asterisk-Users] new spandsp-0.0.3pre1 missing tx and rx fax apps?   Anton Krall
• [Asterisk-Users] Asterisk and a SPA3000 behind NAT peerregistration   Maps
• [Asterisk-Users] iax2 trunking wackyness   Matt Riddell
• [Asterisk-Users] SUCCESS - 512 Simultaneous Calls with Digital Recording   izo
• [Asterisk-Users] How to retrieve voicemail from an IP phone?   brett at websmyths.com
• [Asterisk-Users] AstriCon 2006 Location   Paul Hales
• [Asterisk-Users] Re: Asterisk and a SPA3000 behind NATpeerregistration   Alchaemist
• [Asterisk-Users] SUCCESS - 512 Simultaneous Calls with Digital Recording   Matt Florell
• [Asterisk-Users] AstriCon 2006 Location   Anton Krall
• [Asterisk-Users] Re: Get SIP to work over very limited network access   Alchaemist
• [Asterisk-Users] Web based application for call History   Pradeepa Ramamurthy
• [Asterisk-Users] add 0 (zero) to incoming callerID - how?   Matt Riddell
• [Asterisk-Users] Web based application for call History   Jesse Keating
• [Asterisk-Users] POP3 and TTS (Festival?)   Michiel van Baak
• [Asterisk-Users] add 0 (zero) to incoming callerID - how?   Michiel van Baak
• [Asterisk-Users] Monitor and sox mix quality   Niklas Larsson
• [Asterisk-Users] hints and the sNOM 360   Shanon Swafford
• [Asterisk-Users] Complete NPA-NXX list for USA/Canada npanxx,   Joe Greco
• [Asterisk-Users] SUCCESS - 512 Simultaneous Calls with Digital Recording   lenz
• [Asterisk-Users] hints and the sNOM 360   Shanon Swafford
• [Asterisk-Users] Submitting ISDN-MSN from a SIP-Phone   asterisk-users at freinet.de
• [Asterisk-Users] add 0 (zero) to incoming callerID - how?   Matt Riddell
• [Asterisk-Users] Any problems with Asterisk and "nice"   Chris Modesitt
• [Asterisk-Users] Compile problems on Solaris SPARC   Joseph Rothstein
• [Asterisk-Users] Any problems with Asterisk and "nice"   Matt Riddell
• [Asterisk-Users] Web based application for call History   Tzafrir Cohen
• [Asterisk-Users] Compile problems on Solaris SPARC   Tzafrir Cohen
• [Asterisk-Users] Submitting ISDN-MSN from a SIP-Phone   Armin Schindler
• [Asterisk-Users] add 0 (zero) to incoming callerID - how?   Bruno De Luca
• [Asterisk-Users] oh323 driver and RFC2833   Michael Manousos
• [Asterisk-Users] Re: Asterisk and a SPA3000 behindNATpeerregistration   razza
• [Asterisk-Users] Re: BT100 and BETA 1.0.7.11   Dias Badekas
• [Asterisk-Users] I got "403", "Forbidden"... please help   harry gaillac
• [Asterisk-Users] add 0 (zero) to incoming callerID - how?   Bruno De Luca
• [Asterisk-Users] Early Media with Asterisk   Hauke Zuehl
• [Asterisk-Users] add 0 (zero) to incoming callerID - how?   Matt Riddell
• [Asterisk-Users] ftp.soft-switch.org down?   Rob Thomas
• [Asterisk-Users] Early Media with Asterisk   Andreas Sikkema
• [Asterisk-Users] Early Media with Asterisk   Hauke Zuehl
• [Asterisk-Users] Submitting ISDN-MSN from a SIP-Phone   Rene Kluwen
• [Asterisk-Users] IAX client for Linux text console   Jan Buchal
• [Asterisk-Users] SOHO Survey / Creative Asterisk Solutions   Brian McEntire
• [Asterisk-Users] IAX client for Linux text console   Filippo Carone
• [Asterisk-Users] AgentRecord In and Out streams   Crystal Stream, Incorporated
• [Asterisk-Users] Submitting ISDN-MSN from a SIP-Phone   Rich Adamson
• [Asterisk-Users] Asterisk with iptel.org   Sebastian Milioto
• [Asterisk-Users] Call Pickup issue   taf taffey
• [Asterisk-Users] Initial release of AMPortal Debian/Xorcom-Rapid packages   Tzafrir Cohen
• [Asterisk-Users] MS Live Communication Server   richard Coco
• [Asterisk-Users] ASTCC error when using silent=5   Ricardo Poppi
• [Asterisk-Users] Multiple SIP Phone Calls Overlapping on the Same Phone   Asterisk Mail
• [Asterisk-Users] ftp.soft-switch.org down?   Anton Krall
• [Asterisk-Users] SUCCESS - 512 Simultaneous Calls with Digital Recording   Scott Laird
• [Asterisk-Users] SOHO Survey / Creative Asterisk Solutions   Damon Estep
• [Asterisk-Users] Hardware Recommendations for Junghanns card QuadBRI PCI.   Dpto. Técnico.
• [Asterisk-Users] Asterisk with iptel.org   Steve Totaro
• [Asterisk-Users] Re: POP3 and TTS (Festival?)   Alchaemist
• [Asterisk-Users] AGI Script to interact with ACCESS Databse and Set CID info on the fly.   Tim King
• [Asterisk-Users] AGI Script to interact with ACCESS Databse a nd Set CID info on the fly.   Colin Anderson
• [Asterisk-Users] AGI Script to interact with ACCESS Databse and Set CID info on the fly.   Paul
• [Asterisk-Users] custom ring tone   Marko Rakar
• [Asterisk-Users] AGI Script to interact with ACCESS Databse and Set CID info on the fly.   Tom Hayden
• [Asterisk-Users] AGI Script to interact with ACCESS Databse and Set CID info on the fly.   pbx at itsngroup.com
• [Asterisk-Users] cdr_custom?   Sherwood McGowan
• [Asterisk-Users] Re: Re: Asterisk and a SPA3000behindNATpeerregistration   Alchaemist
• [Asterisk-Users] externpass   Sebastian Kühner
• [Asterisk-Users] SOHO Survey / Creative Asterisk Solutions   Brian McEntire
• [Asterisk-Users] Re: Asterisk with iptel.org   Sebastian Milioto
• [Asterisk-Users] Cisco AS5XXX + CallerID Name   Benjamin Lawetz
• [Asterisk-Users] OT: Sangoma A102u available   Christian M. Watts
• [Asterisk-Users] custom ring tone   Matthew Simpson
• [Asterisk-Users] custom ring tone   Domjan Attila
• [Asterisk-Users] WaitExten   Andrew Nowrot
• [Asterisk-Users] SOHO Survey / Creative Asterisk Solutions   Tom Vile
• [Asterisk-Users] Polycom IP500 Quickstart page or files?   Wilson Pickett
• [Asterisk-Users] custom ring tone   Marko Rakar
• [Asterisk-Users] custom ring tone   Marko Rakar
• [Asterisk-Users] WaitExten   Sherwood McGowan
• [Asterisk-Users] externpass   Tzafrir Cohen
• [Asterisk-Users] Polycom IP500 Quickstart page or files?   Mojo with Horan & Company, LLC
• [Asterisk-Users] AgentRecord In and Out streams   Mojo with Horan & Company, LLC
• (solved) Re: [Asterisk-Users] ISDN-forwarding to intern without cost?   Oliver Rath
• [Asterisk-Users] Polycom IP500 Quickstart page or files?   Tom Hayden
• [Asterisk-Users] Cisco AS5XXX + CallerID Name   info at beprojects.com
• [Asterisk-Users] POP3 and TTS (Festival?)   Tzafrir Cohen
• [Asterisk-Users] Polycom IP500 Quickstart page or files?   Tom Vile
• [Asterisk-Users] logging in problem   prashant yadav
• [Asterisk-Users] Will Digium Wildard work with PCI-X or PCI Express   Chuck Bunn
• [Asterisk-Users] custom ring tone   Domjan Attila
• [Asterisk-Users] Will Digium Wildard work with PCI-X or PCI Express   Kevin Bockman
• [Asterisk-Users] custom ring tone   John Hill
• [Asterisk-Users] custom ring tone   Marko Rakar
• [Asterisk-Users] Cisco AS5XXX + CallerID Name   B. J. Bomar
• [Asterisk-Users] Call getting disconnected in queue   Rajkumar S
• [Asterisk-Users] SUCCESS - 512 Simultaneous Calls with Digital Recording   Waldo Rubinstein
• [Asterisk-Users] custom ring tone   John Novack
• [Asterisk-Users] custom ring tone   Mojo with Horan & Company, LLC
• [Asterisk-Users] Polycom IP500 Quickstart page or files?   Andres Paglayan
• [Asterisk-Users] Tux/Asterisk logo for Cisco phones   Mark Phillips
• [Asterisk-Users] custom ring tone   Marko Rakar
• [Asterisk-Users] custom ring tone   Colin Anderson
• [Asterisk-Users] SUCCESS - 512 Simultaneous Calls with Digital Recording   Matt Florell
• [Asterisk-Users] custom ring tone   Fernando Herrera
• [Asterisk-Users] custom ring tone   John Hill
• [Asterisk-Users] custom ring tone   John Hill
• [Asterisk-Users] Will Digium Wildard work with PCI-X or PCI Express   Matt Roth
• [Asterisk-Users] custom ring tone   trixter http://www.0xdecafbad.com
• [Asterisk-Users] custom ring tone   trixter http://www.0xdecafbad.com
• [Asterisk-Users] custom ring tone   John Hill
• [Asterisk-Users] externpass   Sebastian Kühner
• [Asterisk-Users] Will Digium Wildard work with PCI-X or PCI Express   Matt Roth
• [Asterisk-Users] Set Log Level for Messages log file   Arik Funke
• [Asterisk-Users] custom ring tone   Fernando Herrera
• [Asterisk-Users] rtp problems   Anton Krall
• [Asterisk-Users] Set Log Level for Messages log file   Jesse Keating
• [Asterisk-Users] bristuff-0.2.0RC8m   Arik Funke
• [Asterisk-Users] SUCCESS - 512 Simultaneous Calls with Digital Recording   steve at daviesfam.org
• [Asterisk-Users] Get SIP to work over very limited network access   steve at daviesfam.org
• [Asterisk-Users] Windows Clients problem   Mostafa
• [Asterisk-Users] problems with sending fax from SIP channels   Andy Kuo
• [Asterisk-Users] priindication passthru TE410P EuroISDN?   Bruno Voigt
• [Asterisk-Users] Codec conversion   Asterisk guy
• [Asterisk-Users] Re: Set Log Level for Messages log file   Arik Funke
• [Asterisk-Users] SNOM 190 '486/Busy here' after upgrade to firmwa re 3.60s   Colin Anderson
• [Asterisk-Users] Extended SIP registration failures   Mark Hulber
• [Asterisk-Users] SUCCESS - 512 Simultaneous Calls with Digital Recording   Matt Roth
• [Asterisk-Users] Re: Set Log Level for Messages log file   Jesse Keating
• [Asterisk-Users] Re: Set Log Level for Messages log file   Arik Funke
• [Asterisk-Users] Tux/Asterisk logo for Cisco phones   David McNett
• [Asterisk-Users] [Fwd: ASTCC speaks and cut RTP channel, => Kind of solution...   Darren Wiebe
• [Asterisk-Users] Re: How can i call to a cellphone here in Mexico?   Claudio Canseco
• [Asterisk-Users] custom ring tone   Alex Vishnev
• [Asterisk-Users] anyone know about this company? www.blue-wireless.net   Dean Collins
• [Asterisk-Users] custom ring tone   trixter http://www.0xdecafbad.com
• [Asterisk-Users] custom ring tone   Alex Vishnev
• [Asterisk-Users] Recently reported ASTCC audio issues   Darren Wiebe
• [Asterisk-Users] SUCCESS - 512 Simultaneous Calls with Digital Recording   Matt Florell
• [Asterisk-Users] Sip Groups   Tracy Peek
• [Asterisk-Users] anyone know about this company? www.blue-wireless.net   C. Hatton Humphrey
• [Asterisk-Users] SayUnixTime in CVS?   Brian McEntire
• [Asterisk-Users] SOHO Survey / Creative Asterisk Solutions   Brian McEntire
• [Asterisk-Users] CVS-HEAD and Caller ID -- Pulling my hair out!   pbx at itsngroup.com
• [Asterisk-Users] Sip Groups   Kevin Bockman
• [Asterisk-Users] Re: SayUnixTime in CVS?   Brian McEntire
• [Asterisk-Users] Recently reported ASTCC audio issues   Darren Wiebe
• [Asterisk-Users] sipuras 841 bad sound   Alex Ternero
• [Asterisk-Users] externpass   Tzafrir Cohen
• [Asterisk-Users] Asterisk + GNUGK + Asterisk-Addons ooh323   Brian C. Fertig
• [Asterisk-Users] SNOM 190 '486/Busy here' after upgrade to firmware 3.60s   Christian Stredicke
• [Asterisk-Users] Problems with sipura 1001's and 2002's   Peter Wemm
• [Asterisk-Users] Asterisk + GNUGK + Asterisk-Addons ooh323   brett at websmyths.com
• [Asterisk-Users] custom ring tone   Marko Rakar
• [Asterisk-Users] CPU load   Anders Svensson
