March 2007 Archives by author

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Starting: Thu Mar 1 00:06:10 MST 2007
Ending: Sat Mar 31 23:44:06 MST 2007
Messages: 2482

• [asterisk-users] freepbx -> DB Error messages...   Francesco Peeters (Asterisk)
• [asterisk-users] Issue with Hamlet ISDN PCI card(Cologne Chipset)   Francesco Peeters (Asterisk)
• [asterisk-users] error in FreePBX   Francesco Peeters (Asterisk)
• [asterisk-users] qozap: t3 timer expired for span ...   Chris Earle (CBL)
• [asterisk-users] Newbie Question   Leonardo Kamache (Gmail)
• [asterisk-users] server generated outbound conference calls?   Chris Mason (Lists)
• [asterisk-users] A New Phone Service - www.virtualphoneline.com   Chris Mason (Lists)
• [asterisk-users] Polycom 501 - Auto answer on one line appearance   Chris Mason (Lists)
• [asterisk-users] Polycom 501 - Auto answer on one line appearance   Chris Mason (Lists)
• [asterisk-users] asterisk on mini-itx   Chris Mason (Lists)
• [asterisk-users] asterisk on mini-itx   Chris Mason (Lists)
• [asterisk-users] Which VoIP router and switch to use for medium size business   Chris Mason (Lists)
• [asterisk-users] Polycom 501 - Auto answer on one line appearance - Solution   Chris Mason (Lists)
• [asterisk-users] asterisk on mini-itx   Chris Mason (Lists)
• [asterisk-users] polycom random reboots   Chris Mason (Lists)
• [asterisk-users] Re: format_wav.c:247 update_header: Unable to find our position   Chris Mason (Lists)
• [asterisk-users] Which IP Phones have buttons can be assigned to functions with Asterisk   Chris Mason (Lists)
• [asterisk-users] FAX using T38   Matt Riddell (NZ)
• [asterisk-users] Asterisknow with video and X-Lite not quite working   Matt Riddell (NZ)
• [asterisk-users] RE: what happened to asterisk wiki???   Matt Riddell (NZ)
• [asterisk-users] Asterisk queue and agents   Octavio Ruiz (Ta^3)
• [asterisk-users] Call load balancing   Octavio Ruiz (Ta^3)
• [asterisk-users] deprecated ALERT_INFO var andAMI's Originate command   Octavio Ruiz (Ta^3)
• [asterisk-users] Re: [asterisk-dev] Find the name of queue   Octavio Ruiz (Ta^3)
• [asterisk-users] SIP phone supporting more than 10 extension with a call transfer command   Sven Fischer (support)
• [asterisk-users] IAX2, DTMF and x86_64.   William F. Acker WB2FLW +1-303-722-7209
• [asterisk-users] Conference server (or how to make a call with more than 3 users)   Yehavi Bourvine +972-8-9489444
• [asterisk-users] Conference server (or how to make a call with more than 3 u   Yehavi Bourvine +972-8-9489444
• [asterisk-users] Conference server (or how to make a call withmore than 3 u   Yehavi Bourvine +972-8-9489444
• [asterisk-users] Conference server (or how to make a call with more than 3 u   Yehavi Bourvine +972-8-9489444
• [asterisk-users] UK SIP Gateway   --
• [asterisk-users] UK SIP Gateway   --
• [asterisk-users] gtalk2voip and Asteris   DID 4ME
• [asterisk-users] strange things on call transfer   Christoph Fürstaller
• [asterisk-users] strange things on call transfer   Christoph Fürstaller
• [asterisk-users] Autoprovisioning ST2030S   Christoph Fürstaller
• [asterisk-users] Asterisk Feature attended transfer   Christoph Fürstaller
• [asterisk-users] Polycom Questions   »Steven Ringwald«
• [asterisk-users] Polycom Questions   »Steven Ringwald«
• [asterisk-users] Unicall Brazil   Antonio José dos Santos Brandão
• [asterisk-users] error, install freePbx   Carlos Jerónimo
• [asterisk-users] error, install freePbx   Carlos Jerónimo
• [asterisk-users] ses ActiveDirectory and also Ldap and Kerberos.   Carlos Jerónimo
• [asterisk-users] ses ActiveDirectory and also Ldap and Kerberos.   Carlos Jerónimo
• [asterisk-users] "Couldn't load variables.txt?aldope=xxxxx "   Carlos Jerónimo
• [asterisk-users] "Couldn't load variables.txt?aldope=xxxxx "   Carlos Jerónimo
• [asterisk-users] error in FreePBX   Carlos Jerónimo
• [asterisk-users] "Couldn't load variables.txt?aldope=xxxxx "   Carlos Jerónimo
• [asterisk-users] error in FreePBX   Carlos Jerónimo
• [asterisk-users] error in FreePBX   Carlos Jerónimo
• [asterisk-users] "Couldn't load variables.txt?aldope=xxxxx "   Carlos Jerónimo
• [asterisk-users] "Couldn't load variables.txt?aldope=xxxxx "   Carlos Jerónimo
• [asterisk-users] Set(CALLERID(all) not working with 'unknown' call?   Christoph Fürstaller
• [asterisk-users] 120 concurrent ZAP connections in asterisk open edition. Is that possible?   Héctor Maldonado
• [asterisk-users] 120 concurrent ZAP connections in asterisk open edition. Is that possible?   Héctor Maldonado
• [asterisk-users] 120 concurrent ZAP connections in asterisk open edition. Is that possible?   Héctor Maldonado
• [asterisk-users] Trixbox 1.2.3 - TDM400 FXOs - Outgoing Calls - Transfer # Not Wor king   José Luis Ledesma
• [asterisk-users] A200 card problem   Josué Conti
• [asterisk-users] Using server side phonebook directory with SPA941   Christoph Fürstaller
• [asterisk-users] great problem with sounds and ztdummy   Germán Aracil Boned
• [asterisk-users] great problem with sounds and ztdummy   Germán Aracil Boned
• [asterisk-users] great problem with sounds and ztdummy   Germán Aracil Boned
• [asterisk-users] great problem with sounds and ztdummy   Germán Aracil Boned
• [asterisk-users] great problem with sounds and ztdummy   Germán Aracil Boned
• [asterisk-users] While the VoIP-Info.org site is down...   Germán Aracil Boned
• [asterisk-users] Call center manager for Asterisk (Release 0.3)   Germán Aracil Boned
• [asterisk-users] svn 1.4 - mp3 support and changing the installation directory   tzieleniewski
• Re: Re: [asterisk-users] svn 1.4 - mp3 support and changing the installation directory   tzieleniewski
• [asterisk-users] running error: load_modules: No 'modules.conf' found - vesrion 1.4.1 from svn   tzieleniewski
• [asterisk-users] How can I use the "GET VARIABLE variablename" in AGI   李君
• [asterisk-users] Pickup failover   René Enskat
• [asterisk-users] Hinting and Realtime   René Enskat
• AW: [asterisk-users] Hinting and Realtime   René Enskat
• [asterisk-users] cutting hash in dial app   René Enskat
• [asterisk-users] cutting hash in dial app   René Enskat
• [asterisk-users] WMI from Asterisk to Cisco Call Manager   Frédéric Marti
• [asterisk-users] Shoutcast music-on-hold   Jon Schøpzinsky
• [asterisk-users] Conference server (or how to make a call withmore than 3 u   Jon Schøpzinsky
• [asterisk-users] Conference server (or how to make a call withmorethan 3 u   Jon Schøpzinsky
• [asterisk-users] minimal asterisk for iax2 bridge   Francisco Pérez Botella
• [asterisk-users] SMS Integration and SMS commands   Ignacio Ortega A.
• [asterisk-users] Answer Confirmation with SIP/AIX channels   ASTERISKLIST at KEYWAYS.COM
• [Asterisk-Users] Cannot get back chan_zap.so module!??   OCOSA List Acct.
• [asterisk-users] Follow Up on Cannot get back chan_zap.so module!??   OCOSA List Acct.
• [asterisk-users] voip-info.org status update   OCOSA List Acct.
• [asterisk-users] voip-info.org status update   OCOSA List Acct.
• [asterisk-users] TDM400p reliability   Joe Acquisto
• [asterisk-users] MusicOnHold stops after upgrade from 1.4.0 to 1.4.1   Damian Adamski
• [asterisk-users] New to Asterisk   NetSys Admin
• [asterisk-users] New to Asterisk   NetSys Admin
• [asterisk-users] Jajah.com like script?   Ritesh Agrawal
• [asterisk-users] Follow me on multiple numbers..   Ritesh Agrawal
• [asterisk-users] Channel stuck problem..   Ritesh Agrawal
• [asterisk-users] Follow me on multiple numbers..   Ritesh Agrawal
• [asterisk-users] Jajah.com like script?   Ritesh Agrawal
• [asterisk-users] Jajah.com like script?   Ritesh Agrawal
• [asterisk-users] Jajah.com like script?   Ritesh Agrawal
• [asterisk-users] Test Message   Farooq Ahmed
• [asterisk-users] Problem in using Two BRi Cards in Asterisk   Farooq Ahmed
• [asterisk-users] Problem in using Two BRi Cards in Asterisk   Farooq Ahmed
• [asterisk-users] how to check and set D-channel status   Farooq Ahmed
• [asterisk-users] Two or More Bri Cards   Farooq Ahmed
• [asterisk-users] Two or More Bri Cards   Farooq Ahmed
• [asterisk-users] DTMF issue with TDM404   Al
• [asterisk-users] Testing asterisk with sipp   John Albano
• RES: [asterisk-users] Need comparison between PBXtra, Trixbox, Thirdlane, Druid, Aheeva etc.   Alcides
• [asterisk-users] How to log VERBOSE statement to a file?   Larry Alkoff
• [asterisk-users] Re: [A*UG] How to log VERBOSE statement to a file?   Larry Alkoff
• [asterisk-users] switchtype and signalling query   Simon Alman
• [asterisk-users] switchtype and signalling query   Simon Alman
• [asterisk-users] Problem HandyTone 488 does not call transfer   Pablo Almido
• [asterisk-users] How to best manage my dial plans as the continue to grow, and grow, and grow....   Christopher Aloi
• [asterisk-users] 1.4.1 - T38 Pass Through - Seeing some odd errors but the fax works.....   Christopher Aloi
• [asterisk-users] Zaptel 1.4.0   Carlos Alperin
• [asterisk-users] about bluetooth channel   Facundo Ameal
• [asterisk-users] Re: Setting Sip Headers From Dial App?   Benny Amorsen
• [asterisk-users] Re: SIP unicode support ?   Benny Amorsen
• [asterisk-users] IVR after hangup   Benny Amorsen
• [asterisk-users] Re: While the VoIP-Info.org site is down...   Benny Amorsen
• [asterisk-users] Re: While the VoIP-Info.org site is down...   Benny Amorsen
• [asterisk-users] Re: DECT to SIP gateway experiences   Benny Amorsen
• [asterisk-users] Re: Only secretary can call the boss, all others only reach the secretary when dial the boss extension   Benny Amorsen
• [asterisk-users] Re: Fedora + Linux Kernel 2.6 for Zaptel/Asterisk Installation   Benny Amorsen
• [asterisk-users] Re: Gizmo project answers every call - can I use it in hunt group?   Benny Amorsen
• [asterisk-users] Re: Dial(Local/${EXTEN}@longdistance)?   Benny Amorsen
• [asterisk-users] Re: Multi-registration ?   Benny Amorsen
• [asterisk-users] Re: wireless desktop phones   Benny Amorsen
• [asterisk-users] How to handle SIP-Callerid?   Andreas Anderson
• [asterisk-users] Voicechanger update for asterisk 1.4   Andreas Anderson
• [asterisk-users] AOCD -> SendText()?   Andreas Anderson
• [asterisk-users] AOCD -> SendText()?   Andreas Anderson
• [asterisk-users] AOCD -> SendText()?   Andreas Anderson
• [asterisk-users] SER vs Asterisk?   David Anderson
• [asterisk-users] SER vs Asterisk?   David Anderson
• [asterisk-users] what happened to asterisk wiki???   Erik Anderson
• [asterisk-users] Voip-Wiki Site Information   Erik Anderson
• [asterisk-users] Stepped deployment - T1 PRI passthru   Erik Anderson
• [asterisk-users] A New Phone Service - www.virtualphoneline.com   Andres
• [asterisk-users] Issues with a Linksys SPA 2102 and asterisk   Andres
• [asterisk-users] Asterisk 1.2.13 Caller ID problem   Andres
• [asterisk-users] Asterisk 1.2.13 Caller ID problem   Andres
• [asterisk-users] MOS Score   Andres
• [asterisk-users] Microsoft launches first PABX   James Andrewartha
• [asterisk-users] Alec Saunders post about Mashable Telco's   Cory Andrews
• [asterisk-users] What is the best phone to get when using a headset?   Cory Andrews
• [asterisk-users] GXP-2000 Phones with Cisco 3560 PoE Switch   Cory Andrews
• [asterisk-users] Asterisk Automated Outbound Messaging   Cory Andrews
• [asterisk-users] Asterisk Automated Outbound Messaging   Cory Andrews
• [asterisk-users] wireless desktop phones   Cory Andrews
• [asterisk-users] voicemail advanced options problem with mysql datbase   srinivas Antarvedi
• [asterisk-users] Call Back   Samy Antoun
• [asterisk-users] ${EXTEN} is limited to 17 characters under IAX ?   Oded Arbel
• [asterisk-users] Re-parking (or transfer) a parked call   Marc Archer
• [asterisk-users] Cisco 7970 with skinny on * 1.4.1   Julio Arruda
• [asterisk-users] Outbound SIP call from asterisk extension   Karthik Arumugam
• [asterisk-users] outbound call   Karthik Arumugam
• [asterisk-users] Help Voicemail to SMS using asterisk   Goke Aruna
• [asterisk-users] Voicemail to SMS using asterisk   Goke Aruna
• [asterisk-users] Help Voicemail to SMS using asterisk   Goke Aruna
• [asterisk-users] Help Voicemail to SMS using asterisk   Goke Aruna
• [asterisk-users] 1.4 - SLA   Lacy Moore - Aspendora
• [asterisk-users] hanging an asterisk box off of a PBX analog extension   Lacy Moore - Aspendora
• [asterisk-users] 1.4 - SLA   Lacy Moore - Aspendora
• [asterisk-users] When to use Echo Cancellation cards?   Lacy Moore - Aspendora
• [asterisk-users] Which VoIP router and switch to use for medium size business   Lacy Moore - Aspendora
• [asterisk-users] Call center manager for Asterisk (Release 0.3)   Lacy Moore - Aspendora
• [asterisk-users] Polycom call parking feature and Asterisk call parking   Lacy Moore - Aspendora
• [asterisk-users] Teliax problems, they say use SIP, more mature & better working than IAX   Lacy Moore - Aspendora
• [asterisk-users] Cisco 30VIP Phone   Lacy Moore - Aspendora
• [asterisk-users] Cisco 30VIP Phone   Lacy Moore - Aspendora
• [asterisk-users] Re: About Pickup Grandstream   Lacy Moore - Aspendora
• [asterisk-users] Asterisk 1.4.2   Lacy Moore - Aspendora
• [asterisk-users] Re: About Pickup Grandstream   Lacy Moore - Aspendora
• [asterisk-users] Re: About Pickup Grandstream   Lacy Moore - Aspendora
• [asterisk-users] Re: About Pickup Grandstream   Lacy Moore - Aspendora
• [asterisk-users] Re: About Pickup Grandstream   Lacy Moore - Aspendora
• [asterisk-users] Inbound Voice Quality - Speed Change   Lacy Moore - Aspendora
• [asterisk-users] Re: Inbound Voice Quality - Speed Change   Lacy Moore - Aspendora
• [asterisk-users] Re: Inbound Voice Quality - Speed Change   Lacy Moore - Aspendora
• [asterisk-users] ztdummy and MOH   Lacy Moore - Aspendora
• [asterisk-users] ztdummy and MOH   Lacy Moore - Aspendora
• [asterisk-users] ztdummy and MOH   Lacy Moore - Aspendora
• [asterisk-users] ztdummy and MOH   Lacy Moore - Aspendora
• [asterisk-users] Nice Transfer Feature   Lacy Moore - Aspendora
• [asterisk-users] Re: Problem converting a Cisco 7960 to SIP   Lacy Moore - Aspendora
• [asterisk-users] SMS ON ASTERISK   Assis, Eduardo
• [asterisk-users] IAX best practices   Asterisk
• [asterisk-users] IAX best practices   Asterisk
• [asterisk-users] IAX best practices   Asterisk
• [asterisk-users] BRI Cards   Asterisk
• [asterisk-users] Asterisk 1.4.0 Installation error on Red Hat Linux 9.0-Urgent   Asterisk Asterisk
• [asterisk-users] Fwd: Can't hear any sound   Asterisk Asterisk
• [asterisk-users] Can't hear any sound (This time in plain text)   Asterisk Asterisk
• [asterisk-users] Problem configuring voice conference   Asterisk Asterisk
• [asterisk-users] Problem configuring voice conference   Asterisk Asterisk
• [asterisk-users] Call recording and archiving   Voip Asterisk
• [asterisk-users] Fedora + Linux Kernel 2.6 for Zaptel/Asterisk Installation   George C. Attopany
• [asterisk-users] Fedora + Linux Kernel 2.6 for Zaptel/AsteriskInstallation   George C. Attopany
• [asterisk-users] creating new asterisk application   Giedrius Augys
• [asterisk-users] new kernel and zaptel   Giedrius Augys
• [asterisk-users] capi installation problem   Giedrius Augys
• [asterisk-users] txfax and result   Giedrius Augys
• [asterisk-users] dial question   Alvin Austin
• [asterisk-users] Add current caller to junk-callers-database   Alvin Austin
• [asterisk-users] [Announce] Web-MeetMe V3.0.1 released   Dan Austin
• [asterisk-users] Packetization Rate   Dan Austin
• [asterisk-users] ztdummy and MOH   Dan Austin
• [asterisk-users] A New Phone Service - www.virtualphoneline.com   Dovid B
• [asterisk-users] Coaching in asterisk   Dovid B
• [asterisk-users] Newbie Question   Dovid B
• [asterisk-users] voicemail scenario   Dovid B
• [asterisk-users] shutdown   Dovid B
• [asterisk-users] H.323 with g729   Dovid B
• [asterisk-users] accepting a call, macros, and key presses.   Dovid B
• [asterisk-users] HUD Lite server on Debian   Dovid B
• [asterisk-users] extensions.conf & sccp.conf howto call external number   Michiel van Baak
• [asterisk-users] Re: Sending SMS   Michiel van Baak
• [asterisk-users] Re: Sending SMS   Michiel van Baak
• [asterisk-users] So does 1.4.1 show up in the /branches/1.4, or only in the tags/1.4.1   Michiel van Baak
• [asterisk-users] Another Faxing Question   Wes Baehr
• [asterisk-users] How many outgoing phone line/voip account do I need?   Chris Bagnall
• [asterisk-users] SIP hardphones with good jitter tolerance   Chris Bagnall
• [asterisk-users] Linksys/Sipura SPA-942 phones in larger deployments   Chris Bagnall
• [asterisk-users] Emergency chan_sip issue   Chris Bagnall
• [asterisk-users] TDM400p reliability   Chris Bagnall
• [asterisk-users] play file and action only stop if one defined key has been pressed   Time Bandit
• [asterisk-users] Voicemail mailbox number passed in connection?   Time Bandit
• [asterisk-users] Doorphone   Time Bandit
• [asterisk-users] Doorphone   Time Bandit
• [asterisk-users] error in FreePBX   Remco Barendse
• [asterisk-users] Re: Asterisk and DTMF   Carlos Barros
• [asterisk-users] read write or only read fields in cdr?   Bayrouni
• [asterisk-users] read write or only read fields in cdr?   Bayrouni
• [asterisk-users] [asterisk_voip] asterisk and ogg files   Bayrouni
• [asterisk-users] Asterisk 1.4.1   Forrest Beck
• [asterisk-users] Server Recomendation   Forrest Beck
• [asterisk-users] Macro Dial - External DID   Forrest Beck
• [asterisk-users] Paging   Forrest Beck
• [asterisk-users] Re: Paging   Forrest Beck
• [asterisk-users] g.729 on solaris10/x86   Juraj Bednar
• [asterisk-users] Sender phone ringing while recipient talking   Nathan Bell
• [asterisk-users] Sender phone ringing while recipient talking   Nathan Bell
• [asterisk-users] Polycom 601 loop   Nathan Bell
• [asterisk-users] Re: Polycom 601 loop   Nathan Bell
• [asterisk-users] SIP registration   Nathan Bell
• [asterisk-users] Polycom 601 loop   Nathan Bell
• [asterisk-users] Re: Polycom 601 loop   Nathan Bell
• [asterisk-users] Re: Polycom 601 loop   Nathan Bell
• [asterisk-users] SIP registration   Nathan Bell
• [asterisk-users] Re: Polycom 601 loop   Nathan Bell
• [asterisk-users] SIP registration   Nathan Bell
• [asterisk-users] call file vs. originate   Nathan Bell
• [asterisk-users] call file vs. originate   Nathan Bell
• [asterisk-users] Redirect failed, channel not up.   Nathan Bell
• [asterisk-users] Boot order of 2 TE110P and 1 TDM400P in the same machine   Jose Bertuzzi
• [asterisk-users] Boot order of 2 TE110P and 1 TDM400P in the same   Jose Bertuzzi
• [asterisk-users] Re: Asterisknow with video and X-Lite not quiteworking   Biju
• [asterisk-users] Re: Asterisknow with video and X-Lite notquiteworking   Biju
• [asterisk-users] asterisk on mini-itx   Ioan Biris
• [asterisk-users] PRI progress codes.   John Bittner
• [asterisk-users] What is the best phone to get when using a headset?   Gareth Blades
• [asterisk-users] busy/hangup/answer detection in PRI E1 channels   Gareth Blades
• [asterisk-users] Polycom Questions   Steve Blair
• [asterisk-users] Sipura SPA2000 Transfer Call   Chris Blunt
• [asterisk-users] Re: Sending SMS   Al Bochter
• [asterisk-users] Re: Sending SMS   Al Bochter
• [asterisk-users] While the VoIP-Info.org site is down...   Al Bochter
• [asterisk-users] snom led not working with asterisk 1.4.1   Carsten Bock
• [asterisk-users] snom led not working with asterisk 1.4.1   Carsten Bock