• [Asterisk-Users] Keytouch without effect   Hans-Peter Straub
• [Asterisk-Users] Hangup when dial via Mobile Interface   Thawat Mohchit
• [Asterisk-Users] sipuras 841 bad sound   Anton Krall
• [Asterisk-Users] SIP exten to PSTN calls   Nil S
• [Asterisk-Users] SIP exten to PSTN calls   Nil S
• [Asterisk-Users] Polycom IP500 Quickstart page or files?   Wilson Pickett
• [Asterisk-Users] CPU load   Jean-Michel Hiver
• [Asterisk-Users] Fax detection question   Rudolf Ladyzhenskii
• [Asterisk-Users] zaphfc problem: overlapdial don't work after update bristuff   tuned at fan.priv.at
• [Asterisk-Users] Dial() and BackGround()   Christian Schoepplein
• [Asterisk-Users] Which codec?   Dan Journo
• [Asterisk-Users] Which codec?   Anders Svensson
• [Asterisk-Users] Which codec?   Christoph Eicke
• [Asterisk-Users] Problem setting up TDM22B card   somesh s
• [Asterisk-Users] Dial multiple phones   kurt x
• [Asterisk-Users] hints and the sNOM 360   asterisk-Users at zybercity.dk
• [Asterisk-Users] Dial multiple phones   Jörg Wolf
• [Asterisk-Users] SS7 support ?   Usman
• [Asterisk-Users] Re: Problem setting up TDM22B card   somesh s
• [Asterisk-Users] Re: Get SIP to work over very limited network access   Tzafrir Cohen
• [Asterisk-Users] Dial() and BackGround()   Patrick
• [Asterisk-Users] Re: Problem setting up TDM22B card   Rob Thomas
• [Asterisk-Users] Dial() and BackGround()   Christian Schoepplein
• R: [Asterisk-Users] Problem setting up TDM22B card   Fabrizio Mazzoni
• R: [Asterisk-Users] Problem setting up TDM22B card   Fabrizio Mazzoni
• [Asterisk-Users] Queues   Sander
• [Asterisk-Users] SS7 support ?   Domjan Attila
• [Asterisk-Users] Re: Problem setting up TDM22B card   somesh s
• R: [Asterisk-Users] Problem setting up TDM22B card   somesh s
• [Asterisk-Users] chan_capi-cm-0.6: hangup is detected really late   Tobias Wolf
• [Asterisk-Users] Dial() and BackGround()   Doug Lytle
• [Asterisk-Users] Queues   Sergio Serrano
• [Asterisk-Users] SOHO Survey / Creative Asterisk Solutions   Paul Zimm
• [Asterisk-Users] Re: Problem setting up TDM22B card   Martin Allen
• [Asterisk-Users] Queues   Doug Lytle
• [Asterisk-Users] zaphfc problem: overlapdial don't work after update bristuff   Johann Steinwendtner
• [Asterisk-Users] Fax detection question   gincantalupo
• [Asterisk-Users] Re: Problem setting up TDM22B card   Tzafrir Cohen
• [Asterisk-Users] Dialtone problems with phpagi and asterisk   Michael Häberle
• [Asterisk-Users] Re: Problem setting up TDM22B card   Dave Cotton
• [Asterisk-Users] Dial() and BackGround()   Christian Schoepplein
• [Asterisk-Users] Double cpu   Anders Svensson
• [Asterisk-Users] Re: Recently reported ASTCC audio issues   Ricardo Poppi
• [Asterisk-Users] chan_capi-cm-0.6: hangup is detected really late   Armin Schindler
• [Asterisk-Users] dial (iax/X&sip/y) get y fraction earlier   Sjaak Nabuurs
• [Asterisk-Users] ztdummy compile again   Mark Benson
• [Asterisk-Users] SIP Hangup via Call Files   Paul.Walling at trendcomms.com
• [Asterisk-Users] Re: Problem setting up TDM22B card   somesh s
• [Asterisk-Users] Cisco AS5XXX + CallerID Name   Braz
• [Asterisk-Users] Queues   Sander
• [Asterisk-Users] Re: Problem setting up TDM22B card   somesh s
• [Asterisk-Users] ztdummy compile again   Kevin Collins
• R: [Asterisk-Users] Problem setting up TDM22B card   John Novack
• [Asterisk-Users] Asterisk + GNUGK + Asterisk-Addons ooh323   Brian C. Fertig
• [Asterisk-Users] DTMF translation   Iqbal
• [Asterisk-Users] RE: SNOM 190 '486/Busy here' after upgrade to re 3.60s   Colin Anderson
• [Asterisk-Users] SIP exten to PSTN calls   Appan KH
• [Asterisk-Users] Execute php agi after channel hangup   Arik Funke
• [Asterisk-Users] Re: Problem setting up TDM22B card   Steve Totaro
• [Asterisk-Users] SOHO Survey / Creative Asterisk Solutions   Steve Totaro
• [Asterisk-Users] Execute php agi after channel hangup   Zoa
• [Asterisk-Users] Execute php agi after channel hangup   Kevin Bockman
• [Asterisk-Users] Problems with queue and remote agents   asterisk at frameweb.it
• [Asterisk-Users] Problems with queue and remote agents   Damon Estep
• [Asterisk-Users] ztdummy compile again   Mark Benson
• [Asterisk-Users] Problem with outbound calls   ihsane MOUTAIB
• [Asterisk-Users] zaphfc problem: overlapdial don't work after update bristuff   tuned at fan.priv.at
• [Asterisk-Users] voicemail operation modification   Bruce Ferrell
• [Asterisk-Users] retry times   Iqbal
• [Asterisk-Users] Trunks greyed-out on Flash Operator Panel?   Doug
• [Asterisk-Users] ZAP ISDN losing digits   maka
• [Asterisk-Users] retry times   Kevin Bockman
• [Asterisk-Users] Double cpu   Brian McEntire
• [Asterisk-Users] Double cpu   Tzafrir Cohen
• [Asterisk-Users] ChanSpy performance sub-optimal   Trey Blancher
• [Asterisk-Users] voicetronix openline4 comments   Patrick Fortin
• [Asterisk-Users] ChanSpy performance sub-optimal   Trey Blancher
• [Asterisk-Users] ztdummy compile again   Kevin Collins
• [Asterisk-Users] Polycom IP500 Quickstart page or files?   Wilson Pickett
• [Asterisk-Users] Problem setting up TDM22B card   Tim King
• [Asterisk-Users] goiax expanded with free us domestic calling   Matthew Simpson
• [Asterisk-Users] ZAP ISDN losing digits   Rene Kluwen
• [Asterisk-Users] Polycom IP500 Quickstart page or files?   Doug
• [Asterisk-Users] Continue dialtone after pressing 9   Brian McEntire
• [Asterisk-Users] asterisk invitation problem   jonny hashem
• [Asterisk-Users] CallerID issue   Adam Moffett
• [Asterisk-Users] Continue dialtone after pressing 9   Jesse Keating
• [Asterisk-Users] CallerID issue   Doug Lytle
• [Asterisk-Users] Call Queue ANI   kurt x
• [Asterisk-Users] Continue dialtone after pressing 9   Brian McEntire
• [Asterisk-Users] Can't receive Faxes with Asterisk (help)   adiaz0 at gmail.com
• [Asterisk-Users] Removing "-" (Dash) from Dialed Numbers   Scott Miller
• [Asterisk-Users] Help on RealTime Extensions on Oracle DB   Juan Salas
• [Asterisk-Users] Removing "-" (Dash) from Dialed Numbers   Ron Wellsted
• [Asterisk-Users] CallerID issue   Adam Moffett
• [Asterisk-Users] Removing "-" (Dash) from Dialed Numbers   Colin Anderson
• [Asterisk-Users] CallerID issue   Adam Moffett
• [Asterisk-Users] Re: goiax expanded with free us domestic calling   Steven
• [Asterisk-Users] Asterisk - Dying Signal 11   Sahil Gupta
• [Asterisk-Users] Asterisk CMD MySQL   Tim King
• [Asterisk-Users] Asterisk CMD MySQL   Waldo Rubinstein
• [Asterisk-Users] asterisk invitation problem   chawki hammoud
• [Asterisk-Users] asterisk invitation problem   chawki hammoud
• [Asterisk-Users] SUCCESS - 512 Simultaneous Calls with Digital Recording   Waldo Rubinstein
• [Asterisk-Users] Continue dialtone after pressing 9   Rene Kluwen
• [Asterisk-Users] Play sound on connect   Mir
• [Asterisk-Users] Monitor command volume   José Pablo Ezequiel Fernández
• [Asterisk-Users] CallerID issue   Rich Adamson
• [Asterisk-Users] FW: channel offhook state   Jacqueline Lee
• [Asterisk-Users] X-Lit not picking up callgroup call with *8   Robert G. Ristroph
• [Asterisk-Users] SOHO Survey / Creative Asterisk Solutions   Steve Edwards
• [Asterisk-Users] CallerID issue   Rich Adamson
• [Asterisk-Users] MeetMe Application - Empty Conference room problem   niles at atheos.net
• [Asterisk-Users] RE: [Asterisk-Dev] Open source time card application for Asterisk   John Todd
• [Asterisk-Users] Re: goiax expanded with free us domestic calling   Matthew Simpson
• [Asterisk-Users] DTMF detection problems.   Carlos Alberto Hastenreiter Assumpção
• [Asterisk-Users] dial (iax/X&sip/y) get y fraction earlier   Don Pobanz
• [Asterisk-Users] RE: [Asterisk-Dev] Open source time card application for Asterisk   Paul
• [Asterisk-Users] Wildcard TE110P in Mexico   Jorge Cisneros
• [Asterisk-Users] context question   trixter http://www.0xdecafbad.com
• [Asterisk-Users] BRI Hunting, using both channels on one msn   gw at adcomcorp.com
• [Asterisk-Users] Orinoco Injectors   Corey S. McFadden
• [Asterisk-Users] Wildcard TE110P in Mexico   Alex Kauffmann
• [Asterisk-Users] Orinoco Injectors   Darren Wright
• [Asterisk-Users] Play sound on connect   Mathew McKernan
• [Asterisk-Users] Can't receive Faxes with Asterisk (help)   Chris
• [Asterisk-Users] context question   info at beprojects.com
• [Asterisk-Users] context question   trixter http://www.0xdecafbad.com
• [Asterisk-Users] Skye gateway?   mgraves at mstvp.com
• [Asterisk-Users] delay SIP answer   Mark Willis
• [Asterisk-Users] VM low volume - testers needed   Brian McEntire
• [Asterisk-Users] Is background() fax detect broken?   gw at adcomcorp.com
• [Asterisk-Users] Message Waiting Indicator (MWI) for remote voice mail?   Brian McEntire
• [Asterisk-Users] Do Sifira use Asterisk?   Gurminder Arora
• [Asterisk-Users] Problems with queue and remote agents   lenz
• [Asterisk-Users] wrong password on authentication for INVITE to '"asterisk"   chawki hammoud
• [Asterisk-Users] SS7 support ?   Usman
• [Asterisk-Users] VM low volume - testers needed   Rich Adamson
• [Asterisk-Users] SS7 support ?   Stefan de Konink
• [Asterisk-Users] wrong password on authentication for INVITE to '"asterisk"   Rich Adamson
• [Asterisk-Users] Seperate siptrunks   Anders Svensson
• [Asterisk-Users] wrong password on authentication for INVITE to '"asterisk"   chawki hammoud
• [Asterisk-Users] HP DL360 G4 EM64T and hyperthreading options   Eric Bishop
• [Asterisk-Users] context question   Alex Vishnev
• [Asterisk-Users] Will Digium Wildard work with PCI-X or PCI Express   Chuck Bunn
• [Asterisk-Users] BT100 can't register   Neil Cherry
• [Asterisk-Users] Help!! trying to use an MTA   Calvin Lockhart
• [Asterisk-Users] Asterisk returns 484 ADDRESS INCOMPLETE for incoming SIP calls   Frank Tarczynski
• [Asterisk-Users] dial (iax/X&sip/y) get y fraction earlier   Luki
• [Asterisk-Users] VM low volume - testers needed   Brian McEntire
• [Asterisk-Users] VM low volume - testers needed   Julian Lyndon-Smith
• [Asterisk-Users] unable to use misdn group dial   Dias Badekas
• [Asterisk-Users] Need Help on Areski Calling Card Solution plz   ADEGOKE ARUNA
• [Asterisk-Users] context question   trixter http://www.0xdecafbad.com
• [Asterisk-Users] ASTCC on Fedora 4 and MySQL 4.1.12   Scott Wolfe
• [Asterisk-Users] BRI Hunting, using both channels on one msn   Armin Schindler
• [Asterisk-Users] ASTCC on Fedora 4 and MySQL 4.1.12   Darren Wiebe
• [Asterisk-Users] CallerID issue   Doug Lytle
• [Asterisk-Users] Send DTMF after call bridge   Mohammed Salim
• [Asterisk-Users] VM low volume - testers needed   Brian McEntire
• [Asterisk-Users] wrong password on authentication for INVITE to '"asterisk"   chawki hammoud
• [Asterisk-Users] Play sound on connect   Mir
• [Asterisk-Users] Send DTMF after call bridge   Alvaro G. M.