• [asterisk-users] Is Allison going to be banned from foreign travel over polar bears?   Leo Ann Boon
• [asterisk-users] Voice mail is not giving unavailable or busy prompts   Stephen Bosch
• [asterisk-users] Polycom reject button   Stephen Bosch
• [asterisk-users] REMOTE CRASH FIX   Stephen Bosch
• [asterisk-users] RE: Coaching in asterisk   Stephen Bosch
• [asterisk-users] A New Phone Service - www.virtualphoneline.com   Stephen Bosch
• [asterisk-users] A New Phone Service - www.virtualphoneline.com   Stephen Bosch
• [asterisk-users] Asterisk 1.2.15 chan_vpb with vpb-driver 4.0   Stephen Bosch
• [asterisk-users] IAX2, DTMF and x86_64.   Stephen Bosch
• [asterisk-users] Polycom call parking feature and Asterisk call parking   Stephen Bosch
• [asterisk-users] RE: Coaching in asterisk   Stephen Bosch
• [asterisk-users] RE: Coaching in asterisk   Stephen Bosch
• [asterisk-users] Polycom 501 - Auto answer on one line appearance   Stephen Bosch
• [asterisk-users] Polycom 501 - Auto answer on one line appearance   Stephen Bosch
• [asterisk-users] Polycom 501 - Auto answer on one line appearance   Stephen Bosch
• [asterisk-users] Polycom call parking feature and Asterisk callparking   Stephen Bosch
• [asterisk-users] Polycom call parking feature and Asterisk callparking   Stephen Bosch
• [asterisk-users] Re: Recorded file processing app wanted   Stephen Bosch
• [asterisk-users] Re: DTMF not being detected with 1 provider. Works with the other provider...   Stephen Bosch
• [asterisk-users] Polycom: warble on registration?   Stephen Bosch
• [asterisk-users] TDM-400, Polycom SIP phones, and echo problems   Stephen Bosch
• [asterisk-users] Voicemails with occasional speeded up portions   Stephen Bosch
• [asterisk-users] Call Back   Stephen Bosch
• [asterisk-users] Call Back   Stephen Bosch
• [asterisk-users] Call Back   Stephen Bosch
• [asterisk-users] What happend to voip-info?   Stephen Bosch
• [asterisk-users] What is the best phone to get when using a headset?   Stephen Bosch
• [asterisk-users] what happened to asterisk wiki???   Stephen Bosch
• [asterisk-users] Polycom call parking feature and Asterisk call parking   Stephen Bosch
• [asterisk-users] While the VoIP-Info.org site is down...   Stephen Bosch
• [asterisk-users] While the VoIP-Info.org site is down...   Stephen Bosch
• [asterisk-users] what happened to asterisk wiki???   Stephen Bosch
• [asterisk-users] TDM-400, Polycom SIP phones, and echo problems   Stephen Bosch
• [asterisk-users] While the VoIP-Info.org site is down...   Stephen Bosch
• [asterisk-users] voip-info.org status update   Stephen Bosch
• [asterisk-users] voip-info.org status update   Stephen Bosch
• [asterisk-users] While the VoIP-Info.org site is down...   Stephen Bosch
• [asterisk-users] voip-info.org is back!   Stephen Bosch
• [asterisk-users] Dell poweredge 860 acceptable for office environment ?   Stephen Bosch
• [asterisk-users] Microsoft launches first PABX   Stephen Bosch
• [asterisk-users] proposal: a new mailing list for asterisk 1.4, why not?   Stephen Bosch
• [asterisk-users] Refund from SellVoip?   Stephen Bosch
• [asterisk-users] High Pitched Noise   Stephen Bosch
• [asterisk-users] A request for your input.   Stephen Bosch
• [asterisk-users] Asterisk 1.4.2 chan_zap   Stephen Bosch
• [asterisk-users] TDM analog cards, volume, echo, fxotune, ztmonitor and HPEC   Stephen Bosch
• [asterisk-users] British Summertime & Grandstream Phones   Stephen Bosch
• [asterisk-users] TDM analog cards, volume, echo, fxotune, ztmonitor and HPEC   Stephen Bosch
• [asterisk-users] Anyone having trouble with claling US Domestic on Sellvoip?   Stephen Bosch
• [asterisk-users] Anyone having trouble with claling US Domestic on Sellvoip?   Stephen Bosch
• [asterisk-users] TDM analog cards, volume, echo, fxotune, ztmonitor and HPEC   Stephen Bosch
• [asterisk-users] Re: Polycom 601 loop   Stephen Bosch
• [asterisk-users] wireless desktop phones   Stephen Bosch
• [asterisk-users] wireless desktop phones   Stephen Bosch
• [asterisk-users] Meetme cant handle more than 5 users in a call?? hmmmm   Stephen Bosch
• [asterisk-users] Park & No Announce?   Stephen Bosch
• [asterisk-users] Dialplan Streaming   Stephen Bosch
• [asterisk-users] Dialplan Streaming   Stephen Bosch
• [asterisk-users] Polycom and Asterisk   Stephen Bosch
• [asterisk-users] Dell poweredge 860 acceptable forofficeenvironment ?   Stephen Bosch
• [asterisk-users] Accessing Voicemail by dialing own number   Dave Bour
• [asterisk-users] proposal: a new mailing list for asterisk 1.4, why not?   Peter Bowyer
• [asterisk-users] Multi-registration ?   Peter Bowyer
• [asterisk-users] Re: Problem converting a Cisco 7960 to SIP   David Boyd
• [asterisk-users] Problem with H323   Sebastian Bozioreanu
• [asterisk-users] Playback 5% Too Fast?   David Brazier
• [asterisk-users] RE: Playback 0.5% Too Fast?   David Brazier
• [asterisk-users] While the VoIP-Info.org site is down...   Shane Breen
• [asterisk-users] While the VoIP-Info.org site is down...   Shane Breen
• [asterisk-users] While the VoIP-Info.org site is down...   Shane Breen
• [asterisk-users] Lucent TNT - ring timer   Brent
• [asterisk-users] what happened to asterisk wiki???   Andy Brezinsky
• [asterisk-users] AsteriskNow Beta 4 with T1 Cards?   Brian K. Alexander, Jr. (Vision Point Systems)
• [asterisk-users] AsteriskNow Beta 4 with T1 Cards?   Brian K. Alexander, Jr. (Vision Point Systems)
• [asterisk-users] AsteriskNow Beta 4 with T1 Cards?   Brian K. Alexander, Jr. (Vision Point Systems)
• [asterisk-users] FW: Seamless Multi Office Asterisk Deployment   Sean Bright
• [asterisk-users] voip-info.org is back!   Sean Bright
• [asterisk-users] asterisk 1.2 branch revision 53132 failed to compile   Sean Bright
• [asterisk-users] asterisk 1.2 branch revision 53132 failed to compile   Sean Bright
• [asterisk-users] Cepstral voices   Sean Bright
• [asterisk-users] How to get AEL2   Sean Bright
• [asterisk-users] How to get AEL2   Sean Bright
• [asterisk-users] How to get AEL2   Sean Bright
• [asterisk-users] Jajah.com like script?   Nicolas Brisac
• [asterisk-users] 1.4 - SLA   Russell Bryant
• [asterisk-users] Double DTMF digits sent on IAX native bridge   Russell Bryant
• [asterisk-users] 1.4 - SLA   Russell Bryant
• [asterisk-users] 1.4 - SLA   Russell Bryant
• [asterisk-users] 1.4 - SLA   Russell Bryant
• [asterisk-users] 1.4 - SLA   Russell Bryant
• [asterisk-users] 1.4 compile issue   Russell Bryant
• [Asterisk-Users] Cannot get back chan_zap.so module!??   Russell Bryant
• [asterisk-users] IAX2, DTMF and x86_64.   Russell Bryant
• [asterisk-users] Asterisk 1.4.2 chan_zap   Russell Bryant
• [asterisk-users] Doorphone   Russell Bryant
• [asterisk-users] Problem while using asterisk Realtime   Russell Bryant
• [asterisk-users] Re: format_wav.c:247 update_header: Unable to find our position   Bartek Bulzak
• [asterisk-users] Re: format_wav.c:247 update_header: Unable to find our position   Bartek Bulzak
• [asterisk-users] Re: Replacement Wiki - options (Formerly 'status of voip-info')   Ira Burton
• [asterisk-users] Asterisk 1.4 and chan_misdn   Pierre Burton
• [asterisk-users] Need feedback on vitelity   Dan Burwinkel
• [asterisk-users] Call Waiting problems   Lachek Butalek
• [asterisk-users] Fwd: Problem with TE212P   Benito Camelas
• [asterisk-users] Problem with TE212P   Benito Camelas
• [asterisk-users] Re: Problem with TE212P   Benito Camelas
• [asterisk-users] Re:Problem with TE212P   Benito Camelas
• [asterisk-users] Asterisk Realtime   Brian Capouch
• [asterisk-users] Re: Asterisk Realtime   Brian Capouch
• [asterisk-users] REALTIME and VIEW ENTIRE DIALPLAN   Brian Capouch
• [asterisk-users] REALTIME and VIEW ENTIRE DIALPLAN   Brian Capouch
• [asterisk-users] asterisk on mini-itx   Brian Capouch
• [asterisk-users] voip-info.org status update   Brian Capouch
• [asterisk-users] A request for your input.   Brian Capouch
• [asterisk-users] Need feedback on vitelity   Brian Capouch
• [asterisk-users] wireless desktop phones   Brian Capouch
• [asterisk-users] Accessing Voicemail by dialing own number   Chris Carey
• [asterisk-users] Accessing Voicemail by dialing own number   Chris Carey
• [asterisk-users] Festival works extension to extension, not on trunk   Chris Carey
• [asterisk-users] FAX using T38   Ricardo Carvalho
• [asterisk-users] Zaptel 1.4.0   Ricardo Carvalho
• [asterisk-users] BLF not working with Asterisk 1.4.0   Ricardo Carvalho
• [asterisk-users] How to match wild card inside a GoToIf?   Ricardo Carvalho
• [asterisk-users] Only secretary can call the boss, all others only reach the secretary when dial the boss extension   Ricardo Carvalho
• [asterisk-users] unconditionally redirecting incoming calls by 302 Moved Temporarily messages doing right accounting   Ricardo Carvalho
• [asterisk-users] rx_fax and Asterisk 1.4.2   Heison Chak
• [asterisk-users] Pickup some else's call   Christopher Chan
• [asterisk-users] Microsoft launches first PABX   Christopher Chan
• [asterisk-users] Microsoft launches first PABX   Christopher Chan
• [asterisk-users] Microsoft launches first PABX   Christopher Chan
• [asterisk-users] Microsoft launches first PABX   Christopher Chan
• [asterisk-users] Microsoft launches first PABX   Christopher Chan
• [asterisk-users] Not registering Port with VSP   Davy Chan
• [asterisk-users] Tesco Internet Phone   Alan Chandler
• [asterisk-users] Voice quality issues   Alan Chandler
• [asterisk-users] Debian Asterisk and MeetMe   Alan Chandler
• [asterisk-users] Problem with ztdummy   Alan Chandler
• [asterisk-users] Problem with ztdummy   Alan Chandler
• [asterisk-users] Problem with ztdummy   Alan Chandler
• [asterisk-users] Problem with ztdummy   Alan Chandler
• [asterisk-users] Transfering not working - how to debug?   Alan Chandler
• [asterisk-users] Linksys SPA 3102 causing me problems   Alan Chandler
• [asterisk-users] Linksys SPA 3102 causing me problems   Alan Chandler
• [asterisk-users] Linksys SPA 3102 causing me problems   Alan Chandler
• [asterisk-users] Linksys SPA 3102 causing me problems   Alan Chandler
• [asterisk-users] [SOLVED]Linksys SPA 3102 causing me problems   Alan Chandler
• [asterisk-users] Understanding the dial flags   Alan Chandler
• [asterisk-users] Pickup group   Khaled Chehab
• [asterisk-users] asterisk 1.2.15 fax   Khaled Chehab
• [asterisk-users] cisco 7905   Khaled Chehab
• [asterisk-users] cisco 7902   Khaled Chehab
• [asterisk-users] ChanSpy with Record() : Doesnt seem to work for an unbridged SIP call   Onglipo Chengoli
• [asterisk-users] Can I use an Intertel IPPhone Plus 7704500 with Asterisk somehow   Bill Chmura
• [asterisk-users] transfer=mediaonly : can't hear nothing   Simone Cittadini
• [asterisk-users] transfer=mediaonly : can't hear nothing   Simone Cittadini
• [asterisk-users] transfer=mediaonly : can't hear nothing   Simone Cittadini
• [asterisk-users] transfer=mediaonly : can't hear nothing   Simone Cittadini
• [asterisk-users] chan_cellphone won't pair with phone   Earle Clubb
• [asterisk-users] Empty Wildcard TDM400P as a MeetMe timer.   Henry Cobb
• [asterisk-users] Newbie Question   Henry Cobb
• [asterisk-users] Which VoIP router and switch to use for medium size business   Henry Cobb
• [asterisk-users] Which VoIP router and switch to use for medium size business   Henry Cobb
• [asterisk-users] Empty Wildcard TDM400P as a MeetMe timer.   Henry Cobb
• [asterisk-users] How many outgoing phone line/voip account do I need?   Henry Cobb
• [asterisk-users] How many outgoing phone line/voip account do I need?   Henry Cobb
• [asterisk-users] voip-info.org status update   Henry Cobb
• [asterisk-users] Newbie Question   Henry Cobb
• [asterisk-users] Which parameters of a live Asterisk server would you monitor ?   Henry Cobb
• [asterisk-users] polycom random reboots   Henry Cobb
• [asterisk-users] IAX best practices   Henry J. Cobb
• [asterisk-users] what happened to asterisk wiki???   James Coberly
• [asterisk-users] voicemail scenario   richard Coco
• [asterisk-users] voicemail scenario   richard Coco
• [asterisk-users] voicemail scenario   richard Coco
• [asterisk-users] install and setup app_mp4 application   richard Coco
• [asterisk-users] TDM400p Loaded only once   Tzafrir Cohen
• [asterisk-users] Asterisk and Fax   Tzafrir Cohen
• [asterisk-users] DTMF from TDM400P and X100P   Tzafrir Cohen
• [asterisk-users] IAX best practices   Tzafrir Cohen
• [asterisk-users] DTMF from TDM400P and X100P   Tzafrir Cohen
• [asterisk-users] Help: CallerID Name not being sent on outboundPRI trunk   Tzafrir Cohen
• [asterisk-users] svn 1.4 - mp3 support and changing the installation directory   Tzafrir Cohen
• [asterisk-users] Voice quality issues   Tzafrir Cohen
• [asterisk-users] new kernel and zaptel   Tzafrir Cohen
• [asterisk-users] Re: Problem with TE212P   Tzafrir Cohen
• [asterisk-users] zaptel 1.4 on fedora core 6 with dell pe 2850   Tzafrir Cohen
• [asterisk-users] Manager.conf '127.0.0.1 unable to authenticate'   Tzafrir Cohen
• [asterisk-users] Problems with Zaptel Drivers   Tzafrir Cohen
• [asterisk-users] Zaptel problem after upgrading to 1.2.16   Tzafrir Cohen
• [asterisk-users] RE: Zaptel problem after upgrading to 1.2.16   Tzafrir Cohen
• [asterisk-users] Boot order of 2 TE110P and 1 TDM400P in the same machine   Tzafrir Cohen
• [asterisk-users] Boot order of 2 TE110P and 1 TDM400P in the same machine   Tzafrir Cohen
• [asterisk-users] AEL #include file   Tzafrir Cohen
• [asterisk-users] installation pb on debian etch   Tzafrir Cohen
• [asterisk-users] Empty Wildcard TDM400P as a MeetMe timer.   Tzafrir Cohen
• [asterisk-users] Is there any variable for Voicemail Password in Asterisk   Tzafrir Cohen
• [asterisk-users] How to access Voicemail Password in Asteriskwithout using V   Tzafrir Cohen
• [asterisk-users] Follow Up on Cannot get back chan_zap.so module!??   Tzafrir Cohen
• [asterisk-users] Re: Recorded file processing app wanted   Tzafrir Cohen
• [asterisk-users] Fix for TZ values updates for DST   Tzafrir Cohen
• [asterisk-users] 120 concurrent ZAP connections in asterisk open edition. Is that possible?   Tzafrir Cohen
• [asterisk-users] DST and VM timestamp   Tzafrir Cohen
• [asterisk-users] great problem with sounds and ztdummy   Tzafrir Cohen
• [asterisk-users] Earliest dial tone, after boot up.   Tzafrir Cohen
• [asterisk-users] qozap: t3 timer expired for span ...   Tzafrir Cohen
• [asterisk-users] Re: zapata with Tiger3XX compilation error   Tzafrir Cohen
• [asterisk-users] asterisk 1.2 branch revision 53132 failed to compile   Tzafrir Cohen
• [asterisk-users] asterisk 1.2 branch revision 53132 failed to compile   Tzafrir Cohen
• [asterisk-users] asterisk 1.2 branch revision 53132 failed to compile   Tzafrir Cohen
• [asterisk-users] asterisk 1.2 branch revision 53132 failed to compile   Tzafrir Cohen
• [asterisk-users] asterisk 1.2 branch revision 53132 failed to compile   Tzafrir Cohen
• [asterisk-users] Zaptel silly issue   Tzafrir Cohen
• [asterisk-users] error, install freePbx   Tzafrir Cohen
• [asterisk-users] Re: About Pickup Grandstream   Tzafrir Cohen
• [asterisk-users] AsteriskNow Beta 4 with T1 Cards?   Tzafrir Cohen
• [asterisk-users] GTalk/Jabber passing audio in 1.4.1!   Tzafrir Cohen
• [asterisk-users] Asterisk Viruses?   Tzafrir Cohen
• [asterisk-users] Issue with Hamlet ISDN PCI card(Cologne Chipset)   Tzafrir Cohen
• [asterisk-users] Problem with ztdummy   Tzafrir Cohen
• [asterisk-users] TDM analog cards, volume, echo, fxotune, ztmonitor and HPEC   Tzafrir Cohen
• [asterisk-users] Problem with ztdummy   Tzafrir Cohen
• [asterisk-users] Problem with ztdummy   Tzafrir Cohen
• [asterisk-users] ztdummy install in the new zaptel 1.4.1   Tzafrir Cohen
• [asterisk-users] voicemail is not playing messages   Tzafrir Cohen
• [asterisk-users] Problem with ztdummy   Tzafrir Cohen
• [asterisk-users] AsteriskNow Beta 4 with T1 Cards?   Tzafrir Cohen
• [asterisk-users] Re: Two or More Bri Cards   Tzafrir Cohen
• [asterisk-users] vzaphfc installation...   Tzafrir Cohen
• [asterisk-users] Asterisk: recommended installation   Tzafrir Cohen
• [asterisk-users] Re: Inbound Voice Quality - Speed Change   Tzafrir Cohen
• [asterisk-users] Voice mail   Tzafrir Cohen
• [asterisk-users] Re: Re: Inbound Voice Quality - Speed Change   Tzafrir Cohen
• [asterisk-users] switchtype and signalling query   Tzafrir Cohen
• [asterisk-users] LUSYN patches   Tzafrir Cohen
• [asterisk-users] Tesco Internet Phone   Dean Collins
• [asterisk-users] Alec Saunders post about Mashable Telco's   Dean Collins
• [asterisk-users] Alec Saunders post about Mashable Telco's   Dean Collins
• FW: [asterisk-users] Alec Saunders post about Mashable Telco's   Dean Collins
• [asterisk-users] Alec Saunders post about Mashable Telco's   Dean Collins
• [asterisk-users] Help Voicemail to SMS using asterisk   Dean Collins
• [asterisk-users] Alec Saunders post about Mashable Telco's   Dean Collins
• [asterisk-users] server generated outbound conference calls?   Dean Collins
• [asterisk-users] Long term voicemail archival and synchronisation between multiple storage locations?   Dean Collins
• [asterisk-users] Cancelling a digit in IVR   Dean Collins
• [asterisk-users] asterisk and ssl   Dean Collins
• [asterisk-users] RE: Coaching in asterisk   Dean Collins
• [asterisk-users] Dell poweredge 860 acceptable for officeenvironment ?   Dean Collins
• [asterisk-users] FW: Microsoft buys Tellme   Dean Collins
• [asterisk-users] FW: Microsoft buys Tellme   Dean Collins
• [asterisk-users] Microsoft launches first PABX   Dean Collins
• [asterisk-users] Microsoft launches first PABX   Dean Collins
• [asterisk-users] Dlink i2eye   Dean Collins
• [asterisk-users] Linksys/Sipura SPA-942 phones in largerdeployments   Dean Collins
• [asterisk-users] PC / Phone Combo   Dean Collins
• [asterisk-users] RE: PC / Phone Combo   Dean Collins
• [asterisk-users] Doorphone vs. Grandstream BT101   Dean Collins
• [asterisk-users] Semi-OT: Use T.38 ATAs to Extend fax lines   Dean Collins
• [asterisk-users] Shoretel integration into Salesforce.com   Dean Collins
• [asterisk-users] wireless desktop phones   Dean Collins
• [asterisk-users] Meetme cant handle more than 5 users in a call?? hmmmm   Dean Collins
• [asterisk-users] wireless desktop phones   Dean Collins
• [asterisk-users] wireless desktop phones   Dean Collins
• [asterisk-users] Setting a call to be recorded before Xfer?   Dean Collins
• [asterisk-users] Rx+,Rx-,Tx+,Tx- of TE110P   Michael Collins
• [asterisk-users] Asterisk Auto-dial out   Michael Collins
• [asterisk-users] Replacement Wiki - options (Formerly 'status of voip-info')   Michael Collins
• [asterisk-users] Problem with "AT&T Maintenance" protocol inPRI connection, no B+D channels available   Michael Collins
• [asterisk-users] Asterisk Automated Outbound Messaging   Michael Collins
• [asterisk-users] Problem with "AT&T Maintenance" protocol in PRIconnection, no B+D channels available   Michael Collins
• [asterisk-users] Sendmail and exchange for voicemail integration   Michael Collins
• [asterisk-users] So does 1.4.1 show up in the /branches/1.4, or only in the tags/1.4.1   Joshua Colp
• [asterisk-users] Packet2Packet Bridging Questions   Joshua Colp
• [asterisk-users] FW: Seamless Multi Office Asterisk Deployment   Brandon Comouche
• [asterisk-users] FW: Seamless Multi Office Asterisk Deployment   Brandon Comouche