• [Asterisk-Users] wrong password on authentication for INVITE to '"asterisk"   Rich Adamson
• [Asterisk-Users] Directed pickup syntax?   Rich Adamson
• [Asterisk-Users] Falsh Panel in Xorcom Rapid   Mike Matthews
• [Asterisk-Users] VM low volume - testers needed   Brian McEntire
• [Asterisk-Users] Directed pickup syntax?   Alexander Lopez
• [Asterisk-Users] Play sound on connect   AbdelRahman Tarzi
• [Asterisk-Users] Pictures from VON Fall 2005 Digium/Asterisk booth   Kevin P. Fleming
• [Asterisk-Users] ASTCC on Fedora 4 and MySQL 4.1.12   Darren Wiebe
• [Asterisk-Users] Directed pickup syntax?   Joshua Colp - Asterlink
• [Asterisk-Users] PA1688 Phones using IAX MWI   brett at websmyths.com
• [Asterisk-Users] IBM x306   Marco Supino
• [Asterisk-Users] IBM x306   Alexander Lopez
• [Asterisk-Users] IBM x306   Stefan de Konink
• [Asterisk-Users] IBM x306   Marco Supino
• [Asterisk-Users] IBM x306   Marco Supino
• [Asterisk-Users] Need good explanation on contexts and extensions   telephony at jonjay.com
• [Asterisk-Users] Software to generate an SRTP key pair?   telephony at jonjay.com
• [Asterisk-Users] Cheap Time sources which is best?   Steve Gladden
• [Asterisk-Users] Directed pickup syntax?   Rich Adamson
• [Asterisk-Users] CDR problem   FaberK
• [Asterisk-Users] CDR problem   Darren Wiebe
• [Asterisk-Users] SoundPoint IP Attendant Console   Chris Coulthurst
• [Asterisk-Users] IBM x306   Tzafrir Cohen
• [Asterisk-Users] Cheap Time sources which is best?   Tzafrir Cohen
• [Asterisk-Users] IPSpeedDial has just been released   Thorben Jensen
• [Asterisk-Users] ASTCC on Fedora 4 and MySQL 4.1.12   Scott Wolfe
• [Asterisk-Users] didgium card in india   Capt MS
• [Asterisk-Users] dialplan game   trixter http://www.0xdecafbad.com
• [Asterisk-Users] didgium card in india   Sahil Gupta
• [Asterisk-Users] Re: goiax expanded with free us domestic calling   Andy Hamilton
• [Asterisk-Users] didgium card in india   Capt MS
• [Asterisk-Users] Extension Mobility (roaming) Cisco 7960   Sascha Ferley
• [Asterisk-Users] dialplan game   Tzafrir Cohen
• [Asterisk-Users] dialplan game   trixter http://www.0xdecafbad.com
• [Asterisk-Users] Need good explanation on contexts and extensions   Leif Madsen
• [Asterisk-Users] dialplan game   trixter http://www.0xdecafbad.com
• [Asterisk-Users] Extension Mobility (roaming) Cisco 7960   Michiel van Baak
• [Asterisk-Users] Codec routing?   Anders Svensson
• [Asterisk-Users] SoundPoint IP Attendant Console   Matthew Gibson
• [Asterisk-Users] Play sound on connect   Simone Cittadini
• [Asterisk-Users] didgium card in india   Bashir Ullah
• [Asterisk-Users] Need Help on Areski Calling Card Solution plz   chawki hammoud
• [Asterisk-Users] Best Voip provider   Nana Tandoh
• [Asterisk-Users] Best Voip provider   Anders Svensson
• [Asterisk-Users] Best Voip provider   trixter http://www.0xdecafbad.com
• [Asterisk-Users] iax problem   Piotr Chytla
• [Asterisk-Users] Need Help on Areski Calling Card Solution plz   Junaid Uppal
• [Asterisk-Users] DTMF issue on IVR   larry lin
• [Asterisk-Users] iax problem   Rich Adamson
• [Asterisk-Users] Problem Asterisk: can't make call but can receive calls   Jean-Yves Avenard
• [Asterisk-Users] Skye gateway?   Antonio José dos Santos Brandão
• [Asterisk-Users] pound/hash key not recognized   test at bigbox.ch
• [Asterisk-Users] Cheap Time sources which is best?   Steve Gladden
• [Asterisk-Users] Best Voip provider   Leif Madsen
• [Asterisk-Users] Codec routing?   William Lloyd
• [Asterisk-Users] Send DTMF after call bridge   Mohammed Salim
• [Asterisk-Users] Help!! trying to use an MTA   Nana Tandoh
• [Asterisk-Users] VPB Driver Question   Ian Bonham
• [Asterisk-Users] CALLERID to Sipura Devices (or others for that matter).. CVS-Latest Version   pbx at itsngroup.com
• [Asterisk-Users] Pager Notification Script   Tom Rymes
• [Asterisk-Users] Best Voip provider   Leif Madsen
• [Asterisk-Users] Cheap Time sources which is best?   Tzafrir Cohen
• [Asterisk-Users] Best Voip provider   trixter http://www.0xdecafbad.com
• [Asterisk-Users] Wildcard TE110P in Mexico   Alvaro Parres
• [Asterisk-Users] WRT54GP2 SIP server on LAN port   Johannes
• [Asterisk-Users] Best Voip provider   Brian McEntire
• [Asterisk-Users] Best Voip provider   Shawn Rutledge
• [Asterisk-Users] Best Voip provider   Chris
• [Asterisk-Users] Best Voip provider   Anders Svensson
• [Asterisk-Users] Best Voip provider   Leif Madsen
• [Asterisk-Users] WRT54GP2 SIP server on LAN port   Tom Vile
• [Asterisk-Users] WRT54GP2 SIP server on LAN port   Sherwood McGowan
• [Asterisk-Users] Re: Pager Notification Script   Steven
• [Asterisk-Users] Clicks in audio with TE100P PRI   Rod Bacon
• [Asterisk-Users] Clicks in audio with TE100P PRI   Alejandro G
• [Asterisk-Users] Cisco phone ports   Keith Schmidt
• [Asterisk-Users] Digium T-1 and FXO cards for sale   Tracy R Reed
• [Asterisk-Users] Unable to Transfer an outbound call   hugolivude
• [Asterisk-Users] TE405P V2 changes?   Rod Bacon
• [Asterisk-Users] Vonage-type service   Waldo Rubinstein
• [Asterisk-Users] Best Voip provider   Scott Wolfe
• [Asterisk-Users] Emergency Asterisk Guru help needed -- Yucky sound with MOH   Jeffrey Starin
• [Asterisk-Users] IBM x306   Steve Totaro
• [Asterisk-Users] Best Voip provider   Jaime Lopez
• [Asterisk-Users] Best Voip provider   Chris
• [Asterisk-Users] Vonage-type service   Alexander Lopez
• [Asterisk-Users] Extension Mobility (roaming) Cisco 7960   Alexander Lopez
• [Asterisk-Users] TE405P V2 - Fantastic!   Rod Bacon
• [Asterisk-Users] TE405P V2 - Fantastic!   Kevin P. Fleming
• [Asterisk-Users] TE405P V2 - Fantastic!   Jean-Yves Avenard
• [Asterisk-Users] Vonage-type service   Tom Vile
• [Asterisk-Users] TE405P V2 - Fantastic!   John covici
• [Asterisk-Users] Can an outside caller dial an extension before someone answer?   Simon Glass
• [Asterisk-Users] compute traffic intensity from CDR?   Jim Gottlieb
• R: [Asterisk-Users] Problem setting up TDM22B card   somesh s
• [Asterisk-Users] Problem setting up TDM22B card   somesh s
• [Asterisk-Users] Re: Problem setting up TDM22B card   somesh s
• [Asterisk-Users] Skye gateway?   Steve Daniels
• [Asterisk-Users] Can an outside caller dial an extension before someone answer?   trixter http://www.0xdecafbad.com
• [Asterisk-Users] change codec based on callerid (sip/iax)   trixter http://www.0xdecafbad.com
• [Asterisk-Users] Re: Problem setting up TDM22B card   somesh s
• [Asterisk-Users] context question   Bruno De Luca
• [Asterisk-Users] VOIP in Japan using Freebit   Pikoro
• [Asterisk-Users] IBM x306   George Pajari
• [Asterisk-Users] MS Live Communication Server   richard Coco
• [Asterisk-Users] context question   trixter http://www.0xdecafbad.com
• [Asterisk-Users] ztdummy compile again   Mark Benson
• [Asterisk-Users] IBM x306   Sergio Serrano
• [Asterisk-Users] IAX Registry problems   Sander
• [Asterisk-Users] sip, call ransfer and call waiting   Daniel ANDRE
• [Asterisk-Users] Best Voip provider   Sherwood McGowan
• [Asterisk-Users] sip, call ransfer and call waiting   trixter http://www.0xdecafbad.com
• [Asterisk-Users] Vonage-type service   Sherwood McGowan
• [Asterisk-Users] Date based context inclusion   Alessio Focardi
• [Asterisk-Users] TE405P V2 - Fantastic!   Rich Adamson
• [Asterisk-Users] Date based context inclusion   Sander
• [Asterisk-Users] Satellite Broadband and VOIP   Mark Anthony C. Delfin
• [Asterisk-Users] Satellite Broadband and VOIP   Anders Svensson
• [Asterisk-Users] dialing selected text with asterisk under windows ...   Gerd Mueller
• [Asterisk-Users] TE405P V2 - Fantastic!   Andrew Kohlsmith
• [Asterisk-Users] TE405P V2 - Fantastic!   Andrew Kohlsmith
• [Asterisk-Users] Asterisk returns 484 ADDRESS INCOMPLETE for incoming SIP calls   marley.tarczynski.com
• [Asterisk-Users] Asterisk returns 484 ADDRESS INCOMPLETE for incoming SIP calls   Frank Tarczynski
• [Asterisk-Users] Will Digium Wildard work with PCI-Xor PCI Express   Lee Archer
• [Asterisk-Users] TE405P V2 - Fantastic!   Morten Isaksen
• [Asterisk-Users] Asterisk::AGI - What license ???   Jean-Michel Hiver
• [Asterisk-Users] Pager Notification Script   Joel Vandal
• [Asterisk-Users] Can't receive Faxes with Asterisk (help)   adiaz0 at gmail.com
• [Asterisk-Users] Best drivers for HFC-S ISDN cards   Chris Bagnall
• Subject: [Asterisk-Users] Vonage-type service   Federico Alves
• Subject: [Asterisk-Users] Vonage-type service   Yair Hakak
• [Asterisk-Users] Call Back On Busy?   Sherwood McGowan
• [Asterisk-Users] Re: goiax expanded with free us domestic calling   Matthew Simpson
• [Asterisk-Users] Asterisk - Dying Signal 11   Alberto Risco
• [Asterisk-Users] Call Back On Busy?   Damon Estep
• [Asterisk-Users] iax problem   Piotr Chytla
• [Asterisk-Users] Best drivers for HFC-S ISDN cards   Roy Sigurd Karlsbakk
• [Asterisk-Users] Can't receive Faxes with Asterisk (help)   adiaz0 at gmail.com
• [Asterisk-Users] Call Back On Busy?   Sherwood McGowan
• [Asterisk-Users] CheckGroup accross multiple servers   Benjamin Lawetz
• [Asterisk-Users] Best drivers for HFC-S ISDN cards   Armin Schindler
• [Asterisk-Users] Re: Best drivers for HFC-S ISDN cards   Stefan Tichy
• [Asterisk-Users] Can't receive Faxes with Asterisk (help)   Chris
• [Asterisk-Users] TE405P V2 - Fantastic!   Steve Totaro
• [Asterisk-Users] Asterisk and Broadvoice, no incoming voice   Jason Schafer
• [Asterisk-Users] Asterisk and Broadvoice, no incoming voice   Moises Silva
• [Asterisk-Users] Recent Sphinx integration work?   Joe Hosteny
• [Asterisk-Users] IAX Registry problems   Steve Totaro
• [Asterisk-Users] BRI ISDN on USB   amaury BOSSE
• [Asterisk-Users] Re: Best drivers for HFC-S ISDN cards   Vidar
• [Asterisk-Users] Re: Asterisk vertical service activation codes   hugolivude
• [Asterisk-Users] Asterisk and Broadvoice, no incoming voice   Jason Schafer
• [Asterisk-Users] FW: channel offhook state   Jacqueline Lee
• [Asterisk-Users] 5 way calling?   hugolivude
• [Asterisk-Users] Will Digium Wildard work with PCI-X or PCI Express   Matt Roth
• [Asterisk-Users] Can't receive Faxes with Asterisk (help)   Steve Totaro
• [Asterisk-Users] Asterisk to CCM   Brian J. Rathman
• [Asterisk-Users] FW: channel offhook state   Matt Love
• [Asterisk-Users] I want to send oH323 calls to our Quintum D3000 which is connected to a PSTN   Ade Agbero
• [Asterisk-Users] Re: Need Help on Areski Calling Card Solution plz   Goke Aruna
• [Asterisk-Users] 5 way calling?   Tom Rymes
• [Asterisk-Users] Pager Notification Script   Tom Rymes
• [Asterisk-Users] Asterisk and Broadvoice, no incoming voice   Jason Schafer
• [Asterisk-Users] Early Media in 100 Ringing   Ronald Voermans
• [Asterisk-Users] iax problem   Rich Adamson
• [Asterisk-Users] FW: channel offhook state   Andrew Kohlsmith
• [Asterisk-Users] Sangoma and Digium same machine?   William Lloyd
• [Asterisk-Users] Asterisk to CCM   Arnaldo M. Pereira
• [Asterisk-Users] Tux/Asterisk logo for Cisco phones   Ing CIP Alejandro Celi Mariátegui
• [Asterisk-Users] Re: goiax expanded with free us domestic calling   Steve Totaro
• [Asterisk-Users] FW: channel offhook state   Rich Adamson
• [Asterisk-Users] FW: channel offhook state   Faris Raouf
• [Asterisk-Users] change codec based on callerid (sip/iax)   Michael D Schelin
• [Asterisk-Users] I want to send oH323 calls to our Quintum D3000 which is connected to a PSTN   Daniel Varella de Oliveira
• [Asterisk-Users] Asterisk and Broadvoice, no incoming voice   Darren Wright
• [Asterisk-Users] goiax caller ID   Kevin Scott
• [Asterisk-Users] ? In CLI not working   John Hill
• [Asterisk-Users] Asterisk and Broadvoice, no incoming voice   Jason Schafer
• [Asterisk-Users] Can't receive Faxes with Asterisk (help)   Steve Totaro
• [Asterisk-Users] IptablesAsterisk   Andrea Bencini
• [Asterisk-Users] goiax caller ID   Matthew Simpson
• [Asterisk-Users] Early Media in 180 Ringing   Ronald Voermans
• [Asterisk-Users] Asterisk to CCM   Greg Oliver
• [Asterisk-Users] Sangoma and Digium same machine?   Matt Florell
• [Asterisk-Users] Extension availabilty   Joshua Laroff
• [Asterisk-Users] Re: Message Waiting Indicator (MWI) for remote voice mail?   Brian McEntire
• [Asterisk-Users] SUCCESS - 512 Simultaneous Calls with Digital Recording   Matt Roth
• [Asterisk-Users] WRT54GP2 SIP server on LAN port   Johannes
• [Asterisk-Users] Call Back On Busy?   Sherwood McGowan
• [Asterisk-Users] asterisk SMS and sprintpcs   Jerry Geis
• [Asterisk-Users] Re: VOIP in Japan using Freebit   Alchaemist
• [Asterisk-Users] Carrier Access - Access Bank I config   Time Bandit
• [Asterisk-Users] Re: sipuras 841 bad sound   alan
• [Asterisk-Users] Re: Ring requested on channel already in use   alan
• [Asterisk-Users] Extension availabilty   BJ Weschke
• [Asterisk-Users] Asterisk and Broadvoice, no incoming voice   Paul
• [Asterisk-Users] Extension availabilty   Sherwood McGowan
• [Asterisk-Users] Extension availabilty   Harald Holzer
• [Asterisk-Users] Call Back On Busy?   BJ Weschke
• [Asterisk-Users] CAS Question   Exciting
• [Asterisk-Users] didgium card in india   Rajkumar S
• [Asterisk-Users] Extension availabilty   Mojo with Horan & Company, LLC
• [Asterisk-Users] Extension availabilty   Harald Holzer
• [Asterisk-Users] ZAP ISDN losing digits   Andy Kuo
• [Asterisk-Users] asterisk SMS and sprintpcs   Tom Hayden
• [Asterisk-Users] Need Help on Areski Calling Card Solution plz   ADEGOKE ARUNA
• [Asterisk-Users] Removing "-" (Dash) from Dialed Numbers   Scott Miller
• [Asterisk-Users] Asterisk and Broadvoice, no incoming voice   Jason Schafer
• [Asterisk-Users] TE405P V2 - Fantastic!   Kevin P. Fleming
• [Asterisk-Users] Carrier Access - Access Bank I config   Darren Wright
• [Asterisk-Users] FSX/UK analogue Phone rings all the time   Asterisk
• [Asterisk-Users] change codec based on callerid (sip/iax)   trixter http://www.0xdecafbad.com
• [Asterisk-Users] Call Back On Busy?   Sherwood McGowan
• [Asterisk-Users] Asterisk and Broadvoice, no incoming voice   Greg Oliver
• [Asterisk-Users] IBM x306 - some progress   Marco Supino
• [Asterisk-Users] Asterisk and Broadvoice, no incoming voice   Paul
• [Asterisk-Users] Removing "-" (Dash) from Dialed Numbers   Nathan Pralle
• [Asterisk-Users] Asterisk + GNUGK + Asterisk-Addons ooh323   Martin Vit
• [Asterisk-Users] Dialogic Cards Will they be available to NON AsteriskBE   Ronald Hartmann
• [Asterisk-Users] change codec based on callerid (sip/iax)   Paul
• [Asterisk-Users] Best drivers for HFC-S ISDN cards   Tim Robinson
• [Asterisk-Users] Removing "-" (Dash) from Dialed Numbers   Chris Wade
• [Asterisk-Users] TE405P V2 - Fantastic!   Matthew Fredrickson
• [Asterisk-Users] Performance tuning on dual Xeon EM64T and x86_64 Linux   Eric Bishop
• [Asterisk-Users] Asterisk and Broadvoice, no incoming voice   Jason Schafer
• [Asterisk-Users] Areskicc LCR problem   Sam Tam
• [Asterisk-Users] Command "sip show objects"   Ezequiel A. Sculli
• [Asterisk-Users] how to connect two SIP channels   bman at host.sk
• [Asterisk-Users] Asterisk and Broadvoice, no incoming voice   Manny A. Wise
• [Asterisk-Users] Asterisk and Broadvoice, no incoming voice   Paul
• [Asterisk-Users] Grandstream 496 not working on cordless phone   Nana Tandoh
• [Asterisk-Users] ZapHFC Channel unavailable   Rene Kluwen
• [Asterisk-Users] Polycom Setup Questions   Matthew T. O'Connor
• [Asterisk-Users] What ISDN hardware would you recommend?   Francesco Peeters
• [Asterisk-Users] Asterisk Realtime.. : Unixodbc drivers   Juan Salas
• [Asterisk-Users] netappel   FaberK
• [Asterisk-Users] FSX/UK analogue Phone rings all the time   John Crowhurst
• [Asterisk-Users] voipbuster advise   FaberK
• [Asterisk-Users] AsteriskJava - Queue   Sebastian Silva
• [Asterisk-Users] CPU spiking with TDM400 cards fixed   Richard Scobie
• [Asterisk-Users] voipbuster advise   Don Fanning
• [Asterisk-Users] Socket 478 Motherboard for use with TDM400P   Brent August Torrenga
• [Asterisk-Users] TE110P Hanging up & sometimes not picking up on E&M T1   Sascha Deri
• [Asterisk-Users] Socket 478 Motherboard for use with TDM400P   Chris
• [Asterisk-Users] Faxing via a sip extension with a digium e1 card   Mark Armstrong
• [Asterisk-Users] system() app changed drastically! How do I use it now?   Jim Gottlieb
• R: [Asterisk-Users] Problem setting up TDM22B card   Patrick
• [Asterisk-Users] system() app changed drastically! How do I use itnow?   Alexander Lopez
• [Asterisk-Users] AsteriskJava - Queue   Alexander Lopez
• [Asterisk-Users] system() app changed drastically! How do I use it now?   Jim Gottlieb
• [Asterisk-Users] re: DTMF woes, continued   Esteban Guana-Jarrin
• [Asterisk-Users] system() app changed drastically! How do I useit now?   Alexander Lopez
• [Asterisk-Users] BRI Hunting, using both channels on one msn   gw at adcomcorp.com
• [Asterisk-Users] Flash Panal   Tommy Denton
• [Asterisk-Users] StripMSD or extension parser bug?   Jim Gottlieb
• [Asterisk-Users] asterisk fifo   rkvalmiki
• [Asterisk-Users] ICD with asterisk   rkvalmiki
• [Asterisk-Users] StripMSD or extension parser bug?   Alexander Lopez
• [Asterisk-Users] SPA-3000 and incoming faxes   Tim Litwiller
• [Asterisk-Users] SPA-3000 and incoming faxes   Mark Armstrong
• [Asterisk-Users] Teliax   Jason Schafer
• [Asterisk-Users] IAX provider w/Toronto & Detroit termination   Technical Support
• R: [Asterisk-Users] Problem setting up TDM22B card   Tzafrir Cohen
• [Asterisk-Users] re: DTMF woes, continued   Sherwood McGowan
• [Asterisk-Users] re: DTMF woes, continued   Kevin Scott
• [Asterisk-Users] SPA-3000 and incoming faxes   Joseph
• [Asterisk-Users] Bad FCS nightmare to Nortel SL100 with TE410P   Gil Kloepfer
• [Asterisk-Users] "Non-blocking" Dial (and other commands): is there a way?   Enzo Michelangeli
• [Asterisk-Users] Grandstream 496 not working on cordless phone   Dave Cotton
• [Asterisk-Users] What ISDN hardware would you recommend?   Armin Schindler
• [Asterisk-Users] sip, call ransfer and call waiting   Daniel ANDRE
• R: [Asterisk-Users] Problem setting up TDM22B card   somesh s
• [Asterisk-Users] Re: Problem setting up TDM22B card   somesh s
• [Asterisk-Users] Re: Problem setting up TDM22B card   somesh s
• [Asterisk-Users] Termcap missing (compile error [editline/libedit.a] Error 1)   f6hqz-m at hamwlan.net
• [Asterisk-Users] Removing "-" (Dash) from Dialed Numbers   Andreas Sikkema
• [Asterisk-Users] Termcap missing (compile error [editline/libedit.a] Error 1)   Tzafrir Cohen
• [Asterisk-Users] Termcap missing (compile error[editline/libedit.a] Error 1)   Sergio Serrano
• [Asterisk-Users] pbx_wilcalu.so: undefined symbol:   Pikoro
• [Asterisk-Users] IBM x306 - some progress   Nir Simionovich
• [Asterisk-Users] Early Media in 180 Ringing   Hauke Zuehl
• [Asterisk-Users] Integration with NMS AG-E1/T1   Exciting
• [Asterisk-Users] Initial release of AMPortal Debian/Xorcom-Rapidpackages   Jose Limeres
• [Asterisk-Users] Early Media in 180 Ringing   Ronald Voermans
• [Asterisk-Users] failed make install on Solaris 10   Joseph Rothstein
• R: [Asterisk-Users] Problem setting up TDM22B card   somesh s
• [Asterisk-Users] Listening for DTMF when dialling   Peter Spikings
• [Asterisk-Users] WRT54GP2 SIP server on LAN port   Johannes
• [Asterisk-Users] Listening for DTMF when dialling (sorry, accidentally sent the previous message too early!)   Peter Spikings
• [Asterisk-Users] I got "403", "Forbidden"... please help   Ryan Pagquil
• [Asterisk-Users] radius and *   Matt
• [Asterisk-Users] Best drivers for HFC-S ISDN cards   Chris Bagnall
• [Asterisk-Users] * Accounting with Oracle   René Enskat [Teamware GmbH]
• [Asterisk-Users] Turn off echo-cancellation when fax is detected?   Arne Morten Johansen
• R: [Asterisk-Users] Best drivers for HFC-S ISDN cards   Giordano Grandis
• [Asterisk-Users] Sangoma and Digium same machine?   Reid Forrest
• R: [Asterisk-Users] Best drivers for HFC-S ISDN cards   Giordano Grandis
• [Asterisk-Users] iax problem   Piotr Chytla
• [Asterisk-Users] Teliax   Rich Adamson
• [Asterisk-Users] wait before accepting the call   ChB
• [Asterisk-Users] Integration with NMS AG-E1/T1   BJ Weschke
• [Asterisk-Users] Teliax   Chris Mason (Lists)
• [Asterisk-Users] Re: [Asterisk-Dev] MS Live Communications Server   richard Coco
• [Asterisk-Users] Re: Dialtone problems with phpagi and asterisk   Michael Häberle
• [Asterisk-Users] Integration with NMS AG-E1/T1   Exciting
• [Asterisk-Users] IAX2 hard phone   Alberto Risco
• [Asterisk-Users] Integration with NMS AG-E1/T1   BJ Weschke
• [Asterisk-Users] Teliax   Chris
• [Asterisk-Users] Unable to create channel of type 'Zap'   Mona Meyer
• [Asterisk-Users] Re: Dialtone problems with phpagi and asterisk   Johann
• [Asterisk-Users] IBM x306 - some progress   Colin Anderson
• [Asterisk-Users] IAX2 hard phone   Kanuri, Seshu (Company IT)
• [Asterisk-Users] pbx_wilcalu.so: undefined symbol:   Kevin Bockman
• [Asterisk-Users] function LEN missing   Technical Support
• [Asterisk-Users] Early Media in 180 Ringing   Kevin Bockman
• [Asterisk-Users] 405 "Method Not Allowed" error   Tim, Tim Favorite, Favorite
• [Asterisk-Users] AsteriskJava - Queue   Sebastian Silva
• [Asterisk-Users] Moaning dog...   Rich Adamson
• [Asterisk-Users] Software only Asterisk PBX (commercial)   Matthew Crocker
• [Asterisk-Users] FW: channel offhook state   Doug Reid - Stormcorp
• [Asterisk-Users] iax problem   Rich Adamson
• [Asterisk-Users] Asterisk & European Digital CAS Help   Exciting