• [asterisk-users] Background / Invalid Extension through cell phone   John Congdon
• [asterisk-users] cisco sip firmware update for cisco 7970   Connolly, Tim
• [asterisk-users] How to match wild card inside a GoToIf?   Connolly, Tim
• [asterisk-users] Getting 7970 to update   Connolly, Tim
• [asterisk-users] Getting 7970 to update   Connolly, Tim
• [asterisk-users] Dropped calls in Asterisk - A general question   Connolly, Tim
• [asterisk-users] Re: asterisk-users Digest, Vol 32, Issue 48   David Cook
• [asterisk-users] Re: how to define a pilot number   David Cook
• [asterisk-users] Re: asterisk-users Digest, Vol 32, Issue 106   David Cook
• [asterisk-users] Hang up detection time in FXS module   Gustavo Cordeiro
• [asterisk-users] AOC billing   Stefano Corsi
• [asterisk-users] Re: AOC billing   Stefano Corsi
• [asterisk-users] Security on long distance calls   Stefano Corsi
• [asterisk-users] about bluetooth channel   Dave Cotton
• [asterisk-users] How many outgoing phone line/voip account do I need?   Dave Cotton
• [asterisk-users] How many outgoing phone line/voip account do I need?   Dave Cotton
• [asterisk-users] Rebooting all Aastra phones   Dave Cotton
• [asterisk-users] Rebooting all Aastra phones   Dave Cotton
• [asterisk-users] Re: While the VoIP-Info.org site is down...   Dave Cotton
• [asterisk-users] asterisk on debian   Bobby Crawford
• [asterisk-users] hardware spec   Bobby Crawford
• [asterisk-users] Single sign on PC + phone?   Jonathan k. Creasy
• [asterisk-users] What happend to voip-info?   Jonathan k. Creasy
• [asterisk-users] E-911 and Asterisk   Matthew Crocker
• [asterisk-users] Linksys PAP2 and Caller ID   Gergo Csibra
• [asterisk-users] disable client side hangup after dialing 911   Gergo Csibra
• [asterisk-users] asterisk on debian   Gergo Csibra
• [asterisk-users] FAX & mISDN   Gergo Csibra
• [asterisk-users] Linksys SPA 3102 causing me problems   Gergo Csibra
• [asterisk-users] Re: [asterisk-biz] Need feedback on vitelity   Alistair Cunningham
• [asterisk-users] hanging an asterisk box off of a PBX analog extension   Mike D'Ambrogia
• [asterisk-users] Polycom: warble on registration?   Ken D'Ambrosio
• [asterisk-users] SIP unicode support ?   Klaus Darilion
• [asterisk-users] Re: SIP unicode support ?   Klaus Darilion
• [asterisk-users] Re: SIP unicode support ?   Klaus Darilion
• [asterisk-users] Number of SIP messages per minute   Mark Davies
• [asterisk-users] Zaptel problem after upgrading to 1.2.16   Mark Davies
• [asterisk-users] RE: Zaptel problem after upgrading to 1.2.16   Mark Davies
• [asterisk-users] Zaptel problem after upgrading to 1.2.16   Mark Davies
• [asterisk-users] Zaptel problem after upgrading to 1.2.16   Mark Davies
• [asterisk-users] Asterisk 1.4.1 - Calling problem   Thomas Deillon
• [asterisk-users] Asterisk 1.4.2   Thomas Deillon
• [asterisk-users] Setting rxgain per channel   Delca
• [asterisk-users] Setting rxgain per channel   Delca
• [asterisk-users] Siemens HiPATH 3700 with Asterisk   Jorge de Diego
• [asterisk-users] Call dies when I press *   Mike Diehl
• [asterisk-users] Call dies when I press *   Mike Diehl
• [asterisk-users] strange ring   Pryakhin Dimitry
• [asterisk-users] Re: strange ring   Pryakhin Dimitry
• [asterisk-users] Re: strange ring   Pryakhin Dimitry
• [asterisk-users] Asterisk with AudioCodes Mediant 2000   Kinjal Dixit
• [asterisk-users] Off Topic: Open Source USB Softphone   Michael Van Donselaar
• [asterisk-users] Re: asterisk-users Digest, Vol 32, Issue 67   Yeruult Dorjsuren
• [asterisk-users] spandsp, app_rxfax: apps_Makefile.patch v1.2 > v1.4 = No Workie!   Doug
• [asterisk-users] Call dies when I press *   Doug
• [asterisk-users] TDM-400, Polycom SIP phones, and echo problems   Peter Doyle
• [asterisk-users] Asterisk 1.4.2   Gregory Duchatelet
• [asterisk-users] Registration timed out after a "sip reload"   Gregory Duchatelet
• [asterisk-users] Inbound Voice Quality - Speed Change   Jim Duda
• [asterisk-users] Re: Inbound Voice Quality - Speed Change   Jim Duda
• [asterisk-users] Re: Inbound Voice Quality - Speed Change   Jim Duda
• [asterisk-users] Re: Inbound Voice Quality - Speed Change   Jim Duda
• [asterisk-users] Re: Re: Inbound Voice Quality - Speed Change   Jim Duda
• [asterisk-users] Re: Inbound Voice Quality - Speed Change   Jim Duda
• [asterisk-users] Re: Re: Inbound Voice Quality - Speed Change   Jim Duda
• [asterisk-users] DST changes for the US   Darryl Dunkin
• [asterisk-users] Zaptel Dummy Driver   Darryl Dunkin
• [asterisk-users] Zaptel Dummy Driver   Darryl Dunkin
• [asterisk-users] Polycom and Asterisk   Darryl Dunkin
• [asterisk-users] Polycom SoundPoint 501   Darryl Dunkin
• [asterisk-users] Asterisk and Fax   Michelle Dupuis
• [asterisk-users] IAX best practices   Michelle Dupuis
• [asterisk-users] DTMF detection problems on PRI channels?   Michelle Dupuis
• [asterisk-users] Asterisk 1.4.1 Released   Michelle Dupuis
• [asterisk-users] How many gsm channels   Michelle Dupuis
• [asterisk-users] Fedora + Linux Kernel 2.6 forZaptel/AsteriskInstallation   Michelle Dupuis
• [asterisk-users] Emergency chan_sip issue   Michelle Dupuis
• [asterisk-users] just call to user   Michelle Dupuis
• [asterisk-users] App_RXFax Problem.   Michelle Dupuis
• [asterisk-users] Re: qozap: t3 timer expired for span ...   Chris Earle
• [asterisk-users] Re: qozap: t3 timer expired for span ...   Chris Earle
• [asterisk-users] voip-info.org status update   Gary Eck
• [asterisk-users] Voicemail for an AT&T System 75   Gary Eck
• [asterisk-users] app_queue not using exit context?   Steve Edwards
• [asterisk-users] Call load balancing   Steve Edwards
• [asterisk-users] Call load balancing   Steve Edwards
• [asterisk-users] Recorded file processing app wanted   Steve Edwards
• [asterisk-users] Re: Recorded file processing app wanted   Steve Edwards
• [asterisk-users] Recorded file processing app wanted   Steve Edwards
• [asterisk-users] Complicated callback solution   Steve Edwards
• [asterisk-users] ACM question   Steve Edwards
• [asterisk-users] Jajah.com like script?   Steve Edwards
• [asterisk-users] Jajah.com like script?   Steve Edwards
• [asterisk-users] Queues   Steve Edwards
• [asterisk-users] Metaswitch help needed   Steve Edwards
• [asterisk-users] Timeout for conferences   Steve Edwards
• [asterisk-users] SIP OPTIONS dialog not understood   Steve Edwards
• [asterisk-users] DST and VM timestamp   Damon Estep
• [asterisk-users] DST and VM timestamp   Damon Estep
• [asterisk-users] DTMF not being detected with 1 provider. Works with the other provider...   Evert
• [asterisk-users] Re: DTMF not being detected with 1 provider. Works with the other provider...   Evert
• [asterisk-users] Help: CallerID Name not being sent on outbound PRI trunk   C F
• [asterisk-users] Help Needed: Can't make "local" calls on a brand new PRI   C F
• [asterisk-users] No Caller ID Name PRI NI2   C F
• [asterisk-users] No Caller ID Name PRI NI2   C F
• [asterisk-users] Test   C F
• [asterisk-users] dial question   C F
• [asterisk-users] Help: CallerID Name not being sent on outbound PRI trunk   C F
• [asterisk-users] Problem with TE212P   C F
• [asterisk-users] Read() status?   C F
• [asterisk-users] How to disable MOH completely?   C F
• [asterisk-users] How to disable MOH completely?   C F
• [asterisk-users] Re: Help: CallerID Name not being sent on outbound PRI trunk   C F
• [asterisk-users] How to best manage my dial plans as the continue to grow, and grow, and grow....   C F
• [asterisk-users] Re: Help: CallerID Name not being sent on outbound PRI trunk   C F
• [asterisk-users] Re: Help: CallerID Name not being sent on outbound PRI trunk   C F
• [asterisk-users] DST changes for the US   C F
• [asterisk-users] Re: Help: CallerID Name not being sent on outbound PRI trunk   C F
• [asterisk-users] Microsoft launches first PABX   C F
• [asterisk-users] Microsoft launches first PABX   C F
• [asterisk-users] Microsoft launches first PABX   C F
• [asterisk-users] wrong values in duration and billsec in CDR   C F
• [asterisk-users] Re: wrong values in duration and billsec in CDR   C F
• [asterisk-users] Re: wrong values in duration and billsec in CDR   C F
• [asterisk-users] Sendmail and exchange for voicemail integration   C F
• [asterisk-users] How is this feature called ?   C F
• [asterisk-users] Stepped deployment - T1 PRI passthru   C F
• [asterisk-users] modem passthru   Mark Farver
• [asterisk-users] modem passthru   Mark Farver
• [asterisk-users] About queues and multiple lines.   Nuria Fernandez
• [asterisk-users] CTI   Nuria Fernandez
• [asterisk-users] OS X Frequent console disconnects 1.4.1   Bruce Ferrell
• [asterisk-users] cause 127   Bruce Ferrell
• [asterisk-users] Erased log files   Bruce Ferrell
• [asterisk-users] IAX2 - Congestion   Mario Mayerle Filho
• [asterisk-users] Noob question regarding PCI 2.x & TDM400P Card   Barton Fisher
• [asterisk-users] TDM400P/FXS in a HP DL380 G5   James FitzGibbon
• [asterisk-users] Re: TDM400P/FXS in a HP DL380 G5   James FitzGibbon
• [asterisk-users] Problem with busy and unavailable   James FitzGibbon
• [asterisk-users] REMOTE CRASH FIX   Kevin P. Fleming
• [asterisk-users] Asterisk 1.4.1   Kevin P. Fleming
• [asterisk-users] FAX using T38   Kevin P. Fleming
• [asterisk-users] svn 1.4 - mp3 support and changing the installation directory   Kevin P. Fleming
• [asterisk-users] So does 1.4.1 show up in the /branches/1.4, or only in the tags/1.4.1   Kevin P. Fleming
• [asterisk-users] So does 1.4.1 show up in the /branches/1.4, or only in the tags/1.4.1   Kevin P. Fleming
• [asterisk-users] running error, unable to load *.conf files: load_modules: No 'modules.conf' found - svn version 1.4.1   Kevin P. Fleming
• [asterisk-users] FAX using T38   Kevin P. Fleming
• [asterisk-users] Is the 1.0.X branch vulnerable to the SIP issue?   Kevin P. Fleming
• [asterisk-users] Re: TDM400P/FXS in a HP DL380 G5   Kevin P. Fleming
• [asterisk-users] g.729 on solaris10/x86   Kevin P. Fleming
• [asterisk-users] extra-sounds 1.4.5 timestapmed newer than 1.4.6 ???   Kevin P. Fleming
• [asterisk-users] Zaptel version for asterisk 1.2.16   Kevin P. Fleming
• [asterisk-users] [G.729] Input Gain   Kevin P. Fleming
• [asterisk-users] transfer=mediaonly : can't hear nothing   Kevin P. Fleming
• [asterisk-users] Error compiling zaptel 1.4.0   Kevin P. Fleming
• [asterisk-users] transfer=mediaonly : can't hear nothing   Kevin P. Fleming
• [asterisk-users] transfer=mediaonly : can't hear nothing   Kevin P. Fleming
• [asterisk-users] codec_zap and Asterisk 1.4.1   Kevin P. Fleming
• [asterisk-users] 302 Moved temporarely   Kevin P. Fleming
• [asterisk-users] Re: 302 Moved temporarely   Kevin P. Fleming
• [asterisk-users] About FAX and PAP2   Kevin P. Fleming
• [asterisk-users] Voicemail for an AT&T System 75   Kevin P. Fleming
• [asterisk-users] TDM400p reliability   Kevin P. Fleming
• [asterisk-users] DTMF Corruption Problem in 1.4.2 for SIP RFC2833 plz halp   Kevin P. Fleming
• [asterisk-users] Multi-Level Queue   Kevin P. Fleming
• [asterisk-users] Multi-Level Queue   Kevin P. Fleming
• [asterisk-users] 120 concurrent ZAP connections in asterisk open edition. Is that possible?   Matt Florell
• [asterisk-users] 120 concurrent ZAP connections in asterisk open edition. Is that possible?   Matt Florell
• [asterisk-users] disable client side hangup after dialing 911   Patrick Fortin
• [asterisk-users] Asterisknow with video and X-Lite not quite working   Benedikt Franz
• [asterisk-users] Re: Asterisknow with video and X-Lite not quite working   Benedikt Franz
• [asterisk-users] Re: Asterisknow with video and X-Lite not quiteworking   Benedikt Franz
• [asterisk-users] Asterisknow with video and X-Lite not quite working   Benedikt Franz
• [asterisk-users] TDM-400, Polycom SIP phones, and echo problems   Matthew Fredrickson
• [asterisk-users] Help! Echo problem even at T1 PRI?   Matthew Fredrickson
• [asterisk-users] Problem with "AT&T Maintenance" protocol in PRI connection, no B+D channels available   Matthew Fredrickson
• [asterisk-users] TDM analog cards, volume, echo, fxotune, ztmonitor and HPEC   Matthew Fredrickson
• [asterisk-users] Re: Question about DSP in Digium card   Matthew Fredrickson
• [asterisk-users] Meetme cant handle more than 5 users in a call?? hmmmm   Matthew Fredrickson
• [asterisk-users] Using Asterisk as Voicemail Server on a dinosaur Meridian System   J French
• [asterisk-users] Setting a call to be recorded before Xfer?   J French
• [asterisk-users] Polycom Questions   Dave Fullerton
• [asterisk-users] DST changes for the US   Dave Fullerton
• [asterisk-users] Rebooting ALL polycom phones   Dave Fullerton
• [asterisk-users] OT: Sipura DST Rules   Dave Fullerton
• [asterisk-users] DST and VM timestamp   Dave Fullerton
• [asterisk-users] Semi-OT: Use T.38 ATAs to Extend fax lines   Dave Fullerton
• [asterisk-users] Doorphone vs. Grandstream BT101   Dave Fullerton
• [asterisk-users] TDM400p reliability   Dave Fullerton
• [asterisk-users] TDM400p reliability   Dave Fullerton
• [asterisk-users] PoE - IEEE 802.3af   Dave Fullerton
• [asterisk-users] Ringing does not terminate on mISDN after pickup   Arik Raffael Funke
• [asterisk-users] No application 'Prefix' for extension in1.2x, what app I have to use instead?   Rafael J. Risco G.V.
• [asterisk-users] ChanSpy and MeetMe   GDrayer at guesswho.com
• [asterisk-users] ChanSpy and MeetMe   GDrayer at guesswho.com
• [asterisk-users] About Pickup Grandstream   LKS GMAIL
• [asterisk-users] PickUp a call with feature pickup (*8) from a IAX2channel   LKS GMAIL
• [asterisk-users] Re: About Pickup Grandstream   LKS GMAIL
• [asterisk-users] Re: About Pickup Grandstream   LKS GMAIL
• [asterisk-users] FAX & mISDN   LKS GMAIL
• [asterisk-users] help - UNSUBSCRIBE   LKS GMAIL
• [asterisk-users] FAX & mISDN   LKS GMAIL
• [asterisk-users] FAX & mISDN   LKS GMAIL
• [asterisk-users] FWD outgoing problem   Bogdan GONCIULEA
• [asterisk-users] PC / Phone Combo   Chris Gamble
• [asterisk-users] RE: PC / Phone Combo   Chris Gamble
• [asterisk-users] UK SIP Gateway   George Gardiner
• [asterisk-users] Asterisk Java and Astmanproxy   Doug Garstang
• [asterisk-users] Asterisk Java w/ Threads   Doug Garstang
• [asterisk-users] Asterisk Java w/ Threads   Doug Garstang
• [asterisk-users] Asterisk Java w/ Threads   Doug Garstang
• [asterisk-users] Read() status?   Doug Garstang
• [asterisk-users] RE: Coaching in asterisk   Doug Garstang
• [asterisk-users] Dialplan Streaming   Doug Garstang
• [asterisk-users] Dialplan Streaming   Doug Garstang
• [asterisk-users] Dialplan Streaming   Doug Garstang
• [asterisk-users] Dialplan Streaming   Doug Garstang
• [asterisk-users] 1.4 - SLA   Bill Gibbs
• [asterisk-users] 1.4 - SLA   Bill Gibbs
• [asterisk-users] 1.4 - SLA   Bill Gibbs
• [asterisk-users] 1.4 - SLA   Bill Gibbs
• [asterisk-users] 1.4 - SLA   Bill Gibbs
• [asterisk-users] sip show channels   Bill Gibbs
• [asterisk-users] Sender phone ringing while recipient talking   Bill Gibbs
• [asterisk-users] Polycom call parking feature and Asterisk callparking   Bill Gibbs
• [asterisk-users] Polycom call parking feature and Asteriskcallparking   Bill Gibbs
• [asterisk-users] PROGRESS code   Bill Gibbs
• [asterisk-users] forwarding loop not detected   Bill Gibbs
• [asterisk-users] Problem HandyTone 488 does not call transfer   Drew Gibson
• [asterisk-users] Queue announcing hold sequence instead of hold time   Drew Gibson
• [asterisk-users] Is Allison going to be banned from foreign travel over polar bears?   Drew Gibson
• [asterisk-users] Zaptel problem after upgrading to 1.2.16   Drew Gibson
• [asterisk-users] Queue announcing hold sequence instead of hold time   Drew Gibson
• [asterisk-users] While the VoIP-Info.org site is down...   Drew Gibson
• [asterisk-users] voip-info.org status update   Drew Gibson
• Fwd: [asterisk-users] Multi-registration ?   Drew Gibson
• Fwd: [asterisk-users] Multi-registration ?   Drew Gibson
• [asterisk-users] wireless desktop phones   Drew Gibson
• [asterisk-users] Multi-line phones - Asterisk uses wrong callerid   Drew Gibson
• [asterisk-users] Multi-line phones - Asterisk uses wrong callerid   Drew Gibson
• [asterisk-users] Multi-line phones - Asterisk uses wrong callerid   Drew Gibson
• Fwd: [asterisk-users] Multi-registration ?   Drew Gibson
• [asterisk-users] DST 2007 Config for Cisco 7970   Gary T. Giesen
• [asterisk-users] Anyone having trouble with claling US Domestic on Sellvoip?   Salvatore Giudice
• [asterisk-users] Anyone having trouble with claling US Domesticon Sellvoip?   Salvatore Giudice
• [asterisk-users] Re: Question about DSP in Digium card   Salvatore Giudice
• [asterisk-users] Multi-registration ?   Salvatore Giudice
• [asterisk-users] Multi-registration ?   Salvatore Giudice
• [asterisk-users] Re: Problem converting a Cisco 7960 to SIP   Salvatore Giudice
• [asterisk-users] SIP RTP Tunnel   Salvatore Giudice
• [asterisk-users] One way intermittent static to PSTN   Salvatore Giudice
• [asterisk-users] xten web phone   Salvatore Giudice
• [asterisk-users] Re: wireless desktop phones   Salvatore Giudice
• [asterisk-users] Sponsored development - Monodirectional audio handling   Salvatore Giudice
• [asterisk-users] Polycom Power   David Gomillion
• [asterisk-users] FWD outgoing problem   Bogdan Gonciulea
• [asterisk-users] transfers and CDR   Rodrigo Gonzalez
• [asterisk-users] wireless desktop phones   Matt Gorecki
• [asterisk-users] SRTP vs ZRTP in Asterisk   Michael Graves
• [asterisk-users] voip-info.org status update   Joe Greco
• [asterisk-users] voip-info.org status update   Joe Greco
• [asterisk-users] SIP RTP Tunnel   Ed Greenberg
• [asterisk-users] Asterisk + Panasonic pbx   Sanspareils Greenlans
• [asterisk-users] Connecting two asterisk server.   Sanspareils Greenlans
• [asterisk-users] Newbie extensions.conf question   Chris Griffin
• [asterisk-users] Newbie extensions.conf question   Chris Griffin
• [asterisk-users] transfer function   Denis V. Gudtsov
• [asterisk-users] Problem with busy and unavailable   Stefan Guenther
• [asterisk-users] Problem with busy and unavailable   Stefan Guenther
• [asterisk-users] YAACID and manager.conf security   Todd H
• [asterisk-users] GXP-2000 DST Change   Todd H
• [asterisk-users] A200 card problem   Todd H
• [asterisk-users] A200 card problem   Todd H
• [asterisk-users] managers   Todd H
• [asterisk-users] wireless desktop phones   Todd H
• [asterisk-users] Linksys SPA 3102 causing me problems   Todd H
• [asterisk-users] SPA3102 PSTN fallback   Todd H
• [asterisk-users] voip-info.org status update   Bill Hackensack
• [asterisk-users] Cisco 30VIP Phone   Bill Hackensack
• [asterisk-users] Cisco 7970 with skinny on * 1.4.1   Bill Hackensack
• [asterisk-users] Asterisk 1.4.2 chan_zap   Bill Hackensack
• [asterisk-users] A request for your input.   Bill Hackensack
• [asterisk-users] CN=Diarmaid O'Loughlin/O=QAD1 is out of the office.   Bill Hackensack
• [asterisk-users] Problem configuring voice conference   Paul Hales
• [asterisk-users] How many outgoing phone line/voip account do I need?   Paul Hales
• [asterisk-users] Re: DTMF not being detected with 1 provider. Works with the other provider...   Paul Hales
• [asterisk-users] Problem in using Two BRi Cards in Asterisk   Paul Hales
• [asterisk-users] Server Recomendation   Paul Hales
• [asterisk-users] cmd page crashes Asterisk SVN-branch-1.4-r57207   Hall, Eric M.
• Spam? Re: [asterisk-users] cmd page crashes Asterisk SVN-branch-1.4-r57207   Hall, Eric M.
• [asterisk-users] dial question   Hall, Eric M.
• [asterisk-users] Voicemail question   Hall, Eric M.
• [asterisk-users] Asterisk queue and agents   Hall, Eric M.
• [asterisk-users] Asterisk queue and agents   Hall, Eric M.
• [asterisk-users] auto dialer   Hall, Eric M.
• [asterisk-users] auto dialer   Hall, Eric M.
• [asterisk-users] Asterisk queue and agents   Hall, Eric M.
• [asterisk-users] Asterisk queue and agents   Hall, Eric M.
• [asterisk-users] Asterisk queue and agents   Hall, Eric M.
• [asterisk-users] Asterisk queue and agents   Hall, Eric M.
• [asterisk-users] Asterisk queue and agents   Hall, Eric M.
• Spam? Re: [asterisk-users] cmd page crashes AsteriskSVN-branch-1.4-r57207   Hall, Eric M.
• Spam? Re: [asterisk-users] cmd page crashesAsteriskSVN-branch-1.4-r57207   Hall, Eric M.
• [asterisk-users] Re: Meetme question   Justin Hamade
• [asterisk-users] Which IP Phones have buttons can be assigned to functions with Asterisk   Justin Hamade
• [asterisk-users] Zaptel 1.4.0   Mike Hammett
• [asterisk-users] Snom 320 password   Mike Hammett
• [asterisk-users] Asterisk Realtime   Mike Hammett
• [asterisk-users] Asterisk Realtime   Mike Hammett
• [asterisk-users] IP addresses   Mike Hammett
• [asterisk-users] Asterisk Realtime   Mike Hammett
• [asterisk-users] Re: Asterisk Realtime   Mike Hammett
• [asterisk-users] Re: Asterisk Realtime   Mike Hammett
• [asterisk-users] Polycom and Asterisk   Mike Hammett
• [asterisk-users] Polycom and Asterisk   Mike Hammett
• [asterisk-users] SIP & NAT   Mike Hammett
• [asterisk-users] SIP & NAT   Mike Hammett
• [asterisk-users] help - UNSUBSCRIBE   Mike Hammett
• [asterisk-users] Polycom Power   Mike Hammett
• [asterisk-users] SIP & NAT   Mike Hammett
• [asterisk-users] SIP & NAT   Mike Hammett
• [asterisk-users] New Linksys SPA Daylight Saving Time Rule for US/Canada   Trevor G. Hammonds
• [asterisk-users] Queue announcing hold sequence instead of hold time   Trevor G. Hammonds
• [asterisk-users] Fix for TZ values updates for DST   Darrick Hartman
• [asterisk-users] Fix for TZ values updates for DST   Darrick Hartman
• [asterisk-users] Fix for TZ values updates for DST   Darrick Hartman
• [asterisk-users] voip-info.org status update   Darrick Hartman
• [asterisk-users] Re-parking (or transfer) a parked call   Barry D. Hassler
• [asterisk-users] Re-parking (or transfer) a parked call   Barry D. Hassler
• [asterisk-users] chan_misdn and Asterisk 1.4.1 - Early B3 not working any more   Gary Hawkins
• [asterisk-users] Problems with MFCR2 and Meridian   Angel Heart
• [asterisk-users] proposal: a new mailing list for asterisk 1.4, why not?   Angel Heart
• [asterisk-users] Conference server (or how to make a call with more than 3 u   Angel Heart
• [asterisk-users] How to resolve CallerID from AudioCodes FXO   Angel Heart
• [asterisk-users] Re: how to define a pilot number   Angel Heart
• [asterisk-users] snom led not working with asterisk 1.4.1   Jamie Heckford
• [asterisk-users] automated dialout detect forward   Mike Heininger
• [asterisk-users] automated dialout detect forward   Mike Heininger
• [asterisk-users] disable client side hangup after dialing 911   Jacob Helwig
• [asterisk-users] UK SIP Gateway   Gordon Henderson
• [asterisk-users] Re: Sending SMS   Gordon Henderson
• [asterisk-users] hanging an asterisk box off of a PBX analog extension   Gordon Henderson
• [asterisk-users] VoIP over Alvarion Wireless   Gordon Henderson
• [asterisk-users] Which VoIP router and switch to use for medium size business   Gordon Henderson
• [asterisk-users] Which VoIP router and switch to use for medium size business   Gordon Henderson
• [asterisk-users] asterisk on mini-itx   Gordon Henderson
• [asterisk-users] Which VoIP router and switch to use for medium size business   Gordon Henderson
• [asterisk-users] asterisk on mini-itx   Gordon Henderson
• [asterisk-users] Which VoIP router and switch to use for medium size business   Gordon Henderson
• [asterisk-users] asterisk on mini-itx   Gordon Henderson
• [asterisk-users] asterisk on mini-itx   Gordon Henderson
• [asterisk-users] What happend to voip-info?   Gordon Henderson
• [asterisk-users] voip-info.org status update   Gordon Henderson
• [asterisk-users] What happend to voip-info?   Gordon Henderson
• [asterisk-users] Dell poweredge 860 acceptable for office environment ?   Gordon Henderson
• [asterisk-users] Conference server (or how to make a call with more than 3 users)   Gordon Henderson
• [asterisk-users] Dell poweredge 860 acceptable for office environment ?   Gordon Henderson
• [asterisk-users] Zaptel silly issue   Gordon Henderson
• [asterisk-users] British Summertime & Grandstream Phones   Gordon Henderson
• [asterisk-users] Asterisk incoming caller id problem   Gordon Henderson
• [asterisk-users] Asterisk incoming caller id problem   Gordon Henderson
• [asterisk-users] Odd MeetMe bahaviour with MoH ...   Gordon Henderson
• [asterisk-users] Odd MeetMe bahaviour with MoH ...   Gordon Henderson
• [asterisk-users] wireless desktop phones   Gordon Henderson
• [asterisk-users] Transfering not working - how to debug?   Gordon Henderson
• [asterisk-users] Off Topic: Open Source USB Softphone   Gordon Henderson
• [asterisk-users] Off Topic: Open Source USB Softphone   Gordon Henderson
• [asterisk-users] Timeout for conferences   Andreas v. Heydwolff
• [asterisk-users] Configurations Files of TE110P   Joel Hill
• [asterisk-users] what happened to asterisk wiki???   Rizwan Hisham
• [asterisk-users] Meetme variables   Rizwan Hisham
• [asterisk-users] DNIS/DNID   Rizwan Hisham
• [asterisk-users] asterisk n-way call problem   Rizwan Hisham
• [asterisk-users] Transfer feature not working on asterisk 1.4.0   Rizwan Hisham
• [asterisk-users] Call counter for sip misbehaving   Rizwan Hisham
• [asterisk-users] Dial(Local/${EXTEN}@longdistance)?   Rizwan Hisham
• [asterisk-users] Conference server (or how to make a call with more than 3 u   Rizwan Hisham
• [asterisk-users] How to get AEL2   Rizwan Hisham
• [asterisk-users] reducing the number of extensions for every user   Rizwan Hisham
• [asterisk-users] How to get AEL2   Rizwan Hisham
• [asterisk-users] How to get AEL2   Rizwan Hisham
• [asterisk-users] Limit call duration   Rizwan Hisham
• [asterisk-users] using ael and extensions.conf togather?   Rizwan Hisham
• [asterisk-users] Dial(Local/${EXTEN}@longdistance)?   Rizwan Hisham
• [asterisk-users] Limit call duration   Rizwan Hisham
• [asterisk-users] Voicemailmain not changing password?   Rizwan Hisham
• [asterisk-users] Transfering not working - how to debug?   Rizwan Hisham
• [asterisk-users] Security on long distance calls   Rizwan Hisham
• [asterisk-users] Call dies when I press *   Rizwan Hisham
• [asterisk-users] Call Waiting problems   Rizwan Hisham
• [asterisk-users] Question on Priorities   Rizwan Hisham
• [asterisk-users] Question on Priorities   Rizwan Hisham
• [asterisk-users] Unsetting Global Vars   Johann Hoehn
• [asterisk-users] Unsetting Global Vars   Johann Hoehn
• [asterisk-users] Instant Messaging with SIP Softphone Eyebeam (was: SMS ON ASTERISK)   Anselm Martin Hoffmeister
• [asterisk-users] A New Phone Service - www.virtualphoneline.com   Anselm Martin Hoffmeister
• [asterisk-users] Microsoft launches first PABX   Anselm Martin Hoffmeister
• [asterisk-users] Microsoft launches first PABX   Anselm Martin Hoffmeister
• [asterisk-users] Doorphone vs. Grandstream BT101   Anselm Martin Hoffmeister
• [asterisk-users] wireless desktop phones   Anselm Martin Hoffmeister
• [asterisk-users] cisco 7902   Anselm Martin Hoffmeister
• [asterisk-users] DECT to SIP gateway experiences   Henning Holtschneider
• [asterisk-users] Gizmo project answers every call - can I use it in hunt group?   Russell Horn
• [asterisk-users] pickupgroup / SIP / Cisc phones   Michael Landin Hostbaek
• [asterisk-users] IAX2 and Faxing   Lee Howard
• [asterisk-users] IAX2 and Faxing   Lee Howard
• [asterisk-users] Configuring Faxs any help :)   Lee Howard
• [asterisk-users] Configuring Faxs any help :)   Lee Howard
• [asterisk-users] Configuring Faxs any help :)   Lee Howard
• [asterisk-users] Configuring Faxs any help :)   Lee Howard
• [asterisk-users] FAX & mISDN   Lee Howard
• [asterisk-users] REALTIME and VIEW ENTIRE DIALPLAN   Davis Sylvester III
• [asterisk-users] REALTIME and VIEW ENTIRE DIALPLAN   Davis Sylvester III
• [asterisk-users] GXV3000 & Speakphone   Davis Sylvester III
• [asterisk-users] IAX2 and Faxing   Davis Sylvester III
• [asterisk-users] IAX2 and Faxing   Davis Sylvester III
• [asterisk-users] While the VoIP-Info.org site is down...   Davis Sylvester III
• [asterisk-users] voip-info.org status update   Davis Sylvester III
• [asterisk-users] Re: Replacement Wiki - options (Formerly 'status of voip-info')   Davis Sylvester III
• [asterisk-users] E-911 and Asterisk   Davis Sylvester III
• [asterisk-users] E-911 and Asterisk   Davis Sylvester III
• [asterisk-users] E-911 and Asterisk   Davis Sylvester III
• [asterisk-users] Re: Replacement Wiki - options (Formerly 'status of voip-info')   Davis Sylvester III
• [asterisk-users] PASS CORRECT CALLER ID INFO TO OFFSITE TRANSFER   Davis Sylvester III
• [asterisk-users] Strange issue SIP URI to follow me busy signal   George A. Roberts IV
• [asterisk-users] snom led not working with asterisk 1.4.1   Giorgio Incantalupo
• [asterisk-users] snom led not working with asterisk 1.4.1   Giorgio Incantalupo
• [asterisk-users] proposal: a new mailing list for asterisk 1.4, why not?   Giorgio Incantalupo
• [asterisk-users] asterisk 1.4: choppy voicemail sound after upgrade from 1.2.9.1   Giorgio Incantalupo
• [asterisk-users] TDM400p, no CLI activity   Giorgio Incantalupo
• [asterisk-users] snom led not working with asterisk 1.4.1   Giorgio Incantalupo
• [asterisk-users] snom led not working with asterisk 1.4.1   Giorgio Incantalupo
• [asterisk-users] fax and mISDN: a chimera?   Giorgio Incantalupo
• [asterisk-users] HUD Lite server on Debian   Giorgio Incantalupo
• [asterisk-users] Moving from Bristuff to mISDN   Giorgio Incantalupo
• [asterisk-users] Moving from Bristuff to mISDN   Giorgio Incantalupo
• [asterisk-users] "Couldn't load variables.txt?aldope=xxxxx "   Giorgio Incantalupo
• [asterisk-users] * 1.4.1: connected to gtalk but no voice passing   Giorgio Incantalupo
• [asterisk-users] Can I generate random SIP traffic?   Giorgio Incantalupo
• [asterisk-users] GTalk/Jabber passing audio in 1.4.1!   Giorgio Incantalupo
• [asterisk-users] "Couldn't load variables.txt?aldope=xxxxx "   Giorgio Incantalupo
• [asterisk-users] sip: failed the authenticate on INVITE   Giorgio Incantalupo
• [asterisk-users] How to place a call to a Google Talk user?   Giorgio Incantalupo
• [asterisk-users] "Couldn't load variables.txt?aldope=xxxxx "   Giorgio Incantalupo
• [asterisk-users] "Couldn't load variables.txt?aldope=xxxxx "   Giorgio Incantalupo
• [asterisk-users] SNOM 360   Giorgio Incantalupo
• [asterisk-users] Problem with TE212P   Ioan Indreias
• [asterisk-users] Problem with TE212P   Ioan Indreias
• [asterisk-users] Configurations Files of TE110P   Ioan Indreias
• [asterisk-users] Rebooting all Aastra phones   Ioan Indreias
• [asterisk-users] asterisk on mini-itx   Ira
• [asterisk-users] Rebooting all Aastra phones   Ira
• [asterisk-users] TDM-400, Polycom SIP phones, and echo problems   Ira
• [asterisk-users] Refund from SellVoip?   Ira
• [asterisk-users] Refund from SellVoip?   Ira
• [asterisk-users] Refund from SellVoip?   Ira
• [asterisk-users] TDM400p reliability   Ira
• [asterisk-users] Digium cards on Vmware   Morten Isaksen
• [asterisk-users] Grandstream phones   JNA
• [asterisk-users] g711 -> iLBC garbled voice in 1.4?   Ray Jackson
• [asterisk-users] Asterisk incoming caller id problem   Mike Jagdis
• [asterisk-users] Asterisk incoming caller id problem   Mike Jagdis
• [asterisk-users] SIP OPTIONS dialog not understood   Raj Jain
• [asterisk-users] Asterisk hangs up SIP call after 6 200 retransmits   Raj Jain
• [asterisk-users] cisco 7905   Shaikh Jallaluddin
• [asterisk-users] How can I use the "GET VARIABLE variablename" in AGI   Lee Jenkins
• [asterisk-users] cmd page crashes Asterisk SVN-branch-1.4-r57207   Lee Jenkins
• [asterisk-users] FastAGI Question   Lee Jenkins
• [asterisk-users] Security and DTMF   Lee Jenkins
• [asterisk-users] FastAGI Question   Lee Jenkins
• [asterisk-users] AMI - DBPut   Lee Jenkins
• [asterisk-users] Call Back   Lee Jenkins
• [asterisk-users] How many outgoing phone line/voip account do I need?   Lee Jenkins
• [asterisk-users] what happened to asterisk wiki???   Lee Jenkins
• [asterisk-users] Cepstral voices   Lee Jenkins
• [asterisk-users] Cepstral voices   Lee Jenkins
• [asterisk-users] Cepstral voices   Lee Jenkins
• [asterisk-users] Cepstral voices   Lee Jenkins
• [asterisk-users] Conference server (or how to make a call with more than 3 u   Lee Jenkins
• [asterisk-users] Cepstral and numbers   Lee Jenkins
• [asterisk-users] Asterisk Automated Outbound Messaging   Lee Jenkins
• [asterisk-users] Re: how to define a pilot number   Lee Jenkins
• [asterisk-users] System from AMI   Lee Jenkins
• [asterisk-users] Re: System from AMI   Lee Jenkins
• [asterisk-users] Re: System from AMI   Lee Jenkins
• [asterisk-users] Info: Nokia E65 working with Asterisk   Robert Jenkins
• [asterisk-users] DECT to SIP gateway experiences   Robert Jenkins
• [asterisk-users] Cepstral voices   Kai-Uwe Jensen
• [asterisk-users] IAX2 certificate based (RSA) user auth in 1.4.x   Kai-Uwe Jensen
• [asterisk-users] asterisk on mini-itx   Sune Kloppenborg Jeppesen
• [asterisk-users] asterisk on mini-itx   Sune Kloppenborg Jeppesen
• [asterisk-users] asterisk on mini-itx   Sune Kloppenborg Jeppesen
• [asterisk-users] asterisk on mini-itx   Sune Kloppenborg Jeppesen
• [asterisk-users] help - UNSUBSCRIBE   Jerric
• [asterisk-users] dial question   Pavel Jezek
• [asterisk-users] In Asterisk 1.4.x, Why Digium has two H323 Channels   Pavel Jezek
• [asterisk-users] RE: In Asterisk 1.4.x, Why Digium has two H323 channels?   Pavel Jezek
• [asterisk-users] Cisco 7970 with skinny on * 1.4.1   Pavel Jezek
• [asterisk-users] Cisco 7970 with skinny on * 1.4.1   Pavel Jezek
• [asterisk-users] Cisco 7970 with skinny on * 1.4.1   Pavel Jezek
• [asterisk-users] SRTP testers needed   Pavel Jezek
• [asterisk-users] Asterisk and Fax   Andrew Joakimsen
• [asterisk-users] T38   Andrew Joakimsen
• [asterisk-users] blieve i my TE110P or My teleco provider ??   Andrew Joakimsen
• [asterisk-users] FAX using T38   Andrew Joakimsen
• [asterisk-users] multiple phones registered for the same user   Andrew Joakimsen
• [asterisk-users] FAX using T38   Andrew Joakimsen
• [asterisk-users] Digium S101i - pickupexten doesn't work   Andrew Joakimsen
• [asterisk-users] Voicemail to SMS using asterisk   Andrew Joakimsen
• [asterisk-users] Multiple simultaneous calls   Andrew Joakimsen
• [asterisk-users] A New Phone Service - www.virtualphoneline.com   Andrew Joakimsen
• [asterisk-users] Accessing Voicemail by dialing own number   Andrew Joakimsen
• [asterisk-users] Dell poweredge 860 acceptable for office environment ?   Andrew Joakimsen
• [asterisk-users] IAX2 and Faxing   Andrew Joakimsen
• [asterisk-users] DTMF not being detected with 1 provider. Works with the other provider...   Andrew Joakimsen
• [asterisk-users] FAX using T38   Andrew Joakimsen
• [asterisk-users] Microsoft launches first PABX   Andrew Joakimsen
• [asterisk-users] Zaptel silly issue   Andrew Joakimsen
• [asterisk-users] Teliax problems, they say use SIP, more mature & better working than IAX   Andrew Joakimsen
• [asterisk-users] Linksys/Sipura SPA-942 phones in larger deployments   Andrew Joakimsen
• [asterisk-users] ztdummy and MOH   Andrew Joakimsen
• [asterisk-users] TDM400p reliability   Andrew Joakimsen
• [asterisk-users] Doorphone vs. Grandstream BT101   Andrew Joakimsen
• [asterisk-users] Multi-registration ?   Andrew Joakimsen
• [asterisk-users] Polycom SoundPoint 501   Andrew Joakimsen
• [asterisk-users] Using server side phonebook directory with SPA941   Andrew Joakimsen
• [asterisk-users] ztdummy and MOH   Andrew Joakimsen
• [asterisk-users] Polycom and Asterisk   Andrew Joakimsen
• [asterisk-users] wireless desktop phones   Andrew Joakimsen
• [asterisk-users] Multi-line phones - Asterisk uses wrong callerid   Andrew Joakimsen
• [asterisk-users] Linksys SPA 3102 causing me problems   Andrew Joakimsen
• [asterisk-users] Multi-Level Queue   Andrew Joakimsen
• [asterisk-users] _ALERT_INFO replacement in 1.4?   Nikhil Jogia
• [asterisk-users] _ALERT_INFO replacement in 1.4?   Nikhil Jogia
• [asterisk-users] Building a new voicemail system... Testers needed!   Olle E Johansson
• [asterisk-users] Voicemail question   Olle E Johansson
• [asterisk-users] Real Time, sip.conf, [general]   Olle E Johansson
• [asterisk-users] asterisk and ssl   Olle E Johansson
• [asterisk-users] Asterisk Registering to other SIP servers.   Olle E Johansson
• [asterisk-users] sip show channels   Olle E Johansson
• [asterisk-users] Asterisk 1.4.1 Released   Olle E Johansson
• [asterisk-users] How to handle SIP-Callerid?   Olle E Johansson
• [asterisk-users] Hinting and Realtime   Olle E Johansson
• [asterisk-users] Packet2Packet Bridging Questions   Olle E Johansson
• [asterisk-users] How to enter bridge_native_loop???   Olle E Johansson
• [asterisk-users] disable client side hangup after dialing 911   Olle E Johansson
• [asterisk-users] How to enter bridge_native_loop???   Olle E Johansson
• [asterisk-users] How to access Voicemail Password in Asteriskwithout using V   Olle E Johansson
• [asterisk-users] How to access Voicemail Password in Asteriskwithout using V   Olle E Johansson
• [asterisk-users] Coming events in Europe   Olle E Johansson
• [asterisk-users] SIP unicode support ?   Olle E Johansson
• [asterisk-users] deprecated ALERT_INFO var andAMI's Originate command   Olle E Johansson
• [asterisk-users] _ALERT_INFO replacement in 1.4?   Olle E Johansson