• [Asterisk-Users] Re: Dialtone problems with phpagi and asterisk   Michael Häberle
• [Asterisk-Users] Software only Asterisk PBX (commercial)   Ronald Hartmann
• [Asterisk-Users] IAX2 hard phone   f6hqz-m at hamwlan.net
• [Asterisk-Users] VoIP Buster stopped working?   Arik Funke
• [Asterisk-Users] How to change ${VM_DATE} in voicemail.conf   amaury BOSSE
• [Asterisk-Users] Extension availabilty   Wilson Pickett
• [Asterisk-Users] analogue phone with asterisk   Rajesh Bhairampally
• [Asterisk-Users] failed make install on Solaris 10   Rajesh Bhairampally
• [Asterisk-Users] Re: [Asterisk-Dev] MS Live Communications Server   Jacky
• [Asterisk-Users] [MSG]TDM Error on ASUS Pundit-R   Morgan Gilroy
• [Asterisk-Users] Software only Asterisk PBX (commercial)   Morgan Gilroy
• [Asterisk-Users] Polycom IP 500 - problem dialing extra numbers   Jachin Rupe
• [Asterisk-Users] Termcap missing (compile error[editline/libedit.a] Error 1)   f6hqz-m at hamwlan.net
• R: [Asterisk-Users] Problem setting up TDM22B card   tmassey at obscorp.com
• [Asterisk-Users] blindxfer & atxfer not working?   hugolivude
• [Asterisk-Users] One-way audio with VPN   Francesco Peeters
• [Asterisk-Users] Polycom IP 500 - problem dialing extra numbers   Dean Collins
• [Asterisk-Users] analogue phone with asterisk   Ed Greenberg
• [Asterisk-Users] Polycom IP 500 - problem dialing extra numbers   Matt Gibson
• [Asterisk-Users] analogue phone with asterisk   Francesco Peeters
• [Asterisk-Users] Software only Asterisk PBX (commercial)   Kanuri, Seshu (Company IT)
• [Asterisk-Users] One-way audio with VPN   Alex Lake
• [Asterisk-Users] asterisk@home inbound call problem to SIP trunk. (voipfone UK)   Steve Babb
• [Asterisk-Users] FSX/UK analogue Phone rings all the time   Tim Robinson
• [Asterisk-Users] SIP Tandem Inbound only.   Scott Eisert
• [Asterisk-Users] [MSG]TDM Error on ASUS Pundit-R   tmassey at obscorp.com
• [Asterisk-Users] What ISDN hardware would you recommend?   Tim Robinson
• [Asterisk-Users] SIP Tandem Inbound only.   Peter Bowyer
• R: [Asterisk-Users] Best drivers for HFC-S ISDN cards   Tim Robinson
• [Asterisk-Users] cgi-bin/vmail.cgi - - Invalid Context   dabigshiznizzle at sctelcom.net
• [Asterisk-Users] blindxfer & atxfer not working?   Mojo with Horan & Company, LLC
• [Asterisk-Users] Best drivers for HFC-S ISDN cards   Tim Robinson
• [Asterisk-Users] Polycom Setup Questions   Matthew T. O'Connor
• [Asterisk-Users] Review: Digium TE405P v2   Matt Florell
• [Asterisk-Users] SIP Tandem Inbound only.   Scott Eisert
• [Asterisk-Users] SIP Tandem Inbound only.   Joshua Colp - Asterlink
• [Asterisk-Users] Polycom Setup Questions   Matt Gibson
• [Asterisk-Users] Polycom Setup Questions   Doug
• [Asterisk-Users] Software only Asterisk PBX (commercial)   Tom Rymes
• [Asterisk-Users] AstriCon 2005 - Now With Free Beer!   Steven Sokol
• [Asterisk-Users] wait before accepting the call   Innocent Evil
• Subject: [Asterisk-Users] Vonage-type service   José Pablo Ezequiel Fernández
• [Asterisk-Users] Polycom Setup Questions   Jonathan k. Creasy
• [Asterisk-Users] Creating an OPX from a traditional PBX using Asterisk and a SIP device   Paul Conn
• [Asterisk-Users] asterisk SMS and sprintpcs   Kevin Hanson
• [Asterisk-Users] Hook Flapping on Cisco 7960   Matt
• [Asterisk-Users] Hook Flapping on Cisco 7960   Michiel van Baak
• [Asterisk-Users] Creating an OPX from a traditional PBX using Asterisk and a SIP device   Tom Rymes
• [Asterisk-Users] asterisk SMS and sprintpcs   Ric Moseley
• [Asterisk-Users] ChanSpy and Sipura 2100 jitter.   Trey Blancher
• [Asterisk-Users] Hook Flapping on Cisco 7960   Andy Hamilton
• [Asterisk-Users] Hook Flapping on Cisco 7960   Matt
• [Asterisk-Users] Moaning dog...   John Novack
• [Asterisk-Users] Canada VOIP provider quality   Technical Support
• [Asterisk-Users] ChanSpy and Sipura 2100 jitter.   Domjan Attila
• [Asterisk-Users] wait before accepting the call   ChB
• [Asterisk-Users] Echo after running for several days?   Rod Bacon
• [Asterisk-Users] Hook Flapping on Cisco 7960   Andy Hamilton
• [Asterisk-Users] IAX provider w/Toronto & Detroit termination   Paul Belanger
• [Asterisk-Users] Polycom IP 500 - problem dialing extra numbers   Jachin Rupe
• [Asterisk-Users] Hook Flapping on Cisco 7960   C F
• [Asterisk-Users] SIP Tandem Inbound only.   Scott Eisert
• [Asterisk-Users] Hook Flapping on Cisco 7960   Rich Adamson
• [Asterisk-Users] Canada VOIP provider quality   Leon Sun
• [Asterisk-Users] Extensions go straight to voicemail   Robin
• [Asterisk-Users] Re: [Asterisk-biz] Problem with sending fax from a SIP extension   C F
• [Asterisk-Users] Review: Digium TE405P v2   izo
• [Asterisk-Users] BAD echo problems with Sangoma and Telstra   Andrew Thrift
• [Asterisk-Users] Sipura 2000 Dial Plan   Michael Blood
• [Asterisk-Users] BAD echo problems with Sangoma and Telstra   Matt Florell
• [Asterisk-Users] Sipura 2000 Dial Plan   Andy Kuo
• [Asterisk-Users] polycom dialplan   Jonathan k. Creasy
• [Asterisk-Users] Sipura 2000 Dial Plan   Gonzalo Servat
• [Asterisk-Users] Canada VOIP provider quality   Marcel Eric Loiselle
• [Asterisk-Users] BAD echo problems with Sangoma and Telstra   Andrew Thrift
• [Asterisk-Users] 7960 show queue status   Eric Bishop
• [Asterisk-Users] MeetMe Application - Empty Conference room problem   niles at atheos.net
• [Asterisk-Users] Asterisk to CCM   Dinesh Birlasekaran
• [Asterisk-Users] linux dist. and kernel version   Bin Zhang
• [Asterisk-Users] IAX2 encryption of data packets?   Michael Jia
• [Asterisk-Users] Software only Asterisk PBX (commercial)   rkvalmiki
• [Asterisk-Users] Software only Asterisk PBX (commercial)   Dustin Wildes
• R: [Asterisk-Users] Problem setting up TDM22B card   somesh s
• [Asterisk-Users] Polycom Setup Questions   Matthew T. O'Connor
• [Asterisk-Users] Polycom Setup Questions   Matthew T. O'Connor
• [Asterisk-Users] Polycom Setup Questions   Matthew T. O'Connor
• [Asterisk-Users] oH323 Voice in one direction only   Ade Agbero
• [Asterisk-Users] Sangoma and Digium same machine?   altus
• [Asterisk-Users] Auto CallBack on busy   Abdul Ghafoor
• [Asterisk-Users] Auto CallBack on busy   Abdul Ghafoor
• [Asterisk-Users] Voice Encryption   Scott Huang
• [Asterisk-Users] Creating an OPX from a traditional PBX usingAsterisk and a SIP device   Paul Conn
• [Asterisk-Users] Voice Encryption   Michael Jia
• [Asterisk-Users] Review: Digium TE405P v2   Leif Madsen
• [Asterisk-Users] Auto CallBack on busy   Abdul Ghafoor
• [Asterisk-Users] analogue phone with asterisk   Christoph Eicke
• [Asterisk-Users] Canada VOIP provider quality   George Pajari
• [Asterisk-Users] Re: [Asterisk-Dev] MS Live Communications Server   richard Coco
• [Asterisk-Users] Voice Encryption   tim panton
• [Asterisk-Users] Canada VOIP provider quality   Jean-Michel Hiver
• [Asterisk-Users] Re: unable to use misdn group dial   Dias Badekas
• [Asterisk-Users] call wating and call transfer   Warren Burstein
• [Asterisk-Users] Problems with queue and remote agents   asterisk at frameweb.it
• [Asterisk-Users] MeetMe error   Fabio Montemaggiore
• SV: [Asterisk-Users] Turn off echo-cancellation when fax is detected?   Arne Morten Johansen
• [Asterisk-Users] oH323 Voice in one direction only   Kresimir Petrovic
• [Asterisk-Users] Queue/Agents   Mark Elkins
• [Asterisk-Users] MeetMe error   Paul Belanger
• [Asterisk-Users] Asterisk on windows   Kanishka Somaratne
• [Asterisk-Users] Software only Asterisk PBX (commercial)   Morgan Gilroy
• [Asterisk-Users] IAX2 encryption of data packets?   Mark Phillips
• [Asterisk-Users] Asterisk does not send "Setup acknowledge" on euroISDN E1   Pavel Petrov
• [Asterisk-Users] Asterisk in Production   Waldo Rubinstein
• [Asterisk-Users] Asterisk on windows   Christoph Eicke
• [Asterisk-Users] Asterisk on windows   Matt
• [Asterisk-Users] Queue/Agents   lenz
• [Asterisk-Users] Asterisk on windows   Olivier Taylor
• [Asterisk-Users] BAD echo problems with Sangoma and Telstra   Julien Goodwin
• [Asterisk-Users] problems accessing directory   asterisk at frameweb.it
• [Asterisk-Users] Asterisk on windows   Christoph Eicke
• [Asterisk-Users] setting up asterisk as an sms central?   Roy Sigurd Karlsbakk
• [Asterisk-Users] PSTN-GATEWAY   Reli Loin
• [Asterisk-Users] PSTN-GATEWAY   Brian C. Fertig
• [Asterisk-Users] setting up asterisk as an sms central?   Tom Hayden
• [Asterisk-Users] PSTN-GATEWAY   Brian C. Fertig
• [Asterisk-Users] Asterisk on windows   razza
• R: [Asterisk-Users] Problem setting up TDM22B card   John Novack
• [Asterisk-Users] [Asterisk-User] Does Asterisk just pass thru RTP if the codec is the same between two extension?   Jiang Jinke
• [Asterisk-Users] Does Asterisk just pass thru RTP if the codec is the same between two extension?   Jiang Jinke
• [Asterisk-Users] Does Asterisk just pass thru RTP if the codec is the same between two extensions?   Jiang Jinke
• [Asterisk-Users] Trying to cut out the paper work...   Etienne Pretorius
• [Asterisk-Users] Asterisk to CCM   Brian J. Rathman
• [Asterisk-Users] Asterisk in Production   Carlos Antunes
• [Asterisk-Users] Where MeetMe application   Fabio Montemaggiore
• [Asterisk-Users] Asterisk on windows   Neil Cherry
• [Asterisk-Users] digital receptionist pick up time   Ric Moseley
• [Asterisk-Users] Asterisk on windows   Rich Adamson
• [Asterisk-Users] adit 600 mgcp.conf   Tod Detre
• [Asterisk-Users] Asterisk on windows   Matt
• [Asterisk-Users] Where MeetMe application   Mark Hulber
• [Asterisk-Users] Asterisk on windows   Kanuri, Seshu (Company IT)
• [Asterisk-Users] Asterisk on windows   Steve Underwood
• [Asterisk-Users] Software only Asterisk PBX (commercial)   Kanuri, Seshu (Company IT)
• [Asterisk-Users] Asterisk Manager API   Ezequiel A. Sculli
• [Asterisk-Users] Roll back from CVS Head to v1.09   Mark Phillips
• [Asterisk-Users] How to change ${VM_DATE} in voicemail.conf   gincantalupo
• [Asterisk-Users] Asterisk on windows   Matt
• [Asterisk-Users] Asterisk on windows   Colin Anderson
• [Asterisk-Users] BAD echo problems with Sangoma and Telstra   canuck15
• [Asterisk-Users] Creating an OPX from a traditional PBX usingAsterisk and a SIP device   Tom Rymes
• [Asterisk-Users] Roll back from CVS Head to v1.09   Mark Phillips
• [Asterisk-Users] Roll back from CVS Head to v1.09   Kevin Bockman
• [Asterisk-Users] Reduce ring time to answer on Asterisk @Home 1.5   Matt Love
• [Asterisk-Users] Delay in dial   yusuf
• [Asterisk-Users] Asterisk Manager API   Moises Silva
• [Asterisk-Users] Roll back from CVS Head to v1.09   Alexander Lopez
• [Asterisk-Users] Asterisk on windows   canuck15
• [Asterisk-Users] How to change ${VM_DATE} in voicemail.conf   Nathan Pralle
• [Asterisk-Users] BAD echo problems with Sangoma and, Telstra   yusuf
• [Asterisk-Users] Sep 28 00:42:35 ERROR[5151] chan_zap.c: Unkn own signalling method 'pri_net'   Colin Anderson
• [Asterisk-Users] Asterisk on windows   Tony Hoyle
• [Asterisk-Users] Asterisk sound files, audio bandwidth, and sound quality   Stephen Bosch
• [Asterisk-Users] Asterisk on windows   Rich Adamson
• [Asterisk-Users] Correction: Asterisk sound files, audio bandwidth, and sound quality   Stephen Bosch
• [Asterisk-Users] DID's in CA, WA, BC, FL and NY   u
• [Asterisk-Users] TDM-400 cards, technical limitations   Stephen Bosch
• [Asterisk-Users] Monitor in AGI   Mir
• [Asterisk-Users] Software only Asterisk PBX (commercial)   Morgan Gilroy
• [Asterisk-Users] Asterisk on windows   Rich Adamson
• [Asterisk-Users] adit 600 mgcp.conf   Tod Detre
• [Asterisk-Users] ASTCC - INUSE Flag   Scott Wolfe
• [Asterisk-Users] Reduce ring time to answer on Asterisk @Home 1.5   Kevin Bockman
• [Asterisk-Users] asterisk SMS and sprintpcs   Kevin Hanson
• [Asterisk-Users] Asterisk on windows   Colin Anderson
• [Asterisk-Users] Asterisk sound files, audio bandwidth, and sound quality   Steve Underwood
• [Asterisk-Users] Asterisk on windows   canuck15
• [Asterisk-Users] Asterisk to CCM   Brian J. Rathman
• [Asterisk-Users] Asterisk to CCM   Brian J. Rathman
• [Asterisk-Users] Software only Asterisk PBX (commercial)   Kanuri, Seshu (Company IT)
• [Asterisk-Users] oH323 Voice in one direction only   Ade Agbero
• [Asterisk-Users] Asterisk on windows   Rich Adamson
• [Asterisk-Users] Polycom Setup Questions   Matt Gibson
• [Asterisk-Users] Polycom Setup Questions   Matt Gibson
• [Asterisk-Users] Queue/Agents   Kresimir Petrovic
• [Asterisk-Users] Zap FXO/FXS issues, 1.2.0-beta1   alan
• [Asterisk-Users] Music on Hold Quality   Justin Selleck
• [Asterisk-Users] Early Media in 100 Ringing   Joshua Colp - Asterlink
• [Asterisk-Users] Asterisk on windows   Justin Selleck
• [Asterisk-Users] Asterisk on windows   Kanuri, Seshu (Company IT)
• [Asterisk-Users] Sep 28 00:42:35 ERROR[5151] chan_zap.c: Unknown signalling method 'pri_net'   Steve Totaro
• [Asterisk-Users] Music on Hold Quality   Justin Selleck
• [Asterisk-Users] Delay in dial   Andy Kuo
• [Asterisk-Users] Sep 28 00:42:35 ERROR[5151] chan_zap.c: Unkn own signalling method 'pri_net'   Colin Anderson
• [Asterisk-Users] Asterisk on windows   Colin Anderson
• [Asterisk-Users] oH323 Voice in one direction only   Tom Rymes
• [Asterisk-Users] Tiny Echo on PSTN via Zaptel   Shaw Terwilliger
• [Asterisk-Users] Music on Hold Quality   Colin Anderson
• [Asterisk-Users] Asterisk on windows   Colin Anderson
• [Asterisk-Users] T.38 Faxing   Federico Alves
• [Asterisk-Users] Delay in dial   steve at daviesfam.org
• [Asterisk-Users] To get phone to ring in two or more places   Doug
• [Asterisk-Users] Remote call pickup   Damian Funnell
• [Asterisk-Users] Software only Asterisk PBX (commercial)   E Samuels
• [Asterisk-Users] Can I install latest oH323 on *@home   Ade Agbero
• [Asterisk-Users] oH323 Voice in one direction only   Bashir Ullah
• [Asterisk-Users] Zap FXO/FXS issues, 1.2.0-beta1   Rich Adamson
• [Asterisk-Users] Correction: Asterisk sound files, audio bandwidth, and sound quality   Kevin P. Fleming
• [Asterisk-Users] Linksys RT31P2 and TFTP   list
• [Asterisk-Users] Asterisk installation   Anders Svensson
• [Asterisk-Users] T.38 Faxing   Carlos Alperin
• [Asterisk-Users] Asterisk installation   Cory Andrews
• [Asterisk-Users] Sep 28 00:42:35 ERROR[5151] chan_zap.c: Unknown signalling method 'pri_net'   Steve Totaro
• [Asterisk-Users] Upgrading *   info at beprojects.com
• [Asterisk-Users] Dialogic Cards Will they be available to NON AsteriskBE   Kevin P. Fleming
• [Asterisk-Users] oH323 Voice in one direction only   Ade Agbero
• [Asterisk-Users] Asterisk does not send "Setup acknowledge" on euroISDN E1   Tim Robinson
• [Asterisk-Users] Sep 28 00:42:35 ERROR[5151] chan_zap.c: Unknown signalling method 'pri_net'   Steve Totaro
• [Asterisk-Users] Asterisk installation   Colin Anderson
• [Asterisk-Users] Asterisk sound files, audio bandwidth, and sound quality   Stephen Bosch
• [Asterisk-Users] Correction: Asterisk sound files, audio bandwidth, and sound quality   Stephen Bosch
• [Asterisk-Users] Diax -> * -> Diax delay   Shawn Porter
• [Asterisk-Users] Music on Hold Quality   canuck15
• [Asterisk-Users] Correction: Asterisk sound files, audio bandwidth, and sound quality   Kevin P. Fleming
• [Asterisk-Users] Asterisk installation   Anders Svensson
• [Asterisk-Users] Asterisk installation   Sherwood McGowan
• [Asterisk-Users] Voice Encryption   Michael Jia
• [Asterisk-Users] Asterisk installation   Anders Svensson
• [Asterisk-Users] Music on Hold Quality   Matt
• [Asterisk-Users] Correction: Asterisk sound files, audio bandwidth, and sound quality   steve at daviesfam.org
• [Asterisk-Users] T.38 Faxing   Roger Schreiter
• [Asterisk-Users] Re: Voice Encryption   Ramon F Herrera
• [Asterisk-Users] gsm audio files   adriano ghezzi
• [Asterisk-Users] asterisk 1.0.9 + spandsp 0.0.2pre20 = crash on boot   Matthew Crocker
• [Asterisk-Users] Re: [Asterisk-biz] Problem with sending fax from a SIP extension   u
• [Asterisk-Users] T.38 Faxing   Andy Kuo
• [Asterisk-Users] Motherboard for Digium card   amer karim
• [Asterisk-Users] Music on Hold Quality   Jorge Alayon
• [Asterisk-Users] T.38 Faxing   Lee Howard
• [Asterisk-Users] Correction: Asterisk sound files, audio bandwidth, and sound quality   Stephen Bosch
• [Asterisk-Users] Music on Hold Quality   Colin Anderson
• [Asterisk-Users] Motherboard for Digium card   Kevin Bockman
• [Asterisk-Users] T.38 Faxing   Roger Schreiter
• [Asterisk-Users] Correction: Asterisk sound files, audio bandwidth, and sound quality   Stephen Bosch
• [Asterisk-Users] T.38 Faxing   Carlos Alperin
• [Asterisk-Users] T.38 Faxing   Carlos Alperin
• [Asterisk-Users] T.38 Faxing   Roger Schreiter
• [Asterisk-Users] transfering calls, no ringing sent to caller   Jeremy Koski
• [Asterisk-Users] Asterisk@Home Questions   Michael Janofsky
• [Asterisk-Users] No audio non channels and choopy sound to PSTN network   Fabrizio Mazzoni
• [Asterisk-Users] T.38 Faxing   Lee Howard
• [Asterisk-Users] ISO SIP Based Conference Bridge Solution   Cory Andrews
• [Asterisk-Users] Asterisk@Home Questions   Wiley Siler
• [Asterisk-Users] cisco phones problems   Edwin Lam
• [Asterisk-Users] T.38 Faxing   Roger Schreiter
• [Asterisk-Users] cisco phones problems   Greg Oliver
• [Asterisk-Users] TE205P in loopback?   Franciraldo Cavalcante Junior
• [Asterisk-Users] cisco phones problems   Carlos Alperin
• [Asterisk-Users] Problem redirecting to voicemail through a SIP proxy (Looks like a bug)   Chris St Denis
• [Asterisk-Users] Re: Voice Encryption   Andy Hamilton
• [Asterisk-Users] cisco phones problems   Edwin Lam
• [Asterisk-Users] Music on Hold Quality   canuck15
• [Asterisk-Users] cisco phones problems   Edwin Lam
• [Asterisk-Users] TE205P in loopback?   Colin Anderson
• [Asterisk-Users] Correction: Asterisk sound files, audio bandwidth, and sound quality   Sherwood McGowan
• [Asterisk-Users] Asterisk and Broadvoice, no incoming voice   Mark Phillips
• [Asterisk-Users] chan_capi-cm, Euro ISDN bus: 2 extensions on same BRI port not working   huelbe_garcia at fastimap.com
• [Asterisk-Users] Does the 1.0.9 release contain the Broadvoice patches?   Mark Phillips
• [Asterisk-Users] TE205P in loopback?   Franciraldo Cavalcante Junior
• [Asterisk-Users] Does the 1.0.9 release contain the Broadvoice patches?   Samy Antoun
• [Asterisk-Users] asterisk 1.0.9 + spandsp 0.0.2pre20 = crash on boot   Joel Vandal
• [Asterisk-Users] T.38 Faxing   Steve Underwood
• [Asterisk-Users] Does the 1.0.9 release contain the Broadvoice patches?   Mark Phillips
• [Asterisk-Users] Re: [Asterisk-biz] Problem with sending fax from a SIP extension   C F
• [Asterisk-Users] TE205P in loopback?   Franciraldo Cavalcante Junior
• [Asterisk-Users] Voice Encryption   Scott Huang
• [Asterisk-Users] TE205P in loopback?   Steve Totaro
• [Asterisk-Users] Does the 1.0.9 release contain the Broadvoice patches?   Samy Antoun
• [Asterisk-Users] Changing language to spanish   Hector Elias Menjivar
• [Asterisk-Users] Recommended wireless router to run Asterisk on OpenWRT   Juan Jose Comellas
• [Asterisk-Users] TE205P in loopback?   Steve Totaro
• [Asterisk-Users] * mod core dump help   Matt
• [Asterisk-Users] chan_capi-cm, Euro ISDN bus: 2 extensions on same BRI port not working   gw at adcomcorp.com
• [Asterisk-Users] TE205P in loopback?   Steve Totaro
• [Asterisk-Users] Re: Delay in Dial   yusuf
• [Asterisk-Users] T.38 Faxing   Carlos Alperin
• [Asterisk-Users] cisco phones problems   Carlos Alperin
• [Asterisk-Users] Recording channels   Stephen Bosch
• [Asterisk-Users] T.38 Faxing   Rosario Pingaro
• [Asterisk-Users] How to change ${VM_DATE} in voicemail.conf   gincantalupo
• [Asterisk-Users] Is realtime meetme supported by Asterisk   Voice over IP
• [Asterisk-Users] chan_capi-cm, Euro ISDN bus: 2 extensions on same BRI port not working   Armin Schindler
• [Asterisk-Users] Dealt with IAreaNet before?   Sherwood McGowan
• [Asterisk-Users] Voice Prompts, what do you think? Good voice. Should we record a new prompt-set?   gw at adcomcorp.com
• [Asterisk-Users] zttest - 100% ?   Marco Supino
• [Asterisk-Users] Is realtime meetme supported by Asterisk   Hauke Zuehl
• [Asterisk-Users] PRI value   Giordano Grandis
• [Asterisk-Users] Re: Asterisk-Users Digest, Vol 14, Issue 178   kiran
• [Asterisk-Users] Calling voicemail from external phone.   Dave Cotton
• [Asterisk-Users] Calling voicemail from external phone.   Arne Morten Johansen
• [Asterisk-Users] Calling voicemail from external phone.   Carlos
• [Asterisk-Users] T.38 Faxing -> at astricon ?   Zoa
• [Asterisk-Users] T.38 Faxing -> at astricon ?   Rosario Pingaro
• [Asterisk-Users] T.38 Faxing -> at astricon ?   Kresimir Petrovic
• [Asterisk-Users] Re: Dialtone problems with phpagi and asterisk   Michael Häberle
• [Asterisk-Users] Don't call   Fabio Montemaggiore
• [Asterisk-Users] T.38 Faxing -> at astricon ?   Roger Schreiter
• [Asterisk-Users] digits won't play   Christoph Eicke
• [Asterisk-Users] Recording channels   Abdul Ghafoor
• [Asterisk-Users] setting up asterisk as an sms central?   Roy Sigurd Karlsbakk
• [Asterisk-Users] Calling voicemail from external phone.   Sherwood McGowan
• [Asterisk-Users] T.38 Faxing -> at astricon ?   Roy Sigurd Karlsbakk
• [Asterisk-Users] T.38 Faxing -> at astricon ?   Roy Sigurd Karlsbakk
• [Asterisk-Users] Variable in call parking   Andrew Nowrot
• [Asterisk-Users] digits won't play   Christoph Eicke
• [Asterisk-Users] chan_cap-cm-0.6 is not working for incomming calls   Bastian Schern
• [Asterisk-Users] Asterisk registering with vonage   S S