• [asterisk-users] Update Asterisk 1.2.12 to 1.4.1 ?   Olle E Johansson
• [asterisk-users] Call counter for sip misbehaving   Olle E Johansson
• [asterisk-users] SRTP vs ZRTP in Asterisk   Olle E Johansson
• [asterisk-users] Polycom Questions   Bryan M. Johns
• [asterisk-users] Linksys/Sipura SPA-942 phones in larger deployments   Bryan M. Johns
• [asterisk-users] Linksys/Sipura SPA-942 phones in larger deployments   Bryan M. Johns
• [asterisk-users] Re: OT: Patch to OSS app for CDP without a Cisco switch - TESTERS WANTED   Jerry Jones
• [asterisk-users] Multi-Level Queue   Jerry Jones
• [asterisk-users] Need comparison between PBXtra, Trixbox, Thirdlane, Druid, Aheeva etc.   Senad Jordanovic
• [asterisk-users] Re: Two or More Bri Cards   Senad Jordanovic
• [asterisk-users] Re: Two or More Bri Cards   Senad Jordanovic
• [asterisk-users] Digium S101i - pickupexten doesn't work   Joseph
• [asterisk-users] Asterisk - e164 (enum) lookup confused   Joseph
• [asterisk-users] Digium S101i - Adapter DTMF works perfeclty   Joseph
• [asterisk-users] voicemail is not playing messages   Joseph
• [asterisk-users] voicemail is not playing messages   Joseph
• [asterisk-users] Re: What means: Request to schedule in the past?!?!   Martin Joseph
• [asterisk-users] Re: RE : SIP/IAX peers UNREACHABLE and audio loss   Martin Joseph
• [asterisk-users] Re: Refund from SellVoip?   Martin Joseph
• [asterisk-users] Activating Incoming Demo   Eddie Johnson Jr
• [asterisk-users] Packet2Packet Bridging Questions   Daryl Jurbala
• [asterisk-users] Packet2Packet Bridging Questions   Daryl Jurbala
• [asterisk-users] Asterisk-Addon-1.4.0 MySQL   KC
• [asterisk-users] Problem with "AT&T Maintenance" protocol in PRI connection, no B+D channels available   Kanelbullar
• [asterisk-users] Problem with "AT&T Maintenance" protocol in PRI connection, no B+D channels available   Kanelbullar
• [asterisk-users] Problem with "AT&T Maintenance" protocol in PRIconnection, no B+D channels available   Kanelbullar
• [asterisk-users] Problem with "AT&T Maintenance" protocol inPRI connection, no B+D channels available   Kanelbullar
• [asterisk-users] Asterisk Realtime   Philipp Kempgen
• [asterisk-users] queue information in mySQL   Philipp Kempgen
• [asterisk-users] Queue Announcements for Operators   Philipp Kempgen
• [asterisk-users] Hinting and Realtime   Philipp Kempgen
• AW: [asterisk-users] Hinting and Realtime   Philipp Kempgen
• [asterisk-users] Re: Asterisk Realtime   Philipp Kempgen
• [asterisk-users] Accessing Voicemail by dialing own number   Philipp Kempgen
• [asterisk-users] AEL #include file   Philipp Kempgen
• [asterisk-users] AEL #include file   Philipp Kempgen
• [asterisk-users] AEL #include file   Philipp Kempgen
• [asterisk-users] AEL #include file   Philipp Kempgen
• [asterisk-users] AEL #include file   Philipp Kempgen
• [asterisk-users] How to access Voicemail Password in Asterisk without using VoicemailMain????????   Philipp Kempgen
• [asterisk-users] How to access Voicemail Password in Asteriskwithout using V   Philipp Kempgen
• [asterisk-users] How to access Voicemail Password in Asteriskwithout using V   Philipp Kempgen
• [asterisk-users] camp on off-line phone   Philipp Kempgen
• [asterisk-users] camp on off-line phone   Philipp Kempgen
• [asterisk-users] Conference server (or how to make a call withmore than 3 u   Philipp Kempgen
• [asterisk-users] Dell poweredge 860 acceptable for office environment ?   Philipp Kempgen
• [asterisk-users] Dial(Local/${EXTEN}@longdistance)?   Philipp Kempgen
• [asterisk-users] help - UNSUBSCRIBE   Philipp Kempgen
• [asterisk-users] CallerID + Name   Philipp Kempgen
• [asterisk-users] Need help to strip variable   Philipp Kempgen
• [asterisk-users] Multi-Level Queue   Philipp Kempgen
• [asterisk-users] Realtime call-limit   Philipp Kempgen
• [asterisk-users] Realtime call-limit   Philipp Kempgen
• [asterisk-users] Indicating agent status on the phone   Philipp Kempgen
• [asterisk-users] Re: asterisk-users Digest, Vol 32, Issue 129   Philipp Kempgen
• [asterisk-users] UK SIP Gateway   Steve Kennedy
• [asterisk-users] Re: Sending SMS   Steve Kennedy
• [asterisk-users] Queues   Steve Kennedy
• [asterisk-users] Queues   Steve Kennedy
• [asterisk-users] UK BT PRI   Steve Kennedy
• [asterisk-users] UK PRI and outgoing CLI FYI   Steve Kennedy
• [asterisk-users] UK PRI and outgoing CLI FYI   Steve Kennedy
• [asterisk-users] 1.4 lost internet internal phones loose registration   Thomas Kenyon
• [Asterisk-Users] SIP to IAX - forcing codec pass thru   Thomas Kenyon
• [asterisk-users] Asterisk 1.4.1 Released   Thomas Kenyon
• [asterisk-users] Re: 1.4 lost internet internal phones loose registration   Thomas Kenyon
• [asterisk-users] T38   Khaled
• [asterisk-users] P-Asserted-Identify or Remote-Party-ID, or both?   Kristian Kielhofner
• [asterisk-users] Asterisk 1.4 Follow-Me Application   Kevin Kiely
• [asterisk-users] Follow me on multiple numbers..   Kevin Kiely
• [asterisk-users] Asterisk 1.4 Follow-Me Application   Kevin Kiely
• [asterisk-users] Follow me on multiple numbers..   Kevin Kiely
• [asterisk-users] gtalktovoip and Asteirsk   Klaverstyn, David C
• [asterisk-users] gtalk2voip and Asterisk   Klaverstyn, David C
• [asterisk-users] Call Back   Klaverstyn, David C
• [asterisk-users] Call Back   Klaverstyn, David C
• [asterisk-users] ztdummy and MOH   Klaverstyn, David C
• [asterisk-users] ztdummy and MOH   Klaverstyn, David C
• [asterisk-users] ztdummy and MOH   Klaverstyn, David C
• [asterisk-users] ztdummy and MOH   Klaverstyn, David C
• [asterisk-users] Cisco 7970 with skinny on * 1.4.1   Richard Klingler
• [asterisk-users] Cisco 7970 with skinny on * 1.4.1   Richard Klingler
• [asterisk-users] Cisco 7970 with skinny on * 1.4.1   Richard Klingler
• [asterisk-users] Cisco 7970 with skinny on * 1.4.1   Richard Klingler
• [asterisk-users] Remote host can't match request NOTIFY to call   Richard Klingler
• [asterisk-users] ARI with * 1.4.x   Richard Klingler
• [asterisk-users] ARI with * 1.4.2 won't display recordings   Richard Klingler
• [asterisk-users] asterisk-addons-1.4 write wrong uniqueid   Richard Klingler
• [asterisk-users] ztdummy and MOH   Wooi Koay
• [asterisk-users] ztdummy and MOH   Wooi Koay
• [asterisk-users] voip-info.org status update   Andrew Kohlsmith
• [asterisk-users] SIP Video Camera   KokMengLoh
• [asterisk-users] Re: Two or More Bri Cards   Stelios Koroneos
• [asterisk-users] Cancelling a digit in IVR   Kuba
• [asterisk-users] DISA and repeating calls   Kuba
• [asterisk-users] REG : H.323 Configurations with Asterisk   Anisha Kumar
• FW: [asterisk-users] REG : H.323 Configurations with Asterisk   Anisha Kumar
• [asterisk-users] How many outgoing phone line/voip account do I need?   Kurt Kuo
• [asterisk-users] how to interconnection asterisk(sip) with mera   Julyono Kurniawan
• [asterisk-users] How to get values of local channels context   Yuan LIU
• [asterisk-users] 2 Call locations   Yuan LIU
• [asterisk-users] How can I use the "GET VARIABLE variablename" inAGI   Yuan LIU
• [asterisk-users] DTMF from TDM400P and X100P   Yuan LIU
• [asterisk-users] DTMF from TDM400P and X100P   Yuan LIU
• [asterisk-users] How to fail an AGI   Yuan LIU
• [asterisk-users] Help Voicemail to SMS using asterisk   Yuan LIU
• [asterisk-users] Asterisk - e164 (enum) lookup confused   Yuan LIU
• [asterisk-users] How to fail an AGI   Yuan LIU
• [asterisk-users] When does local leg in call file start?   Yuan LIU
• [asterisk-users] When does local leg in call file start?   Yuan LIU
• [asterisk-users] Read() status?   Yuan LIU
• [asterisk-users] Compiling smsq in 1.2   Yuan LIU
• [asterisk-users] play file and action only stop if one definedkey has been p   Yuan LIU
• [asterisk-users] Compiling smsq in 1.2   Yuan LIU
• [asterisk-users] Noob Question   Yuan LIU
• [asterisk-users] Zaptel version for asterisk 1.2.16   Yuan LIU
• [asterisk-users] Re: Which SIP method/option to display a shorttext message   Yuan LIU
• [asterisk-users] DNIS/DNID   Yuan LIU
• [asterisk-users] Only secretary can call the boss, all othersonly reach the   Yuan LIU
• [asterisk-users] Nomination for Coolest App in 2007   Yuan LIU
• [asterisk-users] zttool always reports "OK" on TDM400P   Yuan LIU
• [asterisk-users] no special context for sip peer   Yuan LIU
• [asterisk-users] zttool always reports "OK" on TDM400P   Yuan LIU
• [asterisk-users] Which parameters of a live Asterisk server wouldyou monitor   Yuan LIU
• [asterisk-users] Asterisk Automated Outbound Messaging   Yuan LIU
• [asterisk-users] Limit call duration   Yuan LIU
• [asterisk-users] Noob question regarding PCI 2.x & TDM400P Card   Yuan LIU
• [asterisk-users] Asterisk with Dialplan or TrixBox for this case?   Yuan LIU
• [asterisk-users] asterisk: error while loading shared libraries: libiksemel.   Yuan LIU
• [asterisk-users] Need help to strip variable   Yuan LIU
• [asterisk-users] Multi-registration ?   Yuan LIU
• [asterisk-users] Interconnexion d'un serveur Asterisk à des PABX LG ( IP LDK   Yuan LIU
• [asterisk-users] Setting rxgain per channel   Yuan LIU
• [asterisk-users] bugetone 200's   Yuan LIU
• [asterisk-users] maximum simultaneous calls   Yuan LIU
• [asterisk-users] Unsetting Global Vars   Yuan LIU
• [asterisk-users] web based sip phone   Yuan LIU
• [asterisk-users] just on my LAN   Yuan LIU
• [asterisk-users] Can I generate random SIP traffic?   Yuan LIU
• Fwd: [asterisk-users] Multi-registration ?   Yuan LIU
• [asterisk-users] how to define a pilot number   Yuan LIU
• [asterisk-users] web based sip phone   Yuan LIU
• [asterisk-users] Refresher course needed!   Yuan LIU
• [asterisk-users] outbound call   Yuan LIU
• [asterisk-users] cutting hash in dial app   Yuan LIU
• [asterisk-users] Re: Paging   Yuan LIU
• [asterisk-users] Question on Priorities   Yuan LIU
• [asterisk-users] extra-sounds 1.4.5 timestapmed newer than 1.4.6 ???   Mr. James W. Laferriere
• [asterisk-users] web based sipphone   Pezhman Lali
• [asterisk-users] sip tunnel   Pezhman Lali
• [asterisk-users] h323   Pezhman Lali
• [asterisk-users] web based sip phone   Pezhman Lali
• [asterisk-users] web based sip phone   Pezhman Lali
• [asterisk-users] xten web phone   Pezhman Lali
• [asterisk-users] how to define a pilot number   Lito Lampitoc
• [asterisk-users] Re: how to define a pilot number   Lito Lampitoc
• [asterisk-users] BLF not working with Asterisk 1.4.0   Steve Langstaff
• [asterisk-users] Using Asterisk as Voicemail Server on a dinosaurMeridian System   Steve Langstaff
• [asterisk-users] wireless desktop phones   Steve Langstaff
• [asterisk-users] Multi-line phones - Asterisk uses wrong callerid   Steve Langstaff
• [asterisk-users] Help Needed: Can't make "local" calls on a brand new PRI   Andrew Latham
• [asterisk-users] Incase anyone wanted it - SNOM USA DST settings   Andrew Latham
• [asterisk-users] snom led not working with asterisk 1.4.1   Andrew Latham
• [asterisk-users] SNOM 360   Andrew Latham
• [asterisk-users] no special context for sip peer   Christophorus Laube
• [asterisk-users] How to fail an AGI   Lenz
• [asterisk-users] detecting missing end-of-queue records   Lenz
• [asterisk-users] automated dialout detect forward   Lenz
• [asterisk-users] asterisk on debian   Josu Lazkano Lete
• [asterisk-users] just call to user   Josu Lazkano Lete
• [asterisk-users] just on my LAN   Josu Lazkano Lete
• [asterisk-users] Question about DSP in Digium card   A. Levy
• [asterisk-users] Re: Question about DSP in Digium card   A. Levy
• [asterisk-users] GTalk/Jabber passing audio in 1.4.1!   Ronald Lewis
• [asterisk-users] Doorphone   Ola Lidholm
• [asterisk-users] Doorphone   Ola Lidholm
• [asterisk-users] Follow me on multiple numbers..   Philippe Lindheimer
• [asterisk-users] Follow me on multiple numbers..   Philippe Lindheimer
• [asterisk-users] Answer Confirmation with SIP/AIX channels   Philippe Lindheimer
• [asterisk-users] error in FreePBX   Philippe Lindheimer
• [asterisk-users] Limit call duration   Robert Lister
• [asterisk-users] automated dialout detect forward   Robert Lister
• [asterisk-users] FWD outgoing problem   Robert Lister
• [asterisk-users] Using server side phonebook directory with SPA941   Robert Lister
• [asterisk-users] asterisk on mini-itx   Mail Lists
• [asterisk-users] asterisk on mini-itx   Mail Lists
• [asterisk-users] asterisk on mini-itx   Mail Lists
• [asterisk-users] asterisk on mini-itx   Mail Lists
• [asterisk-users] Need comparison between PBXtra, Trixbox, Thirdlane, Druid, Aheeva etc.   Mailing Lists
• [asterisk-users] SIP & NAT   Alexander Lopez
• [asterisk-users] Re: System from AMI   Alexander Lopez
• [asterisk-users] Re: System from AMI   Alexander Lopez
• [asterisk-users] Development of new features in Asterisk Manager   Moacir O. de Souza Junior - Personalsoft Sistemas Ltda.
• RES: [asterisk-users] Development of new features in Asterisk Manager   Moacir O. de Souza Junior - Personalsoft Sistemas Ltda.
• RES: RES: [asterisk-users] Development of new features in AsteriskManager   Moacir O. de Souza Junior - Personalsoft Sistemas Ltda.
• RES: RES: RES: [asterisk-users] Development of new featuresin AsteriskManager   Moacir O. de Souza Junior - Personalsoft Sistemas Ltda.
• [asterisk-users] Problem with "AT&T Maintenance" protocol in PRI connection, no B+D channels available   Bruno De Luca
• [asterisk-users] ChanSpy and MeetMe   Bruno De Luca
• [asterisk-users] About FAX and PAP2   Lukas
• [asterisk-users] Re: About Pickup Grandstream   Lukas
• [asterisk-users] About FAX and PAP2   Lukas
• [asterisk-users] Re: About Pickup Grandstream   Lukas
• [asterisk-users] Re: About Pickup Grandstream   Lukas
• [asterisk-users] PickUp a call with feature pickup (*8) from a IAX2channel   Lukas
• [asterisk-users] accepting a call, macros, and key presses.   Lukassky
• [asterisk-users] Pickup over IAX   Lukassky
• [asterisk-users] Issues with a Linksys SPA 2102 and asterisk   Luki
• [asterisk-users] Number of SIP messages per minute   Luki
• [asterisk-users] Packetization Rate   Luki
• [asterisk-users] Linksys not Ringing   Luki
• [asterisk-users] SIP peer disappearing   Luki
• [asterisk-users] Problems with Zaptel Drivers   Lutgring, Sam
• [asterisk-users] Voicemail mailbox number passed in connection?   Lutgring, Sam
• OT: Re: [asterisk-users] voip-info.org status update   Richard Lyman
• [asterisk-users] Re: System from AMI   Richard Lyman
• RES: RES: [asterisk-users] Development of new features in AsteriskManager   Richard Lyman
• RES: RES: RES: [asterisk-users] Development of new featuresin AsteriskManager   Richard Lyman
• [asterisk-users] Re: System from AMI   Richard Lyman
• [asterisk-users] Re: System from AMI   Richard Lyman
• [asterisk-users] call file vs. originate   Richard Lyman
• [asterisk-users] REMOTE CRASH FIX   Mike Lynchfield
• [asterisk-users] REMOTE CRASH FIX   Mike Lynchfield
• [asterisk-users] Help: CallerID Name not being sent on outboundPRI trunk   Mike Lynchfield
• [asterisk-users] Help: CallerID Name not being sent on outboundPRI trunk   Mike Lynchfield
• [asterisk-users] Off Topic: Open Source USB Softphone   Mike Lynchfield
• [asterisk-users] Tesco Internet Phone   Julian Lyndon-Smith
• [asterisk-users] Tesco Internet Phone   Julian Lyndon-Smith
• [asterisk-users] How to access Voicemail Password in Asteriskwithout using V   Julian Lyndon-Smith
• [asterisk-users] Cepstral voices   Julian Lyndon-Smith
• [asterisk-users] Cepstral voices   Julian Lyndon-Smith
• [asterisk-users] Cepstral and numbers   Julian Lyndon-Smith
• [asterisk-users] Cepstral and numbers   Julian Lyndon-Smith
• [asterisk-users] How to access Voicemail Password in Asteriskwithout using V   Julian Lyndon-Smith
• [asterisk-users] UK PRI and outgoing CLI FYI   Julian Lyndon-Smith
• [asterisk-users] Anybody having problems using sellvoip?   Tom Lynn
• [asterisk-users] Cisco 7912   Tom Lynn
• [asterisk-users] Refund from SellVoip?   Tom Lynn
• [asterisk-users] Refund from SellVoip?   Tom Lynn
• [asterisk-users] Refund from SellVoip?   Tom Lynn
• [asterisk-users] Refund from SellVoip?   Tom Lynn
• [asterisk-users] Re: Refund from SellVoip?   Tom Lynn
• [asterisk-users] Anyone having trouble with claling US Domestic on Sellvoip?   Tom Lynn
• [asterisk-users] Anyone having trouble with claling US Domestic on Sellvoip?   Tom Lynn
• [asterisk-users] ztdummy and MOH   Tom Lynn
• [asterisk-users] Running Fax on E1 line   Doug Lytle
• [asterisk-users] RE: Polycom reject button   Doug Lytle
• [asterisk-users] When does local leg in call file start?   Doug Lytle
• [asterisk-users] Polycom Questions   Doug Lytle
• [asterisk-users] Background / Invalid Extension through cell phone   Doug Lytle
• [asterisk-users] DST changes for the US   Doug Lytle
• [asterisk-users] DST changes for the US   Doug Lytle
• [asterisk-users] Re-parking (or transfer) a parked call   Doug Lytle
• [asterisk-users] busy/hangup/answer detection in PRI E1 channels   Doug Lytle
• [asterisk-users] Configuring Faxs any help :)   Doug Lytle
• [asterisk-users] Problem with "AT&T Maintenance" protocol in PRI connection, no B+D channels available   Doug Lytle
• [asterisk-users] cause 127   Doug Lytle
• [asterisk-users] Doorphone vs. Grandstream BT101   Doug Lytle
• [asterisk-users] Semi-OT: Use T.38 ATAs to Extend fax lines   Doug Lytle
• [asterisk-users] Semi-OT: Use T.38 ATAs to Extend fax lines   Doug Lytle
• [asterisk-users] Doorphone vs. Grandstream BT101   Doug Lytle
• [asterisk-users] Doorphone vs. Grandstream BT101   Doug Lytle
• [asterisk-users] voicemail is not playing messages   Doug Lytle
• [asterisk-users] British Summertime & Grandstream Phones   Doug Lytle
• [asterisk-users] voicemail is not playing messages   Doug Lytle
• [asterisk-users] wireless desktop phones   Doug Lytle
• [asterisk-users] Transfering not working - how to debug?   Doug Lytle
• [asterisk-users] Set(CALLERID(all) not working with 'unknown' call?   Doug Lytle
• [asterisk-users] switchtype and signalling query   Doug Lytle
• [asterisk-users] Rebooting ALL polycom phones   Julian J. M.