• [Asterisk-Users] sip calleridnum   Michal Olejnik
• [Asterisk-Users] chan_cap-cm-0.6 is not working for incomming calls   Armin Schindler
• [Asterisk-Users] Music on Hold Quality   Kim Culhan
• [Asterisk-Users] Calling voicemail from external phone.   Dean Collins
• [Asterisk-Users] Re: Dealt with IAreaNet before?   Steven
• [Asterisk-Users] Re: Calling voicemail from external phone.   Steven
• [Asterisk-Users] Major bug solved in IPSwitchBoard   Thorben Jensen
• [Asterisk-Users] soft phones for Zaurus PDA   Jason Schafer
• [Asterisk-Users] setting up asterisk as an sms central?   Tom Hayden
• [Asterisk-Users] asterisk 1.0.9 + spandsp 0.0.2pre20 = crash on boot   Matthew Crocker
• [Asterisk-Users] Tiny Echo on PSTN via Zaptel   Tony Nichols
• [Asterisk-Users] cisco phones problems   Leandro Tenorio
• [Asterisk-Users] Re: Calling voicemail from external phone.   Matt
• [Asterisk-Users] Audio Files, Filtering, and Formats for Asterisk   Sherwood McGowan
• [Asterisk-Users] Asterisk Echo problems, Urgent, please help,   Ian Bonham
• [Asterisk-Users] Asterisk Echo problems, Urgent, please help,   Tom Hayden
• [Asterisk-Users] Re: * mod core dump help   Gustavo A. Gonzalez
• [Asterisk-Users] chan_cap-cm-0.6 deflect support   Louis-David Mitterrand
• [Asterisk-Users] Re: Dealt with IAreaNet before?   Nir Simionovich
• [Asterisk-Users] maximum retries exceeded on call   Michael Häberle
• [Asterisk-Users] Change language to spanish   Hector Elias Menjivar
• [Asterisk-Users] Caller ID, Attended Transfers, Polycom   David Gomillion
• [Asterisk-Users] T.38 Faxing -> at astricon ?   Steve Underwood
• [Asterisk-Users] T.38 Faxing -> at astricon ?   Kresimir Petrovic
• [Asterisk-Users] Getting asterisk to send e-mail to mailbox-users   Arne Morten Johansen
• [Asterisk-Users] Getting asterisk to send e-mail to mailbox-users   Hector Elias Menjivar
• [Asterisk-Users] Remotely dialing calls from a polycom phone   Eric Lawman
• [Asterisk-Users] Asterisk Echo problems, Urgent, please help,   Ian Bonham
• [Asterisk-Users] T.38 Faxing -> at astricon ?   Zoa
• [Asterisk-Users] Re: [Asterisk-biz] Problem with sending fax froma SIP extension   Jonathan k. Creasy
• [Asterisk-Users] Prueba   Hector Elias Menjivar
• [Asterisk-Users] Change language to spanish   Moises Silva
• [Asterisk-Users] Getting asterisk to send e-mail to mailbox-users   Ian Bonham
• [Asterisk-Users] T.38 Faxing -> at astricon ?   Hector Elias Menjivar
• [Asterisk-Users] Re: [Asterisk-biz] Problem with sending fax froma SIP extension   Tom Hayden
• [Asterisk-Users] Getting asterisk to send e-mail to mailbox-users   Doug Lytle
• [Asterisk-Users] PRI value   Colin Anderson
• [Asterisk-Users] Remotely dialing calls from a polycom phone   Gary Reuter
• [Asterisk-Users] Change language to spanish   Hector Elias Menjivar
• [Asterisk-Users] OOH323C   Kanishka Somaratne
• [Asterisk-Users] T.38 Faxing -> at astricon ?   Roger Schreiter
• [Asterisk-Users] Cannot figure out why calls from my Asterisk appear to be from country code +34?   Angus Comber
• [Asterisk-Users] Asterisk Echo problems, Urgent, please help,   Matt
• R: [Asterisk-Users] PRI value   Giordano Grandis
• [Asterisk-Users] H323 and Asterisk   Kanishka Somaratne
• [Asterisk-Users] Cannot figure out why calls from my Asteriskappear to be fr   Ian Bonham
• [Asterisk-Users] Asterisk Echo problems, Urgent, please help,   Ian Bonham
• [Asterisk-Users] T.38 Faxing -> at astricon ?   Steve Underwood
• [Asterisk-Users] Re: Audio Files, Filtering, and Formats for Asterisk   Stephen Bosch
• [Asterisk-Users] soft phones for Zaurus PDA   William Suffill
• [Asterisk-Users] Tiny Echo on PSTN via Zaptel   Shaw Terwilliger
• [Asterisk-Users] OOH323C   Hans-Peter Straub
• [Asterisk-Users] call center software and asterisk   Bartosz Jozwiak
• R: [Asterisk-Users] PRI value   Jens Kübler
• [Asterisk-Users] PRI value   Colin Anderson
• [Asterisk-Users] T.38 Faxing -> at astricon ?   Lee Howard
• [Asterisk-Users] Correction: Asterisk sound files, audio bandwidth, and sound quality   steve at daviesfam.org
• [Asterisk-Users] call center software and asterisk   Nathan Pralle
• [Asterisk-Users] Variable in call parking   Damian Funnell
• [Asterisk-Users] call center software and asterisk   Sergio Serrano
• [Asterisk-Users] zttest - 100% ?   Damian Funnell
• [Asterisk-Users] T.38 Faxing -> at astricon ?   Craig Guy
• [Asterisk-Users] Remotely dialing calls from a polycom phone   Paul Davidson
• [Asterisk-Users] minor(? ) Grandstream phone issue   Bob Weber
• [Asterisk-Users] soft phones for Zaurus PDA   Leif Madsen
• [Asterisk-Users] zttest - 100% ?   Marco Supino
• [Asterisk-Users] TE205P in loopback?   Franciraldo Cavalcante Junior
• [Asterisk-Users] zttest - 100% ?   Carlos Antunes
• [Asterisk-Users] Cannot figure out why calls from myAsterisk appear to be fr   David J Carter
• [Asterisk-Users] T.38 Faxing -> at astricon ?   C F
• [Asterisk-Users] Tiny Echo on PSTN via Zaptel   Neil Lewis
• [Asterisk-Users] NAT   Anders Svensson
• [Asterisk-Users] Cannot figure out why calls from myAsteriskappear to be fr   Angus Comber
• [Asterisk-Users] Zap FXO/FXS issues, 1.2.0-beta1   alan
• [Asterisk-Users] Cannot figure out why calls from myAsteriskappear to be fr   Angus Comber
• [Asterisk-Users] DTMF tones from PSTN not reaching SIP device   Bruce Komito
• [Asterisk-Users] NAT   Doug
• [Asterisk-Users] Remotely dialing calls from a polycom phone   Eric Lawman
• [Asterisk-Users] Tiny Echo on PSTN via Zaptel   Shaw Terwilliger
• [Asterisk-Users] NAT   Anders Svensson
• [Asterisk-Users] Any way to not overwrite sound files on compile?   Matt
• [Asterisk-Users] Re: Voice Encryption   Scott Huang
• [Asterisk-Users] Polycom Setup Questions   Matthew T. O'Connor
• [Asterisk-Users] Unable to send fax using BroadVoice   Rene Nelson
• [Asterisk-Users] Unable to send fax using BroadVoice   Tom Vile
• [Asterisk-Users] Hardware Specifications   Dan Journo
• [Asterisk-Users] Unable to send fax using BroadVoice   Rene Nelson
• [Asterisk-Users] NAT   Samy Antoun
• [Asterisk-Users] Bad FCS nightmare to Nortel SL100 with TE410P   Gil Kloepfer
• [Asterisk-Users] H323 and Asterisk   Ade Agbero
• [Asterisk-Users] Unable to send fax using BroadVoice   Tom Vile
• [Asterisk-Users] minor(? ) Grandstream phone issue   Ade Agbero
• [Asterisk-Users] Unable to send fax using BroadVoice   Ariel Batista
• [Asterisk-Users] Hardware Specifications   José Pablo Ezequiel Fernández
• [Asterisk-Users] Asterisk as a Voice Logger alternative to NICE or Witness Systems   Machen, Matthew T.
• [Asterisk-Users] Hardware Specifications   Nathan Pralle
• [Asterisk-Users] NAT   Anders Svensson
• [Asterisk-Users] Polycom Setup Questions   Jonathan k. Creasy
• [Asterisk-Users] Asterisk for "Man-In-The-Middle" Trunk Side Call Recording?   Verlin Henderson
• [Asterisk-Users] Yada table in oracle   Han van Hulst
• [Asterisk-Users] Unable to send fax using BroadVoice   Rene Nelson
• [Asterisk-Users] FWD: '486 Busy here' and 'All Circuits are busy now'   Zeeshan
• [Asterisk-Users] NAT   Samy Antoun
• [Asterisk-Users] Problems using SIPURA and MFC/R2   Flávio Eduardo de Andrade
• [Asterisk-Users] Yada table in oracle   Han van Hulst
• [Asterisk-Users] NAT   Anders Svensson
• [Asterisk-Users] TDM40B - "Unable to play dialtone on channel X" ?   Derek Conniffe
• [Asterisk-Users] * mod core dump help   Alberto Risco
• [Asterisk-Users] minor(? ) Grandstream phone issue   Casey Boone
• [Asterisk-Users] Broadvoice inbound issues   Rene Nelson
• [Asterisk-Users] Cisco AS5300 --> [SIP] --> Asterisk - NO AUDIO   Steve Ducat
• [Asterisk-Users] FWD via Trunk from DMZ to LAN   michel vank
• [Asterisk-Users] Asterisk for "Man-In-The-Middle" Trunk Side Call Recording?   Matthew Crocker
• [Asterisk-Users] Broadvoice inbound issues   Klaus Sonnenleiter
• [Asterisk-Users] FWD: '486 Busy here' and 'All Circuits are busynow'   Hector Elias Menjivar
• [Asterisk-Users] Broadvoice inbound issues   Samy Antoun
• [Asterisk-Users] minor(? ) Grandstream phone issue   Shawn Rutledge
• [Asterisk-Users] Broadvoice inbound issues   Rene Nelson
• [Asterisk-Users] Auto Answer Fax   Rene Nelson
• [Asterisk-Users] Broadvoice inbound issues   michel vank
• [Asterisk-Users] Best echo canceller?   Claudio Canseco
• [Asterisk-Users] minor(? ) Grandstream phone issue   John Novack
• [Asterisk-Users] Auto Answer Fax   Scott Eisert
• [Asterisk-Users] Auto Answer Fax   Tom Hayden
• [Asterisk-Users] Tiny Echo on PSTN via Zaptel   Neil Lewis
• [Asterisk-Users] Best echo canceller?   Andrew Kohlsmith
• [Asterisk-Users] Best echo canceller?   Neil Lewis
• [Asterisk-Users] dtmfmode type   Joseph
• [Asterisk-Users] files conflict after CVS update   Claudio Canseco
• [Asterisk-Users] Asterisk Echo problems, Urgent, please help,   Matt
• [Asterisk-Users] Tiny Echo on PSTN via Zaptel   Shaw Terwilliger
• [Asterisk-Users] Re: * mod core dump help   Matt
• Mathematicians wanted (was RE: [Asterisk-Users] Best echo canceller?)   Kris Boutilier
• [Asterisk-Users] Re: * mod core dump help   Kevin Bockman
• [Asterisk-Users] Using Realtime queues and queue members   Tim Karl
• [Asterisk-Users] Meet me conferencing without blind transfers (Asterisk@home)   jennyw
• [Asterisk-Users] Using Realtime queues and queue members   William Boehlke
• [Asterisk-Users] Meet me conferencing without blind transfers (Asterisk@home)   Mojo with Horan & Company, LLC
• [Asterisk-Users] TDMoE + kernel badness   astgroups
• [Asterisk-Users] SIP Gateway wants T38, Asterisk rejects but media path not established.   Ray Van Dolson
• [Asterisk-Users] Voice Prompts, what do you think? Good voice.   gw at adcomcorp.com
• [Asterisk-Users] Meetme issue   niles at atheos.net
• [Asterisk-Users] Using Realtime queues and queue members   Kevin P. Fleming
• [Asterisk-Users] SIP Gateway wants T38, Asterisk rejects but media path not established.   Kevin P. Fleming
• [Asterisk-Users] Can't make outside call with SIP softphone   David H
• [Asterisk-Users] Is this normal?   Matthew T. O'Connor
• [Asterisk-Users] Is this normal?   Kevin Bockman
• [Asterisk-Users] Is this normal?   Mark Johnson
• [Asterisk-Users] OOH323C   Dan Austin
• [Asterisk-Users] please help on FreeTDS (writing CDR to MS-SQL or MySQL)   Andy Kuo
• [Asterisk-Users] Re: NOTICE[180235]: chan_zap.c:7358 pri_dchannel: PRI got event: 6 on Primary D-channel of span 1   Lance Grover
• [Asterisk-Users] files conflict after CVS update   Michiel van Baak
• R: [Asterisk-Users] Problem setting up TDM22B card   somesh s
• [Asterisk-Users] Any way to not overwrite sound files on compile?   gincantalupo
• [Asterisk-Users] [Asterisk-User] linux/Asterisk change ip address   Andrea Cristofanini - Gedam Europe Srl
• R: [Asterisk-Users] Problem setting up TDM22B card   somesh s