• [asterisk-users] ChanSpy and MeetMe   Julian J. M.
• [asterisk-users] French PRI channel - exact signaling used   Cedric MILLET
• [asterisk-users] installation pb on debian etch   Philippe MONROUX
• [asterisk-users] In Asterisk 1.4.x, Why Digium has two H323 Channels   Thiago Maluf
• [asterisk-users] RE: In Asterisk 1.4.x, Why Digium has two H323 channels?   Thiago Maluf
• [asterisk-users] Meetme question   Adrian Marsh
• [asterisk-users] chan_misdn   Tiziano Martelli
• [asterisk-users] Building a new voicemail system... Testers needed!   John Marvin
• [asterisk-users] Zaptel version for asterisk 1.2.16   John Marvin
• [asterisk-users] Polycom Questions   Marty Mastera
• [asterisk-users] [G.729] Input Gain   Victor Mateevitsi
• [asterisk-users] Conference server (or how to make a call with more than 3 u   Victor Mateevitsi
• [asterisk-users] asterisk on debian   Victor Mateevitsi
• [asterisk-users] Help Needed: Can't make "local" calls on a brand new PRI   Matt
• [asterisk-users] Help: CallerID Name not being sent on outbound PRI trunk   Matt
• [asterisk-users] VoIP over Alvarion Wireless   Matt
• [asterisk-users] VoIP over Alvarion Wireless   Matt
• [asterisk-users] VoIP over Alvarion Wireless   Matt
• [asterisk-users] Re: Help: CallerID Name not being sent on outbound PRI trunk   Matt
• [asterisk-users] Rebooting all Aastra phones   Matt
• [asterisk-users] Rebooting all Aastra phones   Matt
• [asterisk-users] New to Asterisk   Matt
• [asterisk-users] Rebooting all Aastra phones   Matt
• [asterisk-users] LIDB/CNAM STORAGE DATABASE NEEDED   Matt
• [asterisk-users] Number of SIP messages per minute   Matt
• [asterisk-users] Number of SIP messages per minute   Matt
• [asterisk-users] Packetization Rate   Matt
• [asterisk-users] Packetization Rate   Matt
• [asterisk-users] While the VoIP-Info.org site is down...   Matt
• [asterisk-users] Voip-Wiki Site Information   Matt
• [asterisk-users] Packetization Rate   Matt
• [asterisk-users] voip-info.org status update   Matt
• [asterisk-users] Voip-Wiki Site Information   Matt
• [asterisk-users] Voip-Wiki Site Information   Matt
• [asterisk-users] Only secretary can call the boss, all others only reach the secretary when dial the boss extension   Matt
• [asterisk-users] E-911 and Asterisk   Matt
• [asterisk-users] Counting callers   Matt
• [asterisk-users] Getting an ASR Number   Matt
• [asterisk-users] Asterisk and T38 ?   Matt
• [asterisk-users] P-Asserted-Identify or Remote-Party-ID, or both?   Matt
• [asterisk-users] Park & No Announce?   Matt
• [asterisk-users] Park & No Announce?   Matt
• [asterisk-users] MOS Score   Matt
• [asterisk-users] Meetme cant handle more than 5 users in a call?? hmmmm   Matt
• [asterisk-users] Inbound Voice Quality - Speed Change   Matt
• [asterisk-users] Inbound Voice Quality - Speed Change   Matt
• [asterisk-users] Multi-line phones - Asterisk uses wrong callerid   Matt
• [asterisk-users] Re: Inbound Voice Quality - Speed Change   Matt
• [asterisk-users] Meetme cant handle more than 5 users in a call?? hmmmm   Matt
• [asterisk-users] voip-info.org status update   Mattt
• [asterisk-users] Looking for a terminations provider (carrier grade)   Mattt
• [asterisk-users] While the VoIP-Info.org site is down...   Patrick May
• [asterisk-users] Voice mail   Khayelihle Mbona
• [asterisk-users] Zap Channel Deadlocks   Ron McCarthy
• [asterisk-users] Zap Channel Deadlocks   Ron McCarthy
• [asterisk-users] AgentCallBackLogin Help!   Ron McCarthy
• [asterisk-users] Asterisk with Dialplan or TrixBox for this case?   Brian McEntire
• [asterisk-users] Extensions +International   McGhee, Stefano
• [asterisk-users] Asterisk 1.4 Realtime problems   Callum McGillivray
• [asterisk-users] SIP/Polycom Issue, Asterisk 1.2.16, calls dropped   Timothy McKee
• [asterisk-users] LUSYN patches   Marc McLaughlin
• [asterisk-users] Background / Invalid Extension through cell phone   Jorge Mendoza
• [asterisk-users] mobility with asterisk   Jorge Mendoza
• [asterisk-users] Asterisk Auto-dial out   Phil Menico
• [asterisk-users] Asterisk Auto-dial out   Phil Menico
• [asterisk-users] RE: Polycom reject button   Mike
• [asterisk-users] Rebooting ALL polycom phones   Mike
• [asterisk-users] Rebooting ALL polycom phones   Mike
• [asterisk-users] CDR and CallerID   Mike
• [asterisk-users] PoE - IEEE 802.3af   Mike
• [asterisk-users] PoE - IEEE 802.3af   Mike
• [asterisk-users] Doorphone vs. Grandstream BT101   Jay Milk
• [asterisk-users] Doorphone vs. Grandstream BT101   Jay Milk
• [asterisk-users] Doorphone vs. Grandstream BT101   Jay Milk
• [asterisk-users] Doorphone vs. Grandstream BT101   Jay Milk
• [asterisk-users] Doorphone vs. Grandstream BT101   Jay Milk
• [asterisk-users] Doorphone vs. Grandstream BT101   Jay Milk
• [asterisk-users] Chan_cellphone and CentOS 4.x   Jay Milk
• [asterisk-users] Doorphone vs. Grandstream BT101   Jay Milk
• [asterisk-users] codec_zap and Asterisk 1.4.1   Jeremiah Millay
• [asterisk-users] Asterisk 1.4.2 chan_zap   Jeremiah Millay
• [asterisk-users] Polycom Questions   Dave Miller
• [asterisk-users] Meetme cant handle more than 5 users in a call?? hmmmm   Dave Miller
• [asterisk-users] Polycom call parking feature and Asterisk call parking   Noah Miller
• [asterisk-users] Help! Echo problem even at T1 PRI?   Noah Miller
• [asterisk-users] Can't Compile w/HPEC   Noah Miller
• [asterisk-users] Re: Can't Compile w/HPEC   Noah Miller
• [asterisk-users] Re: Can't Compile w/HPEC   Noah Miller
• [asterisk-users] Re: Can't Compile w/HPEC   Noah Miller
• [asterisk-users] High Pitched Noise   Noah Miller
• [asterisk-users] hardware spec   Noah Miller
• [asterisk-users] Re: Polycom 601 loop   Noah Miller
• [asterisk-users] SIP registration   Noah Miller
• [asterisk-users] Question about DSP in Digium card   Noah Miller
• [asterisk-users] IAX Experiences [WAS: Question about DSP in Digium card]   Noah Miller
• [asterisk-users] TDM400p reliability   Noah Miller
• [asterisk-users] PoE - IEEE 802.3af   Noah Miller
• [asterisk-users] Polycom Power   Noah Miller
• [asterisk-users] Polycom 501 + Asterisk +Edit buttons   Noah Miller
• [asterisk-users] Polycom 501 + Asterisk +Edit buttons   Noah Miller
• [asterisk-users] Call dies when I press *   Noah Miller
• [asterisk-users] preventing voicemail pickup after SIP redirect ?   Louis-David Mitterrand
• [asterisk-users] Re: preventing voicemail pickup after SIP redirect ?   Louis-David Mitterrand
• [asterisk-users] polycom random reboots   Louis-David Mitterrand
• [asterisk-users] polycom random reboots   Louis-David Mitterrand
• [asterisk-users] polycom random reboots   Louis-David Mitterrand
• [asterisk-users] polycom random reboots   Louis-David Mitterrand
• [asterisk-users] polycom random reboots   Louis-David Mitterrand
• [asterisk-users] no incoming dad with mISDN 1.1.1 and asterisk?   Louis-David Mitterrand
• [asterisk-users] bad case of buzzing   Louis-David Mitterrand
• [asterisk-users] Asterisk Java w/ Threads   Jesus Mogollon
• [asterisk-users] Asterisk Java w/ Threads   Jesus Mogollon
• [asterisk-users] Help understanding SIP SHOW CHANNELS   Mojo with Horan & Company, LLC
• [asterisk-users] Test   Mojo with Horan & Company, LLC
• [asterisk-users] RE: Polycom reject button   Mojo with Horan & Company, LLC
• [asterisk-users] transfer function   Mojo with Horan & Company, LLC
• [asterisk-users] How to disable MOH completely?   Mojo with Horan & Company, LLC
• [asterisk-users] running asterisk through cellphone   Mojo with Horan & Company, LLC
• [asterisk-users] Empty Wildcard TDM400P as a MeetMe timer.   Mojo with Horan & Company, LLC
• [asterisk-users] Polycom call parking feature and Asterisk call parking   Mojo with Horan & Company, LLC
• [asterisk-users] Re: strange ring   Mojo with Horan & Company, LLC
• [asterisk-users] Sangoma A102d and Asterisk on Debian 3.1.   Markus Monka
• [asterisk-users] voip-info.org status update   Jay Moore
• [asterisk-users] Voicemail mailbox number passed in connection?   Jay Moore
• [asterisk-users] Device not registering after boot   Jay Moore
• [asterisk-users] modem passthru   William Moore
• [asterisk-users] auto dialer   Melcon Moraes
• [asterisk-users] auto dialer   Melcon Moraes
• [asterisk-users] Boot order of 2 TE110P and 1 TDM400P in the same machine   Melcon Moraes
• [asterisk-users] A200 card problem   Melcon Moraes
• [asterisk-users] DTMF detection problems on PRI channels?   Tony Mountifield
• [asterisk-users] Re: DTMF detection problems on PRI channels?   Tony Mountifield
• [asterisk-users] Re: Question about DSP in Digium card   Tony Mountifield
• [asterisk-users] Re: UK PRI and outgoing CLI FYI   Tony Mountifield
• [asterisk-users] Dial(Local/${EXTEN}@longdistance)?   Marco Mouta
• [asterisk-users] SER vs Asterisk?   Marco Mouta
• [asterisk-users] Asterisk 1.4 with Digium B410P - Timing problem   Marco Mouta
• [asterisk-users] build rpm fails   Devraj Mukherjee
• [asterisk-users] Re: build rpm fails   Devraj Mukherjee
• [asterisk-users] Re: build rpm fails   Devraj Mukherjee
• [asterisk-users] Asterisk 1.4.1   Steve Murphy
• [asterisk-users] AEL #include file   Steve Murphy
• [asterisk-users] AEL #include file   Steve Murphy
• [asterisk-users] play file and action only stop if one defined key has been pressed   Steve Murphy
• [asterisk-users] IAX2 audio issues   Steve Murphy
• [asterisk-users] Noob Question   Steve Murphy
• [asterisk-users] New to Asterisk   Steve Murphy
• [asterisk-users] snom led not working with asterisk 1.4.1   Steve Murphy
• [asterisk-users] proposal: a new mailing list for asterisk 1.4, why not?   Steve Murphy
• [asterisk-users] snom led not working with asterisk 1.4.1   Steve Murphy
• [asterisk-users] snom led not working with asterisk 1.4.1   Steve Murphy
• [asterisk-users] How to get AEL2   Steve Murphy
• [asterisk-users] Metaswitch help needed   Steve Murphy
• [asterisk-users] using ael and extensions.conf togather?   Steve Murphy
• [asterisk-users] Development of new features in Asterisk Manager   Steve Murphy
• RES: [asterisk-users] Development of new features in Asterisk Manager   Steve Murphy
• [asterisk-users] error in FreePBX   Steve Murphy
• [asterisk-users] L options in Dial() dont seem to work....   Steve Murphy
• [asterisk-users] Asterisk does not reINVITE after 302Redirect & 401Unauthorized   Mushtaq_Ahmed at 3com.com
• [asterisk-users] Asterisknow with video and X-Lite not quite working   Pari Nannapaneni
• [asterisk-users] Warning LSP Low   Rajeev Natarajan
• [asterisk-users] Re: Warning LSP Low   Rajeev Natarajan
• [asterisk-users] G729 'disappears' randomly   Rajeev Natarajan
• [asterisk-users] asterisk n-way call problem   Rajeev Natarajan
• [asterisk-users] Re: G729 'disappears' randomly   Rajeev Natarajan
• [asterisk-users] SIP/IAX peers UNREACHABLE and audio loss   Rajeev Natarajan
• [asterisk-users] Asterisk Registering to other SIP servers.   Bala Neelakantan
• [asterisk-users] Remote host can't match request NOTIFY to call   Bala Neelakantan
• [asterisk-users] camp on off-line phone   Leif Neland
• [asterisk-users] Dell poweredge 860 acceptable for officeenvironment ?   Leif Neland
• [asterisk-users] RE: Linksys PAP2 and Caller ID   Stewart Nelson
• [asterisk-users] TDM400p Loaded only once   Il Neofita
• [asterisk-users] TDM400p Loaded only once   Il Neofita
• [asterisk-users] Realtime Extensions and "Include"   Peder at NetworkOblivion
• [asterisk-users] Realtime Extensions and "Include"   Peder at NetworkOblivion
• [asterisk-users] DST changes for the US   Peder at NetworkOblivion
• [asterisk-users] Pickup some else's call   Peder at NetworkOblivion
• [asterisk-users] Need feedback on vitelity   Peder at NetworkOblivion
• [asterisk-users] Multi-Level Queue   Peder at NetworkOblivion
• [asterisk-users] Realtime call-limit   Peder at NetworkOblivion
• [asterisk-users] Multi-Level Queue   Peder at NetworkOblivion
• [asterisk-users] Re: asterisk-users Digest, Vol 32, Issue 21   Justin Newman
• [asterisk-users] Re: Coaching in asterisk   Justin Newman
• [asterisk-users] Re: Recorded file processing app wanted   Justin Newman
• [asterisk-users] Which hylafax client ?   Darren Nickerson
• [asterisk-users] VoIP over Alvarion Wireless   Andrew Niemantsverdriet
• [asterisk-users] VoIP over Alvarion Wireless   Andrew Niemantsverdriet
• [asterisk-users] Newbie Question   Chris Nighswonger
• [asterisk-users] Newbie Question   Chris Nighswonger
• [asterisk-users] Newbie Question   Chris Nighswonger
• [asterisk-users] Error compiling zaptel 1.4.0   Chris Nighswonger
• [asterisk-users] Error compiling zaptel 1.4.0   Chris Nighswonger
• [asterisk-users] Cisco 30VIP Phone   Chris Nighswonger
• [asterisk-users] Cisco 30VIP Phone   Chris Nighswonger
• [asterisk-users] Cisco 30VIP Phone   Chris Nighswonger
• [asterisk-users] Cisco 30VIP Phone   Chris Nighswonger
• [asterisk-users] Cisco 30VIP Phone   Chris Nighswonger
• [asterisk-users] Cisco 30VIP Phone   Chris Nighswonger
• [asterisk-users] Can be called on FreeWorldDialup/IAX channel, but can't make calls   Chris Nighswonger
• [asterisk-users] Can be called on FreeWorldDialup/IAX channel, but can't make calls   Chris Nighswonger
• [asterisk-users] Cisco 30VIP Phone   Chris Nighswonger
• [asterisk-users] Cisco 30VIP Phone   Chris Nighswonger
• [asterisk-users] Cisco 30VIP Phone   Chris Nighswonger
• [asterisk-users] Speed Dial Application in *   Chris Nighswonger
• [asterisk-users] Cisco 30VIP Phone   Chris Nighswonger
• [asterisk-users] Zaptel version for asterisk 1.2.16   John Novack
• [asterisk-users] Zaptel version for asterisk 1.2.16   John Novack
• [asterisk-users] A200 card problem   John Novack
• [asterisk-users] asterisk 1.2 branch revision 53132 failed to compile   John Novack
• [asterisk-users] asterisk 1.2 branch revision 53132 failed to compile   John Novack
• [asterisk-users] T1 cable for Digium T1/E1 Cards   John Novack
• [asterisk-users] Fedora + Linux Kernel 2.6 for Zaptel/Asterisk Installation   John Novack
• [asterisk-users] How to get AEL2   John Novack
• [asterisk-users] TDM400p reliability   John Novack
• [asterisk-users] Manager connection problems   Jordan Novak
• [asterisk-users] Sendmail and exchange for voicemail integration   Jordan Novak
• [asterisk-users] wireless desktop phones   Jordan Novak
• [asterisk-users] wireless desktop phones   Jordan Novak
• [asterisk-users] Queue priority   Jordan Novak
• [asterisk-users] queue priority causes crash   Jordan Novak
• [asterisk-users] queue priority   Jordan Novak
• [asterisk-users] Digium cards on Vmware   Massimo Nuvoli
• [asterisk-users] visdn, misdn and the hell   Massimo Nuvoli
• [asterisk-users] beronet BN4S0   Massimo Nuvoli
• [asterisk-users] CN=Diarmaid O'Loughlin/O=QAD1 is out of the office.   Diarmaid O'Loughlin
• [asterisk-users] Asterisk: recommended installation   Alejandro Cabrera Obed
• [asterisk-users] Problems with MFCR2 and Meridian   Arturo Ochoa
• [asterisk-users] Asterisk Automated Outbound Messaging   Arturo Ochoa
• [Asterisk-Users] Re: Pickup *8 with CallerID   Olivier
• [asterisk-users] Info: Nokia E65 working with Asterisk   Olivier
• [asterisk-users] Which hylafax client ?   Olivier
• [asterisk-users] Which SIP method/option to display a short text message ?   Olivier
• [asterisk-users] Re: Which SIP method/option to display a short text message ?   Olivier
• [asterisk-users] Re: Which SIP method/option to display a shorttext message   Olivier
• [asterisk-users] Dell poweredge 860 acceptable for office environment ?   Olivier
• [asterisk-users] Dell poweredge 860 acceptable for office environment ?   Olivier
• [asterisk-users] Dell poweredge 860 acceptable for office environment ?   Olivier
• [asterisk-users] Dell poweredge 860 acceptable for office environment ?   Olivier
• [asterisk-users] Dell poweredge 860 acceptable for office environment ?   Olivier
• [asterisk-users] Which parameters of a live Asterisk server would you monitor ?   Olivier
• [asterisk-users] Moving from Bristuff to mISDN   Olivier
• [asterisk-users] Moving from Bristuff to mISDN   Olivier
• [asterisk-users] How is this feature called ?   Olivier
• [asterisk-users] Multi-registration ?   Olivier
• [asterisk-users] Re: Two or More Bri Cards   Olivier
• Fwd: [asterisk-users] Multi-registration ?   Olivier
• Fwd: [asterisk-users] Multi-registration ?   Olivier
• [asterisk-users] Multi-registration ?   Olivier
• [asterisk-users] Re: How is this feature called ?   Olivier
• [asterisk-users] wireless desktop phones   Olivier
• [asterisk-users] SRTP vs ZRTP in Asterisk   Olivier
• [asterisk-users] Where are Spandsp changelogs or bugs available ?   Olivier
• [asterisk-users] Is it possible to install CCM on a Linux platform ?   Olivier
• [asterisk-users] Re: Multi-registration ?   Olivier
• [asterisk-users] Dropped calls in Asterisk - A general question   J. Oquendo
• [asterisk-users] SNOM 360   J. Oquendo
• [asterisk-users] mobility with asterisk   Alvaro Pacho
• [asterisk-users] RE: Polycom reject button   Kenneth Padgett
• [asterisk-users] proposal: a new mailing list for asterisk 1.4, why not?   Kenneth Padgett
• [asterisk-users] Microsoft launches first PABX   George Pajari
• [asterisk-users] Zapateller not playing audio via SIP Trunk?   Benoit Panizzon
• [asterisk-users] Zapateller not playing audio via SIP Trunk?   Benoit Panizzon
• [asterisk-users] Scratchy Audio with Asterisk 1.2.4 over IAX on FreeBSD?   Benoit Panizzon
• [asterisk-users] Alec Saunders post about Mashable Telco's   Tim Panton
• [asterisk-users] IAX best practices   Tim Panton
• [asterisk-users] Call load balancing   Tim Panton
• [asterisk-users] Playback 5% Too Fast?   Tim Panton
• [asterisk-users] FW: Seamless Multi Office Asterisk Deployment   Tim Panton
• [asterisk-users] Single sign on PC + phone?   Tim Panton
• [asterisk-users] Dell poweredge 860 acceptable for office environment ?   Tim Panton
• [asterisk-users] Dell poweredge 860 acceptable for office environment ?   Tim Panton
• [asterisk-users] Dell poweredge 860 acceptable for office environment ?   Tim Panton
• [asterisk-users] Teliax problems, they say use SIP, more mature & better working than IAX   Tim Panton
• [asterisk-users] Dell poweredge 860 acceptable for officeenvironment ?   Tim Panton
• [asterisk-users] transfer=mediaonly : can't hear nothing   Tim Panton
• [asterisk-users] wct4xxp problem   Tim Panton
• [asterisk-users] Re: wct4xxp problem   Tim Panton
• [asterisk-users] 1.4 - IAX2 - No registration for peer   Tim Panton
• [asterisk-users] Moving from Bristuff to mISDN   Tim Panton
• [asterisk-users] Multi-registration ?   Tim Panton
• [asterisk-users] SRTP vs ZRTP in Asterisk   Tim Panton
• [asterisk-users] web based sip phone   Tim Panton
• [asterisk-users] UK PRI and outgoing CLI FYI   Tim Panton