• [Asterisk-Users] zttest - 100% ?   DRi at b-w-computer.de
• R: [Asterisk-Users] PRI value   Peter Svensson
• [Asterisk-Users] * T.38 fax   Matt
• [Asterisk-Users] Siemens TC35 GSM gateway   Andrew Smith
• [Asterisk-Users] G.729 patent in France   Amaury BOSSE
• [Asterisk-Users] G.729 patent in France   Jean-Michel Hiver
• [Asterisk-Users] G.729 patent in France   Steve Kennedy
• [Asterisk-Users] Fritz, mISDN, Help   Derek Conniffe
• [Asterisk-Users] IAXPhone   Erwan DESVERGNES
• [Asterisk-Users] Empty ACK   Ronald Voermans
• [Asterisk-Users] Zaptel TDM questions   Chee Foong
• [Asterisk-Users] Don't call   Fabio Montemaggiore
• [Fwd: [Asterisk-Users] TDM40B - "Unable to play dialtone on channel X" ?]   Derek Conniffe
• [Asterisk-Users] Fritz, mISDN, Help   Jörg Wolf
• [Asterisk-Users] Fritz, mISDN, Help   Derek Conniffe
• [Asterisk-Users] Compile broken on FreeBSD ?   Kim Culhan
• [Asterisk-Users] Fritz, mISDN, Help   Derek Conniffe
• [Asterisk-Users] Diva   Giordano Grandis
• [Asterisk-Users] Fritz, mISDN, Help   Jörg Wolf
• [Asterisk-Users] chan_cap-cm-0.6 deflect support   Armin Schindler
• [Asterisk-Users] Why does the s extension not work in my extensions.conf file   Angus Comber
• [Asterisk-Users] Diva   Armin Schindler
• [Asterisk-Users] Zaptel TDM questions   Angus Comber
• [Asterisk-Users] chan_capi-0.3.5   Giordano Grandis
• [Asterisk-Users] Register times out on internet outage   Chris Mason (Lists)
• [Asterisk-Users] Don't call   Jason Walker
• [Asterisk-Users] VideoConference with UMTS   Sergio Serrano
• [Asterisk-Users] Don't call   Andreas Sikkema
• [Asterisk-Users] Register times out on internet outage   brett at websmyths.com
• [Asterisk-Users] OOH323C   brett at websmyths.com
• [Asterisk-Users] chan_capi-0.3.5   Jörg Wolf
• [Asterisk-Users] G.729 patent in France   Kevin Walsh
• [Asterisk-Users] Asterisk Echo problems, Urgent, please help,   Matt
• [Asterisk-Users] Any way to not overwrite sound files on compile?   Matt
• [Asterisk-Users] Register times out on internet outage   Matt
• [Asterisk-Users] Will a VIA Epia ME6000 with a 600MHz Eden fanless CPU be suffiecient for 8 extension system?   Angus Comber
• [Asterisk-Users] chan_cap-cm-0.6 deflect support   Louis-David Mitterrand
• [Asterisk-Users] Will a VIA Epia ME6000 with a 600MHz Eden fanless CPU be suffiecient for 8 extension system?   Dustin Wildes
• [Asterisk-Users] analog phone/door buzzer going through a Sipura SPA2000 ATA dials really slowly   Angus Comber
• [Asterisk-Users] One-way audio with VPN   Alex Lake
• [Asterisk-Users] G.729 patent in France   Steve Underwood
• [Asterisk-Users] VideoConference with UMTS   Matt Riddell
• [Asterisk-Users] Voice Prompts, what do you think? Good voice.   Faris Raouf
• [Asterisk-Users] analog phone/door buzzer going through a Sipura SPA2000 ATA dials really slowly   Matt Riddell
• [Asterisk-Users] OOH323C   Dan Austin
• [Asterisk-Users] Why does the s extension not work in my extensions.conf file   Matt Riddell
• [Asterisk-Users] Best echo canceller?   Tzafrir Cohen
• R: [Asterisk-Users] chan_capi-0.3.5   Giordano Grandis
• [Asterisk-Users] Re:Any way to not overwrite sound files on compile?   Gustavo A. Gonzalez
• [Asterisk-Users] analog phone/door buzzer going through a SipuraSPA2000 ATA dials really slowly   Angus Comber
• [Asterisk-Users] TE410P not working   Simone Cittadini
• [Asterisk-Users] Best echo canceller?   Andrew Kohlsmith
• [Asterisk-Users] Not Authenticate   Fabio Montemaggiore
• [Asterisk-Users] Not authenticate   Fabio Montemaggiore
• [Asterisk-Users] Echo Cancellation not working in Zapata.conf   Alberto Risco
• Mathematicians wanted (was RE: [Asterisk-Users] Best echo canceller?)   Steve Underwood
• [Asterisk-Users] Echo Cancellation not working in Zapata.conf   Alberto Risco
• [Asterisk-Users] OT: SIPSAK usage   Colin Anderson
• [Asterisk-Users] Echo Cancellation not working in Zapata.conf   Andrew Kohlsmith
• [Asterisk-Users] 911 Q   Joel Newkirk
• [Asterisk-Users] chan_zap.so ?   cyril SIMON
• [Asterisk-Users] No ringback tone generated by Asterisk with OH323 connections   Juan Jose Comellas
• [Asterisk-Users] zttest - 100% ?   Kevin Bockman
• [Asterisk-Users] chan_zap.so ?   Andrew Kohlsmith
• [Asterisk-Users] No Incoming Calls on Asterisk   Zeeshan
• [Asterisk-Users] No ringback tone generated by Asterisk with OH323connections   Brian C. Fertig
• [Asterisk-Users] No Incoming Calls on Asterisk   Fabio Montemaggiore
• [Asterisk-Users] Asterisk and telephone volume   Angus Comber
• [Asterisk-Users] 911 Q   Alexander Lopez
• [Asterisk-Users] strange wave like noise on sip handset   Angus Comber
• [Asterisk-Users] maximum retries exceeded on call   Michael Häberle
• [Asterisk-Users] Asterisk and telephone volume   Colin Anderson
• [Asterisk-Users] Question about 3Com(r) 3101 Basic Phone   Jorge Cisneros
• [Asterisk-Users] Any way to not overwrite sound files on compile?   Kevin P. Fleming
• [Asterisk-Users] Re:Any way to not overwrite sound files on compile?   Matt
• [Asterisk-Users] Any way to not overwrite sound files on compile?   Matt
• [Asterisk-Users] Calls Dropping w/ Cisco 7960 Phones   Jon Dahl
• [Asterisk-Users] Correction: Asterisk sound files, audio bandwidth, and sound quality   Stephen Bosch
• [Asterisk-Users] G.729 patent in France   Kevin Walsh
• [Asterisk-Users] Any way to not overwrite sound files on compile?   Andrew Kohlsmith
• [Asterisk-Users] OT: SIPSAK usage   Christian Stredicke
• [Asterisk-Users] Any way to not overwrite sound files on compile?   Kevin P. Fleming
• [Asterisk-Users] It is possible to have 2 AVM Fritz! USB for multiple BRI access?   Amaury BOSSE
• [Asterisk-Users] zttest - 100% ?   DRi at b-w-computer.de
• [Asterisk-Users] Correction: Asterisk sound files, audio bandwidth, and sound quality   steve at daviesfam.org
• [Asterisk-Users] Re:Any way to not overwrite sound files on compile?   Ron Wellsted
• [Asterisk-Users] CRITICAL PROBLEM   Tim McKee
• [Asterisk-Users] Any way to not overwrite sound files on compile?   Kevin P. Fleming
• [Asterisk-Users] mISDN, HFC, W6692, one-way-voice problem   Kovács Attila
• [Asterisk-users]   Fabio Montemaggiore
• [Asterisk-Users] zttest - 100% ?   Carlos Antunes
• [Asterisk-Users] 911 Q   Ray Van Dolson
• [Asterisk-Users] OT: SIPSAK usage   Colin Anderson
• [Asterisk-Users] Siemens TC35 GSM gateway   Bartek Kania
• [Asterisk-Users] CRITICAL PROBLEM   Kevin P. Fleming
• [Asterisk-Users] CRITICAL PROBLEM   Kevin P. Fleming
• [Asterisk-Users] Re: No ringback tone generated by Asterisk with OH323 connections   Tony Mountifield
• [Asterisk-Users] CRITICAL PROBLEM   Kevin Bockman
• [Asterisk-Users] Re: Correction: Asterisk sound files, audio bandwidth, and sound quality   Tony Mountifield
• [Asterisk-Users] Maximum number of Digium Trunk Cards   james.texter at cox.net
• [Asterisk-Users] C Manager Interface Client   Tressler, Joshua A
• [Asterisk-Users] SIP make outside call   David H
• [Asterisk-Users] Will a VIA Epia ME6000 with a 600MHz Eden fanless CPU be suffiecient for 8 extension system?   Tzafrir Cohen
• [Asterisk-Users] chan_zap.so ?   Mojo with Horan & Company, LLC
• [Asterisk-Users] Required hardware   Anders Svensson
• [Asterisk-Users] (no subject)   Jonathan k. Creasy
• [Asterisk-Users] Best way to create IVR/voicemail system   Angus Comber
• [Asterisk-Users] Any way to not overwrite sound files on compile?   James Moore
• [Asterisk-Users] OOH323C   Asterisk guy
• [Asterisk-Users] Any way to not overwrite sound files on compile?   Moore James
• [Asterisk-Users] oh323 implementation 0.67 has call-id problem   Asterisk guy
• [Asterisk-Users] X100p Problem, randomly hungup pstn line   pcman theMan
• [Asterisk-Users] Co-author of O'Reilly's Asterisk book presenting in Utah Valley   Gabriel Gunderson
• [Asterisk-Users] Asterisk useable DID in Newmarket, Ontario   Shawn Porter
• [Asterisk-Users] SIP make outside call   Alex Lake
• [Asterisk-Users] quick question on ztdummy   Jerry Geis
• [Asterisk-Users] quick question on ztdummy   Michiel van Baak
• [Asterisk-Users] Why does the s extension not work inmy extensions.conf file   Angus Comber
• [Asterisk-Users] CRITICAL PROBLEM   Terry Wilson
• [Asterisk-Users] Canada VOIP provider quality   George Pajari
• [Asterisk-Users] Any way to not overwrite sound files on compile?   Terry Wilson
• [Asterisk-Users] Meet me conferencing without blind transfers (Asterisk@home)   jennyw
• [Asterisk-Users] quick question on ztdummy   Jerry Geis
• [Asterisk-Users] Linksys register hangs Asterisk!   Johannes
• [Asterisk-Users] voip alarm circuit   Alex Pavlovic
• [Asterisk-Users] No ringback tone generated by Asterisk with OH323connections   Juan Jose Comellas
• [Asterisk-Users] Polycom IP301 Hangs on boot.   Jonathan k. Creasy
• [Asterisk-Users] quick question on ztdummy   Tzafrir Cohen
• [Asterisk-Users] Revieving some fax problems   Alexandre Leclerc
• [Asterisk-Users] quick question on ztdummy   Tzafrir Cohen
• [Asterisk-Users] 911 Q   Joel Newkirk
• [Asterisk-Users] SPA-841 "Decode Latency"?   alan
• [Asterisk-Users] Revieving some fax problems   Joseph
• [Asterisk-Users] SPA-841 "Decode Latency"?   Luki
• [Asterisk-Users] Fax with asterisk   Rajesh kumar
• [Asterisk-Users] quick question on ztdummy   Michiel van Baak
• [Asterisk-Users] Maximum number of Digium Trunk Cards   Steve Totaro
• [Asterisk-Users] cisco phones problems   Edwin Lam
• [Asterisk-Users] Fritz, mISDN, Help   John Fawcett
• [Asterisk-Users] cisco phones problems   Michiel van Baak
• [Asterisk-Users] CDR and TDS   Andy Kuo
• [Asterisk-Users] Music on hold not initiating RTP stream?   Ray Van Dolson
• [Asterisk-Users] Music on hold not initiating RTP stream?   Kevin P. Fleming
• [Asterisk-Users] transfering calls, no ringing sent to caller   Jeremy Koski
• [Asterisk-Users] Music on hold not initiating RTP stream?   Ray Van Dolson
• [Asterisk-Users] H323 and Asterisk   Martin Vit
• [Asterisk-Users] Re: How can i call to a cellphone here in Mexico?   Claudio Canseco
• [Asterisk-Users] OOH323C   Martin Vit
• [Asterisk-Users] Music on hold not initiating RTP stream?   Ray Van Dolson
• [Asterisk-Users] CDR and TDS   Jim Kou
• [Asterisk-Users] WRT54GP2 SIP server on LAN port   Sherwood McGowan
• [Asterisk-Users] Asterisk and RTP streams   Sherwood McGowan
• [Asterisk-Users] Problems using SIPURA and MFC/R2   acriollo
• [Asterisk-Users] is a dual 1.5Ghz server better than a single 3Ghz for a 100 Iax users asterisk server   Adrien Laurent
• [Asterisk-Users] How to get names into the *411 directory   Doug
• [Asterisk-Users] is a dual 1.5Ghz server better than a single 3Ghz for a 100 Iax users asterisk server   Jason Walker
• [Asterisk-Users] Revieving some fax problems   Jason Walker