• [asterisk-users] wireless desktop phones   Tim Panton
• [asterisk-users] Monitor queue calls with Local channel agents   Miguel Paolino
• [asterisk-users] Re: queue information into db   Tomislav Parcina
• [asterisk-users] Re: fax support   Tomislav Parcina
• [asterisk-users] Re: SIP interface status and calllimit   Tomislav Parcina
• [asterisk-users] Revolution Call Accounting Desktop   Tomislav Parcina
• [asterisk-users] Re: queue information into db   Tomislav Parcina
• [asterisk-users] Re: Sending SMS   Tomislav Parcina
• [asterisk-users] Re: Registrations, how many is too many?   Tomislav Parcina
• [asterisk-users] Re: 1.4 lost internet internal phones loose registration   Tomislav Parcina
• [asterisk-users] Re: OT: Patch to OSS app for CDP without a Cisco switch - TESTERS WANTED   Tomislav Parcina
• [asterisk-users] Re: Asterisk Faxing Support   Tomislav Parcina
• [asterisk-users] Re: Digium cards on Vmware   Tomislav Parcina
• [asterisk-users] Re: FAX using T38   Tomislav Parcina
• [asterisk-users] Micros-Fidelio - billing in hotel   Tomislav Parcina
• [asterisk-users] Re: build rpm fails   Tomislav Parcina
• [asterisk-users] Re: build rpm fails   Tomislav Parcina
• [asterisk-users] Re: build rpm fails   Tomislav Parcina
• [asterisk-users] Re: Help: CallerID Name not being sent on outbound PRI trunk   Tomislav Parcina
• [asterisk-users] Re: visdn, misdn and the hell   Tomislav Parcina
• [asterisk-users] AMI - DBPut   Tomislav Parcina
• [asterisk-users] Re: queue information into db   Tomislav Parcina
• [asterisk-users] Re: Help: CallerID Name not being sent on outbound PRI trunk   Tomislav Parcina
• [asterisk-users] Re: AMI - DBPut   Tomislav Parcina
• [asterisk-users] Re: Number of SIP messages per minute   Tomislav Parcina
• [asterisk-users] Re: 1.4 compile issue   Tomislav Parcina
• [asterisk-users] Re: asterisk on mini-itx   Tomislav Parcina
• [asterisk-users] Re: What happend to voip-info?   Tomislav Parcina
• [asterisk-users] Re: While the VoIP-Info.org site is down...   Tomislav Parcina
• [asterisk-users] Re: While the VoIP-Info.org site is down...   Tomislav Parcina
• [asterisk-users] Re: voip-info.org status update   Tomislav Parcina
• [asterisk-users] Re: Cisco 7912   Tomislav Parcina
• [asterisk-users] Re: Call counter for sip misbehaving   Tomislav Parcina
• [asterisk-users] Re: Zaptel 1.2.16 Released   Tomislav Parcina
• [asterisk-users] Re: wrong values in duration and billsec in CDR   Tomislav Parcina
• [asterisk-users] Re: About Pickup Grandstream   Tomislav Parcina
• [asterisk-users] Re: wct4xxp problem   Tomislav Parcina
• [asterisk-users] Re: strange ring   Tomislav Parcina
• [asterisk-users] 302 Moved temporarely   Tomislav Parcina
• [asterisk-users] Re: wrong values in duration and billsec in CDR   Tomislav Parcina
• [asterisk-users] Re: strange ring   Tomislav Parcina
• [asterisk-users] Re: 302 Moved temporarely   Tomislav Parcina
• [asterisk-users] Re: wrong values in duration and billsec in CDR   Tomislav Parcina
• [asterisk-users] Re: wct4xxp problem   Tomislav Parcina
• [asterisk-users] Re: 302 Moved temporarely   Tomislav Parcina
• [asterisk-users] Re: SRTP testers needed   Tomislav Parcina
• [asterisk-users] AMI - delete voicemail   Tomislav Parcina
• [asterisk-users] Re: How is this feature called ?   Tomislav Parcina
• [asterisk-users] Re: SIP Video Camera   Tomislav Parcina
• [asterisk-users] System from AMI   Tomislav Parcina
• [asterisk-users] Re: AOC billing   Tomislav Parcina
• [asterisk-users] Re: just on my LAN   Tomislav Parcina
• [asterisk-users] Re: How is this feature called ?   Tomislav Parcina
• [asterisk-users] Re: System from AMI   Tomislav Parcina
• [asterisk-users] Re: System from AMI   Tomislav Parcina
• [asterisk-users] Re: System from AMI   Tomislav Parcina
• [asterisk-users] Re: Is it possible to install CCM on a Linux platform ?   Tomislav Parcina
• [asterisk-users] Can be called on FreeWorldDialup/IAX channel, but can't make calls   Timothy Parez
• [asterisk-users] Can be called on FreeWorldDialup/IAX channel, but can't make calls   Timothy Parez
• [asterisk-users] proposal: a new mailing list for asterisk 1.4, why not?   Jason Parker
• [asterisk-users] Cisco 30VIP Phone   Jason Parker
• [asterisk-users] Cisco 30VIP Phone   Jason Parker
• [asterisk-users] Cisco 30VIP Phone   Jason Parker
• [asterisk-users] PickUp a call with feature pickup (*8) from a IAX2 channel   Alvaro Parres
• [asterisk-users] PickUp a call with feature pickup (*8) from a IAX2channel   Alvaro Parres
• [asterisk-users] Single sign on PC + phone?   Patrick
• [asterisk-users] Single sign on PC + phone?   Patrick
• [asterisk-users] Too Many Open Files, Hung SIP Sessions, Can I Increase File Count?   Patrick
• [asterisk-users] SRTP testers needed   Patrick
• [asterisk-users] Re: build rpm fails   Thomas Patterson
• [asterisk-users] Noob Question   Thomas Patterson
• [asterisk-users] Asterisk 1.4.1   Paul
• [asterisk-users] voip-info.org status update   Paul
• [asterisk-users] voip-info.org status update   Paul
• [asterisk-users] E-911 and Asterisk   Paul
• [asterisk-users] Park & No Announce?   Paul
• [asterisk-users] Park & No Announce?   Paul
• [asterisk-users] Cannot hear ringback music from telco   Trevor Peirce
• [asterisk-users] Single sign on PC + phone?   Trevor Peirce
• [asterisk-users] Voip-Wiki Site Information   Trevor Peirce
• [asterisk-users] Single sign on PC + phone?   Trevor Peirce
• [asterisk-users] DNIS/DNID   Trevor Peirce
• [asterisk-users] Incoming Caller ID   Trevor Peirce
• [asterisk-users] Doorphone   Trevor Peirce
• [asterisk-users] CallerID + Name   Trevor Peirce
• [asterisk-users] Issues with a Linksys SPA 2102 and asterisk   Erick Perez
• [asterisk-users] Issues with a Linksys SPA 2102 and asterisk   Erick Perez
• [asterisk-users] Update Asterisk 1.2.12 to 1.4.1 ?   Noc Phibee
• [asterisk-users] Asterisk and T38 ?   Noc Phibee
• [asterisk-users] Asterisk and T38 ?   Noc Phibee
• [asterisk-users] UK SIP Gateway   Wilson Pickett
• [asterisk-users] Zaptel version for asterisk 1.2.16   Wilson Pickett
• [asterisk-users] Zaptel version for asterisk 1.2.16   Wilson Pickett
• [asterisk-users] asterisk 1.2 branch revision 53132 failed to compile   Wilson Pickett
• [asterisk-users] Re: Zaptel version for asterisk 1.2.16   Wilson Pickett
• [asterisk-users] asterisk 1.2 branch revision 53132 failed to compile   Wilson Pickett
• [asterisk-users] which spandsp for asterisk 1.2.16 (eom)   Wilson Pickett
• [Asterisk-Users] Sipura SPA-841 and firewall   Wilson Pickett
• [asterisk-users] FWD outgoing problem   Wilson Pickett
• [asterisk-users] Friday asterisk users live conference/podcast at 12:30PM EDT   Wilson Pickett
• [asterisk-users] Doorphone vs. Grandstream BT101   Wilson Pickett
• [asterisk-users] Re: Friday asterisk users live conference/podcast at 12:30PM EDT   Wilson Pickett
• [asterisk-users] Re: Friday asterisk users live conference/podcast at 12:30PM EDT   Wilson Pickett
• [asterisk-users] DECT to SIP gateway experiences   Daniel Pittman
• [asterisk-users] Teliax problems, they say use SIP, more mature & better working than IAX   Scott Plante
• [asterisk-users] outdial to phone for new VM notification   Porier, Jeremy M.
• [asterisk-users] One way intermittent static to PSTN   Porier, Jeremy M.
• [asterisk-users] voip-info.org status update   Jon Pounder
• [asterisk-users] While the VoIP-Info.org site is down...   Jon Pounder
• [asterisk-users] Microsoft launches first PABX   Jon Pounder
• [asterisk-users] Microsoft launches first PABX   Jon Pounder
• [asterisk-users] Microsoft launches first PABX   Jon Pounder
• [asterisk-users] Microsoft launches first PABX   Jon Pounder
• [asterisk-users] Microsoft launches first PABX   Jon Pounder
• [asterisk-users] Doorphone vs. Grandstream BT101   Jon Pounder
• [asterisk-users] HITBSecConf2007 - Malaysia: Call for Papers now Open   Praburaajan
• [asterisk-users] HITBSecConf2007 - Malaysia: Call for Papers now Open   Praburaajan
• [asterisk-users] Is Allison going to be banned from foreign travel over polar bears?   Steve Prior
• [asterisk-users] Cepstral voices   Steve Prior
• [asterisk-users] Cepstral voices   Steve Prior
• [asterisk-users] Cepstral and numbers   Steve Prior
• [asterisk-users] gtalktovoip and Asteirsk   Cosmin Prund
• [asterisk-users] Playback 5% Too Fast?   Cosmin Prund
• [asterisk-users] Cisco 7912   Matt Putnam
• [asterisk-users] Cisco 7912   Matt Putnam
• [asterisk-users] Cisco 7912   Matt Putnam
• [asterisk-users] Cisco 7912   Matt Putnam
• [asterisk-users] Linksys SPA 3102 causing me problems   Matt Putnam
• [asterisk-users] Double DTMF digits sent on IAX native bridge   Remi Quezada
• [asterisk-users] Double DTMF digits sent on IAX native bridge   Remi Quezada
• [asterisk-users] Double DTMF digits sent on IAX native bridge   Remi Quezada
• [asterisk-users] DNIS/DNID   Mark Quitoriano
• [asterisk-users] DNIS/DNID   Mark Quitoriano
• [asterisk-users] maximum simultaneous calls   Mark Quitoriano
• [asterisk-users] How to enter bridge_native_loop???   Santosh Raghuram
• [asterisk-users] How to enter bridge_native_loop???   Santosh Raghuram
• [asterisk-users] How to enter bridge_native_loop???   Santosh Raghuram
• [asterisk-users] Problem while using asterisk Realtime   Sanjay Rajdev
• [asterisk-users] SIP RTP Tunnel   Sanjay Rajdev
• [asterisk-users] Problem while using asterisk Realtime   Sanjay Rajdev
• [asterisk-users] Anyone having trouble with claling US Domestic on Sellvoip?   Balu Raman
• [asterisk-users] Asterisk and Fax   Robert A. Rawlinson
• [asterisk-users] L options in Dial() dont seem to work....   Mark Reardon
• [asterisk-users] L options in Dial() dont seem to work....   Mark Reardon
• [asterisk-users] voip-info.org status update   Walt Reed
• [asterisk-users] Dell poweredge 860 acceptable for office environment ?   Walt Reed
• [asterisk-users] Asterisk Realtime   Bruce Reeves
• [asterisk-users] Newbie extensions.conf question   Bruce Reeves
• [asterisk-users] rtsavesysname not working in 1.4   Bruce Reeves
• [asterisk-users] rtsavesysname not working in 1.4   Bruce Reeves
• [asterisk-users] A New Phone Service - www.virtualphoneline.com   Bruce Reeves
• [asterisk-users] Asterisk distributed deployment   Bruce Reeves
• [asterisk-users] OT Vonage V-Phone Adapter (Possible Hack)   Bruce Reeves
• [asterisk-users] _ALERT_INFO replacement in 1.4?   Bruce Reeves
• [asterisk-users] FW: Seamless Multi Office Asterisk Deployment   Bruce Reeves
• [asterisk-users] FW: Seamless Multi Office Asterisk Deployment   Bruce Reeves
• [asterisk-users] What is the best phone to get when using a headset?   Bruce Reeves
• [asterisk-users] FW: Seamless Multi Office Asterisk Deployment   Bruce Reeves
• [asterisk-users] Incoming Caller ID   Bruce Reeves
• [asterisk-users] Follow me on multiple numbers..   Bruce Reeves
• [asterisk-users] polycom random reboots   Bruce Reeves
• [asterisk-users] polycom random reboots   Bruce Reeves
• [asterisk-users] Dlink i2eye   Bruce Reeves
• [asterisk-users] Sendmail and exchange for voicemail integration   Bruce Reeves
• [asterisk-users] PC / Phone Combo   Bruce Reeves
• [asterisk-users] freepbx -> DB Error messages...   Bruce Reeves
• [asterisk-users] Chan_cellphone and CentOS 4.x   Bruce Reeves
• [asterisk-users] Chan_cellphone and CentOS 4.x   Bruce Reeves
• [asterisk-users] PoE - IEEE 802.3af   Bruce Reeves
• [asterisk-users] Polycom and Asterisk   Bruce Reeves
• [asterisk-users] Speed Dial Application in *   Bruce Reeves
• [asterisk-users] Speed Dial Application in *   Bruce Reeves
• [asterisk-users] Asterisk Java w/ Threads   Stefan Reuter
• [asterisk-users] Asterisk Java w/ Threads   Stefan Reuter
• [asterisk-users] Re: Asterisk Java w/ Threads   Stefan Reuter
• [asterisk-users] Re: Asterisk Java w/ Threads   Stefan Reuter
• [asterisk-users] managers   Stefan Reuter
• [asterisk-users] TDM400p reliability   Phil Reynolds
• [asterisk-users] Cisco + Asterisk list anyone?   Gary Richardson
• [asterisk-users] RE: Polycom reject button   JR Richardson
• [asterisk-users] Asterisk SIP to MAX TNT Gateway, Sporadic Echo   JR Richardson
• [asterisk-users] RE: what happened to asterisk wiki???   JR Richardson
• [asterisk-users] MAX TNT Question   JR Richardson
• [asterisk-users] Too Many Open Files, Hung SIP Sessions, Can I Increase File Count?   JR Richardson
• [asterisk-users] RE: Polycom 501 + Asterisk +Edit buttons   JR Richardson
• [asterisk-users] Re: Lucent TNT - ring timer   JR Richardson
• [asterisk-users] Polycom: warble on registration?   Steven Ringwald
• [asterisk-users] Manager.conf '127.0.0.1 unable to authenticate'   Alex Robar
• [asterisk-users] mobility with asterisk   Alex Robar
• [asterisk-users] DST changes for the US   Alex Robar
• [asterisk-users] Fix for TZ values updates for DST   Alex Robar
• [asterisk-users] Freepbx Incoming call's configuration   Alex Robar
• [asterisk-users] asterisk on debian   Alex Robar
• [asterisk-users] error, install freePbx   Alex Robar
• [asterisk-users] FWD outgoing problem   Alex Robar
• [asterisk-users] error in FreePBX   Alex Robar
• [asterisk-users] error in FreePBX   Alex Robar
• [asterisk-users] Voicemails with occasional speeded up portions   Anthony Rodgers
• [asterisk-users] Correct latency values in "sip show peers"   Rolz
• [asterisk-users] T1 cable for Digium T1/E1 Cards   Jeronimo Romero
• [asterisk-users] T1 cable for Digium T1/E1 Cards   Jeronimo Romero
• [asterisk-users] T1 cable for Digium T1/E1 Cards   Jeronimo Romero
• [asterisk-users] camp on off-line phone   Jeronimo Romero
• [asterisk-users] TE110P: Error ==> Asterisk died with code 1.   Jeronimo Romero
• [asterisk-users] Nice Transfer Feature   Jeronimo Romero
• [asterisk-users] While the VoIP-Info.org site is down...   Darryl Ross
• [asterisk-users] Call recording and archiving   Matthew J. Roth
• [asterisk-users] maximum simultaneous calls   Matthew J. Roth
• [asterisk-users] Choppy sound with chan_capi + Fritz Card USB   Christoph Rothe
• [asterisk-users] Asterisk Viruses?   Matthew Rubenstein
• [asterisk-users] Call load balancing   David Ruggles
• [asterisk-users] Call load balancing   David Ruggles
• [asterisk-users] Call load balancing   David Ruggles
• [asterisk-users] Call load balancing   David Ruggles
• [asterisk-users] Call load balancing   David Ruggles
• [asterisk-users] Cdr_mysql compile question   David Ruggles
• [asterisk-users] Cdr_mysql compile question   David Ruggles
• [asterisk-users] ACM question   David Ruggles
• [asterisk-users] ACM question   David Ruggles
• [asterisk-users] IAX2 Question (Asterisk 1.4 tarball)   David Ruggles
• [asterisk-users] IAX2 Question (Asterisk 1.4 tarball)   David Ruggles
• [asterisk-users] ExternalIVR() Dialplan function and Festival   David Ruggles
• [asterisk-users] ExternalIVR() Dialplan function and Festival   David Ruggles
• [asterisk-users] strange things on call transfer   MIVOC Systemes SAS
• [asterisk-users] strange things on call transfer   MIVOC Systemes SAS
• [asterisk-users] Issue with Calling Name ID in SIP: Asterisk sets Caller ID Number as Name if NO Name   Jean-Marc Salsa
• [asterisk-users] Real Time, sip.conf, [general]   André Santos
• [asterisk-users] Fix for TZ values updates for DST   Luis Claudio Santos
• [asterisk-users] Call Back   Luis Claudio Santos
• [asterisk-users] Erased log files   Luis Claudio Santos
• [asterisk-users] Off Topic: Open Source USB Softphone   Luis Claudio Santos
• [asterisk-users] wrong values in duration and billsec in CDR   Jovanny Saravia
• [asterisk-users] Re: Inbound Voice Quality - Speed Change   Travis Schafer
• [asterisk-users] Extensions +International   Rob Schall
• [asterisk-users] Extensions +International   Rob Schall
• [asterisk-users] Realtime Extensions and "Include"   Rob Schall
• [asterisk-users] Queue announcing hold sequence instead of hold time   Rob Schall
• [asterisk-users] Queue announcing hold sequence instead of hold time   Rob Schall
• [asterisk-users] Another Faxing Question   Rob Schall
• [asterisk-users] Rebooting ALL polycom phones   Rob Schall
• [asterisk-users] voip-info.org status update   Rob Schall
• [asterisk-users] Pickup some else's call   Rob Schall
• [asterisk-users] High Pitched Noise   Rob Schall
• [asterisk-users] High Pitched Noise   Rob Schall
• [asterisk-users] Asterisk Automated Outbound Messaging   Rob Schall
• [asterisk-users] High Pitched Noise   Rob Schall
• [asterisk-users] Polycom 501 + Asterisk +Edit buttons   Rob Schall
• [asterisk-users] CallerID + Name   Rob Schall
• [asterisk-users] CallerID + Name   Rob Schall
• [asterisk-users] Polycom 501 + Asterisk +Edit buttons   Rob Schall
• [asterisk-users] Current voip-info.org Status   David Schardin
• [asterisk-users] capi installation problem   Armin Schindler
• [asterisk-users] T1 Integrator Birch   John Schmerold
• [asterisk-users] SIP phone supporting more than 10 extension with acall transfer command   Griepentrog Scott
• [asterisk-users] Asterisk 1.2.13 Caller ID problem   Griepentrog Scott
• [asterisk-users] Somebody can help me?   Andrea Seghezzi
• [asterisk-users] Back to back E1 - asterisk <=> toshiba pbx - Call droping issue   Vidura Senadeera
• [asterisk-users] Re: Back to back E1 - asterisk <=> toshiba pbx - Call droping   Vidura Senadeera
• [asterisk-users] Fwd: Back to back E1 - asterisk <=> toshiba pbx - Call droping   Vidura Senadeera
• [asterisk-users] "pritimer" parameter in zapata.conf   Vidura Senadeera
• [asterisk-users] Re: Back to back E1 - asterisk <=> toshiba pbx - Call droping issue [ ref:00D36mPe.50032wycQ:ref ]   Vidura Senadeera
• [asterisk-users] busy/hangup/answer detection in PRI E1 channels   Vidura Senadeera
• [asterisk-users] Re: busy/hangup/answer detection in PRI E1 channels (Vidura Senadeera)   Vidura Senadeera
• [asterisk-users] SIP/IAX peers UNREACHABLE and audio loss   Edoardo Serra
• [asterisk-users] Re: SIP/IAX peers UNREACHABLE and audio loss   Edoardo Serra
• [asterisk-users] Re: RE : SIP/IAX peers UNREACHABLE and audio loss   Edoardo Serra
• [asterisk-users] Re: RE : SIP/IAX peers UNREACHABLE and audio loss   Edoardo Serra
• [asterisk-users] Re: RE : Re: RE : SIP/IAX peers UNREACHABLE and audio loss   Edoardo Serra
• [asterisk-users] Re: Two or More Bri Cards   Edoardo Serra
• [asterisk-users] Re: SIP/IAX peers UNREACHABLE and audio loss   Edoardo Serra
• [asterisk-users] Sponsored development - Monodirectional audio handling   Edoardo Serra
• [asterisk-users] Re: Sponsored development - Monodirectional audio handling   Edoardo Serra
• [asterisk-users] Virtual IP PBX Hosting   Service
• [asterisk-users] OT: Sipura DST Rules   Curt Shaffer
• [asterisk-users] Cisco + Asterisk list anyone?   Curt Shaffer
• [asterisk-users] Setting Sip Headers From Dial App?   Stuart Sheldon
• [asterisk-users] SIP/Polycom Issue, Asterisk 1.2.16, calls dropped   Stuart Sheldon
• [asterisk-users] Asterisk 1.4.2 Requires Zaptel from 1.4 svn branch for zap_chan?   Stuart Sheldon
• [asterisk-users] Asterisk 1.2.13 Caller ID problem   Igor Shmukler
• [asterisk-users] Asterisk 1.2.13 Caller ID problem   Igor Shmukler
• [asterisk-users] Unicall/R2 for Asterisk 1.4 Available for TESTING   Moises Silva
• [asterisk-users] Problems with MFCR2 and Meridian   Moises Silva
• [asterisk-users] What happend to voip-info?   Nir Simionovich
• [asterisk-users] A request for your input.   Jaswinder Singh
• [asterisk-users] Outbound SIP call from asterisk extension   Jaswinder Singh
• [asterisk-users] Failure acknowledgement time   Jaswinder Singh
• [asterisk-users] error in FreePBX   Jaswinder Singh
• [asterisk-users] Asterisk MSOutlook Dialer   San Singhania
• [asterisk-users] Digium b410p and 2.6.17 kernel bug?   Garth van Sittert
• [asterisk-users] asterisk: error while loading shared libraries: libiksemel.so.3: cannot open shared object file: No such file or directory   Dmitri Smirnoff
• [asterisk-users] ztdummy install in the new zaptel 1.4.1   Dmitri Smirnoff
• [asterisk-users] asterisk log analyzer   Eric Smith
• [asterisk-users] FW: Seamless Multi Office Asterisk Deployment   Rick Smith
• [asterisk-users] CDR Billing   Richard Soderblom
• [asterisk-users] chan_misdn   Richard Soderblom
• [asterisk-users] BLF not working with Asterisk 1.4.0   Andrey Solovjov
• [asterisk-users] proposal: a new mailing list for asterisk 1.4, why not?   Barzilai Spinak
• [asterisk-users] gtalk2voip and Asterisk   Mani Sridhar
• [asterisk-users] sip_nat.conf - Asterisk with two Ethernet Interfaces   Anjul Srivastava
• [asterisk-users] Re: sip_nat.conf - Asterisk with two Ethernet Interfaces   Anjul Srivastava
• [asterisk-users] Grandstream phones   Gerwin van de Steeg
• [asterisk-users] beronet BN4S0   Thomas Stein
• [asterisk-users] beronet BN8S0 and isdn phone   Thomas Stein
• [asterisk-users] cause 127   Thomas Stein
• [asterisk-users] cause 127   Thomas Stein
• [asterisk-users] UK SIP Gateway   Chris Stenton
• [asterisk-users] Citel Handset Gateway DST fix - FYI   Steven
• [asterisk-users] Re: Problem converting a Cisco 7960 to SIP   Brad Stockdale
• [asterisk-users] Re: Problem converting a Cisco 7960 to SIP   Brad Stockdale
• [asterisk-users] Tesco Internet Phone   Antony Stone
• [asterisk-users] Tesco Internet Phone   Antony Stone
• [asterisk-users] Tesco Internet Phone   Antony Stone
• [asterisk-users] Voicemail mailbox number passed in connection?   Antony Stone
• [asterisk-users] Zaptel silly issue   Brad Sumrall
• [asterisk-users] Refresher course needed!   Brad Sumrall
• [asterisk-users] Re: Question about DSP in Digium card   Brad Sumrall
• [asterisk-users] Need help making a voice record server $$$   Brad Sumrall
• [asterisk-users] Call connected, cannot hear or speak - $20 for fix   Supa
• [asterisk-users] Re: What happend to voip-info?   Supa
• [asterisk-users] Voicechanger update for asterisk 1.4   Supa
• [asterisk-users] Polycom SoundPoint 501   Paolo Supino
• [asterisk-users] Polycom SoundPoint 501   Paolo Supino
• [asterisk-users] Polycom SoundPoint 501   Paolo Supino
• [asterisk-users] Polycom SoundPoint 501   Paolo Supino
• [asterisk-users] chan_capi and only one B channel usable?   Torge Szczepanek
• [asterisk-users] Doorphone vs. Grandstream BT101   Administrator TOOTAI
• [asterisk-users] Doorphone vs. Grandstream BT101   Administrator TOOTAI
• [asterisk-users] Asterisk 1.4 with Digium B410P - Timing problem   Administrator TOOTAI
• [asterisk-users] Asterisk 1.4 with Digium B410P - Timing problem   Administrator TOOTAI
• [asterisk-users] running error, unable to load *.conf files: load_modules: No 'modules.conf' found - svn version 1.4.1   TZieleniewski
• [asterisk-users] running error, unable to load *.conf files: load_modules: No 'modules.conf' found - svn version 1.4.1   TZieleniewski
• [asterisk-users] running error: load_modules: No 'modules.conf' found - vesrion 1.4.1 from svn   TZieleniewski
• [asterisk-users] Any asterisk scripter around?   Sam Tam
• [asterisk-users] Cannot hear ringback music from telco   Vincent Tam
• [asterisk-users] Cannot hear ringback music from telco   Vincent Tam
• [asterisk-users] Help! Echo problem even at T1 PRI?   Vincent Tam
• [asterisk-users] Zaptel 1.2.15 Released   Asterisk Development Team
• [asterisk-users] Asterisk 1.2.16 Released   Asterisk Development Team
• [asterisk-users] Asterisk 1.4.1 Released   Asterisk Development Team
• [asterisk-users] Zaptel 1.2.16 Released   Asterisk Development Team
• [asterisk-users] Asterisk 1.2.17 Released   Asterisk Development Team
• [asterisk-users] Asterisk 1.4.2 Released   Asterisk Development Team
• [asterisk-users] Zaptel 1.4.1 Released   Asterisk Development Team
• [asterisk-users] So does 1.4.1 show up in the /branches/1.4, or only in the tags/1.4.1   Brad Templeton
• [asterisk-users] So does 1.4.1 show up in the /branches/1.4, or only in the tags/1.4.1   Brad Templeton
• [asterisk-users] So does 1.4.1 show up in the /branches/1.4, or only in the tags/1.4.1   Brad Templeton
• [asterisk-users] Nomination for Coolest App in 2007   Brad Templeton
• [asterisk-users] Nomination for Coolest App in 2007   Brad Templeton
• Bluetooth Re: [asterisk-users] Nomination for Coolest App in 2007   Brad Templeton
• [asterisk-users] Refund from SellVoip?   Brad Templeton
• [asterisk-users] Refund from SellVoip?   Brad Templeton
• [asterisk-users] Re: Refund from SellVoip?   Brad Templeton
• [asterisk-users] 3 way calling independent of phone hw.   Simon Tennant
• [asterisk-users] Re: build rpm fails   Axel Thimm
• [asterisk-users] Re: build rpm fails   Axel Thimm
• [asterisk-users] Re: build rpm fails   Axel Thimm
• [asterisk-users] rtsavesysname not working in 1.4   David Thomas
• [asterisk-users] rtsavesysname not working in 1.4   David Thomas
• [asterisk-users] How to disable MOH completely?   David Thomas
• [asterisk-users] rtsavesysname not working in 1.4   David Thomas
• [asterisk-users] How to disable MOH completely?   David Thomas
• [asterisk-users] A New Phone Service - www.virtualphoneline.com   David Thomas
• [asterisk-users] How to disable MOH completely?   David Thomas
• [asterisk-users] voip-info.org status update   James H Thompson
• [asterisk-users] Sped up recordings with 1.4.1   Joshua Thompson
• [asterisk-users] Bridged ZAP calls do not release   Mitch Thompson
• [asterisk-users] Press quotes needed from ENUM users on Asterisk   John Todd
• [asterisk-users] T1 cable for Digium T1/E1 Cards   Tom
• [asterisk-users] Indicating agent status on the phone   Alexander Topolanek
• [asterisk-users] Re: Zaptel version for asterisk 1.2.16   Brent Torrenga
• [asterisk-users] Multiple simultaneous calls   Stefano Totaro
• [asterisk-users] about bluetooth channel   Steve Totaro
• [asterisk-users] Cannot hear ringback music from telco   Steve Totaro
• [asterisk-users] IAX best practices   Steve Totaro
• [asterisk-users] Multiple simultaneous calls   Steve Totaro
• [asterisk-users] Multiple simultaneous calls   Steve Totaro
• [asterisk-users] Re: Sending SMS   Steve Totaro
• [asterisk-users] Cannot hear ringback music from telco   Steve Totaro
• [asterisk-users] IAX best practices   Steve Totaro
• [asterisk-users] Asterisk and Fax   Steve Totaro
• [asterisk-users] Alec Saunders post about Mashable Telco's   Steve Totaro
• [asterisk-users] IAX best practices   Steve Totaro
• [asterisk-users] Alec Saunders post about Mashable Telco's   Steve Totaro
• [asterisk-users] Help Voicemail to SMS using asterisk   Steve Totaro
• [asterisk-users] IP addresses   Steve Totaro
• [asterisk-users] Re: Sending SMS   Steve Totaro
• [asterisk-users] Using Asterisk as Voicemail Server on a dinosaur Meridian System   Steve Totaro
• [asterisk-users] Long term voicemail archival and synchronisationbetween multiple storage locations?   Steve Totaro
• [asterisk-users] Nomination for Coolest App in 2007   Steve Totaro
• [asterisk-users] Back to back E1 - asterisk <=> toshiba pbx - Calldroping issue   Steve Totaro
• [asterisk-users] Background / Invalid Extension through cell phone   Steve Totaro
• [asterisk-users] OT Vonage V-Phone Adapter (Possible Hack)   Steve Totaro
• [asterisk-users] Background / Invalid Extension through cell phone   Steve Totaro
• [asterisk-users] OT Vonage V-Phone Adapter (Possible Hack)   Steve Totaro
• [asterisk-users] asterisk and ssl   Steve Totaro
• [asterisk-users] queue information in mySQL   Steve Totaro
• [asterisk-users] Empty Wildcard TDM400P as a MeetMe timer.   Steve Totaro
• [asterisk-users] Empty Wildcard TDM400P as a MeetMe timer.   Steve Totaro
• [asterisk-users] Fwd: Back to back E1 - asterisk <=> toshiba pbx -Call droping   Steve Totaro
• [asterisk-users] Queue Announcements for Operators   Steve Totaro
• [asterisk-users] Asterisk + Panasonic pbx   Steve Totaro
• [asterisk-users] Asterisk distributed deployment   Steve Totaro
• [asterisk-users] Re: Asterisk distributed deployment   Steve Totaro
• [asterisk-users] Sender phone ringing while recipient talking   Steve Totaro
• [asterisk-users] Empty Wildcard TDM400P as a MeetMe timer.   Steve Totaro
• [asterisk-users] Re: Coaching in asterisk   Steve Totaro
• [asterisk-users] Sender phone ringing while recipient talking   Steve Totaro
• [asterisk-users] OT Vonage V-Phone Adapter (Possible Hack)   Steve Totaro
• [asterisk-users] RE: Coaching in asterisk   Steve Totaro
• [asterisk-users] How to enter bridge_native_loop???   Steve Totaro
• [asterisk-users] IAX2 audio issues   Steve Totaro
• [asterisk-users] Re: Help: CallerID Name not being sent on outbound PRI trunk   Steve Totaro
• [asterisk-users] How to best manage my dial plans as the continueto grow, and grow, and grow....   Steve Totaro
• [asterisk-users] Re: Recorded file processing app wanted   Steve Totaro
• [asterisk-users] Re: Recorded file processing app wanted   Steve Totaro
• [asterisk-users] Empty Wildcard TDM400P as a MeetMe timer.   Steve Totaro
• [asterisk-users] Recorded file processing app wanted   Steve Totaro
• [asterisk-users] Create meetme conference rooms on the flight.   Steve Totaro
• [asterisk-users] What happend to voip-info?   Steve Totaro
• [asterisk-users] T1 Integrator Birch   Steve Totaro
• [asterisk-users] Nomination for Coolest App in 2007   Steve Totaro
• [asterisk-users] What is the best phone to get when using a headset?   Steve Totaro
• [asterisk-users] Manager connection problems   Steve Totaro
• [asterisk-users] Manager connection problems   Steve Totaro
• [asterisk-users] what happened to asterisk wiki???   Steve Totaro
• [asterisk-users] Call center manager for Asterisk (Release 0.3)   Steve Totaro
• [asterisk-users] what happened to asterisk wiki???   Steve Totaro
• [asterisk-users] While the VoIP-Info.org site is down...   Steve Totaro
• [asterisk-users] While the VoIP-Info.org site is down...   Steve Totaro
• [asterisk-users] Any asterisk scripter around?   Steve Totaro
• [asterisk-users] Teliax problems, they say use SIP, more mature & better working than IAX   Steve Totaro
• [asterisk-users] Configuring Faxs any help :)   Steve Totaro
• [asterisk-users] Festival works extension to extension, not on trunk   Steve Totaro
• [asterisk-users] Zaptel silly issue   Steve Totaro
• [asterisk-users] Dell poweredge 860 acceptable for office environment ?   Steve Totaro
• [asterisk-users] Configuring Faxs any help :)   Steve Totaro
• [asterisk-users] Re: wct4xxp problem   Steve Totaro
• [asterisk-users] managers   Steve Totaro
• [asterisk-users] Re: wct4xxp problem   Steve Totaro
• [asterisk-users] Microsoft launches first PABX   Steve Totaro
• [asterisk-users] Doorphone vs. Grandstream BT101   Steve Totaro
• [asterisk-users] Question about DSP in Digium card   Steve Totaro
• [asterisk-users] Doorphone vs. Grandstream BT101   Steve Totaro
• [asterisk-users] the age old telephone tree... why re-invent the wheel?   Steve Totaro
• [asterisk-users] Doorphone vs. Grandstream BT101   Steve Totaro
• [asterisk-users] Doorphone vs. Grandstream BT101   Steve Totaro
• [asterisk-users] SER vs Asterisk?   Steve Totaro
• [asterisk-users] Doorphone   Steve Totaro
• [asterisk-users] Doorphone vs. Grandstream BT101   Steve Totaro
• [asterisk-users] Timeouts not working   Richard Trenchard
• [asterisk-users] Create meetme conference rooms on the flight.   Tristan
• [asterisk-users] Re: While the VoIP-Info.org site is down...   Tristan
• [asterisk-users] DTMF Corruption Problem in 1.4.2 for SIP RFC2833 plz halp   Justin Tunney
• [asterisk-users] wct4xxp problem   Matija Turk
• [asterisk-users] How to place a call to a Google Talk user?   Am Turnip
• [asterisk-users] Call Back   Vernier Umali
• [asterisk-users] FAX using T38   Steve Underwood
• [asterisk-users] FAX using T38   Steve Underwood
• [asterisk-users] how to determine the remote IP address when dialing in ooh323   VoIP User
• [asterisk-users] Re: Sending SMS   Jens Vagelpohl
• [asterisk-users] Info: Nokia E65 working with Asterisk   Jens Vagelpohl
• [asterisk-users] Voicemail question   Race Vanderdecken
• [asterisk-users] Using server side phonebook directory with SPA941   Maxim Veksler
• [asterisk-users] Using server side phonebook directory with SPA941   Maxim Veksler
• [asterisk-users] IAX2 audio issues   Vulpes Velox
• [asterisk-users] IAX2 audio issues   Vulpes Velox
• [asterisk-users] IAX2 audio issues   Vulpes Velox
• [asterisk-users] IAX2 audio issues   Vulpes Velox
• [asterisk-users] AMI Originate and release channels   Paulo Vicentini
• [asterisk-users] sip tunnel   Vicky
• [asterisk-users] Refund from SellVoip?   Vicky
• [asterisk-users] Incoming Caller ID   Rob Vinson
• [asterisk-users] Doorphone   Ray Wadkins
• [asterisk-users] Doorphone   Ray Wadkins
• [asterisk-users] Doorphone   Ray Wadkins
• [asterisk-users] Doorphone   Ray Wadkins
• [asterisk-users] Polycom reject button   Jason Walker
• [asterisk-users] 2 Call locations   Jason Walker
• [asterisk-users] RE: Polycom reject button   Jason Walker
• [asterisk-users] RE: Polycom reject button   Jason Walker
• [asterisk-users] Linksys not Ringing   Jason Walker
• [asterisk-users] Linksys not Ringing   Jason Walker
• [asterisk-users] GTalk/Jabber passing audio in 1.4.1!   Charles Wang
• [asterisk-users] A New Phone Service - www.virtualphoneline.com   Singer Wang
• [asterisk-users] Re: Help: CallerID Name not being sent   Matthew Warren
• [asterisk-users] Help: CallerID Name not being sent on outboundPRI trunk   Webster, Andrew
• [asterisk-users] Help: CallerID Name not being sent on outboundPRI trunk   Webster, Andrew
• [asterisk-users] No Caller ID Name PRI NI2   Webster, Andrew
• [asterisk-users] Number of groups?   Webster, Andrew
• [asterisk-users] Cisco 7912   Hermann Wecke
• [asterisk-users] Cisco 7912   Hermann Wecke
• [asterisk-users] zttool always reports "OK" on TDM400P   Hermann Wecke
• [asterisk-users] asterisk on debian   Hermann Wecke
• [asterisk-users] Cisco 7970 with skinny on * 1.4.1   Hermann Wecke
• [asterisk-users] Asterisk x Mera MVTS   Hermann Wecke
• [asterisk-users] Problem in using Two BRi Cards in Asterisk   Hermann Wecke
• [asterisk-users] cisco 7905   Hermann Wecke
• [asterisk-users] Re: Problem converting a Cisco 7960 to SIP   Ron Wellsted
• [asterisk-users] PoE - IEEE 802.3af   Michael Welter
• [asterisk-users] chan_capi and only one B channel usable?   Karsten Wemheuer
• [asterisk-users] REMOTE CRASH FIX   BJ Weschke
• [asterisk-users] Asterisk queue and agents   BJ Weschke
• [asterisk-users] Asterisk queue and agents   BJ Weschke
• [asterisk-users] Asterisk queue and agents   BJ Weschke
• [asterisk-users] RE: Coaching in asterisk   BJ Weschke
• [asterisk-users] Queue announcing hold sequence instead of hold time   BJ Weschke
• [asterisk-users] Asterisk 1.4 Follow-Me Application   BJ Weschke
• [asterisk-users] Follow me on multiple numbers..   BJ Weschke
• [asterisk-users] Queues   BJ Weschke
• [asterisk-users] T1 cable for Digium T1/E1 Cards   BJ Weschke
• [asterisk-users] T1 cable for Digium T1/E1 Cards   BJ Weschke
• [asterisk-users] T1 cable for Digium T1/E1 Cards   BJ Weschke
• [asterisk-users] Queue App - Free agent and waiting calls   BJ Weschke
• [asterisk-users] queue priority   BJ Weschke
• [asterisk-users] Multi-Level Queue   BJ Weschke
• [asterisk-users] Is the 1.0.X branch vulnerable to the SIP issue?   Thermal Wetland
• [asterisk-users] 1.4 sample postgresql configs   Derek Whitten
• [asterisk-users] FWD outgoing problem   Derek Whitten
• [asterisk-users] Re: Fedora + Linux Kernel 2.6 for Zaptel/Asterisk Installation   Derek Whitten
• [asterisk-users] Cisco 30VIP Phone   Derek Whitten
• [asterisk-users] Cisco 30VIP Phone   Derek Whitten
• [asterisk-users] PRI progress codes.   Eric "ManxPower" Wieling
• [asterisk-users] hanging an asterisk box off of a PBX analog extension   Eric "ManxPower" Wieling
• [asterisk-users] When does local leg in call file start?   Eric "ManxPower" Wieling
• [asterisk-users] Digium S101i - pickupexten doesn't work   Eric "ManxPower" Wieling
• [asterisk-users] Configurations Files of TE110P   Eric "ManxPower" Wieling
• [asterisk-users] Rx+,Rx-,Tx+,Tx- of TE110P   Eric "ManxPower" Wieling
• [asterisk-users] Re: Asterisk Java w/ Threads   Eric "ManxPower" Wieling
• [asterisk-users] preventing voicemail pickup after SIP redirect ?   Eric "ManxPower" Wieling
• [asterisk-users] Polycom Questions   Eric "ManxPower" Wieling
• [asterisk-users] Background / Invalid Extension through cell phone   Eric "ManxPower" Wieling
• [asterisk-users] Back to back E1 - asterisk <=> toshiba pbx - Call droping issue   Eric "ManxPower" Wieling
• [asterisk-users] Background / Invalid Extension through cell phone   Eric "ManxPower" Wieling
• [asterisk-users] No application 'Prefix' for extension in1.2x, what app I have to use instead?   Eric "ManxPower" Wieling
• [asterisk-users] Issues with a Linksys SPA 2102 and asterisk   Eric "ManxPower" Wieling
• [asterisk-users] Polycom call parking feature and Asterisk callparking   Eric "ManxPower" Wieling
• [asterisk-users] Number of SIP messages per minute   Eric "ManxPower" Wieling
• [asterisk-users] How to match wild card inside a GoToIf?   Eric "ManxPower" Wieling
• [asterisk-users] Incoming Caller ID   Eric "ManxPower" Wieling
• [asterisk-users] zttool always reports "OK" on TDM400P   Eric "ManxPower" Wieling
• [asterisk-users] High Pitched Noise   Eric "ManxPower" Wieling
• [asterisk-users] 302 Moved temporarely   Eric "ManxPower" Wieling
• [asterisk-users] Re: 302 Moved temporarely   Eric "ManxPower" Wieling
• [asterisk-users] PickUp a call with feature pickup (*8) from a IAX2channel   Eric "ManxPower" Wieling
• [asterisk-users] Doorphone vs. Grandstream BT101   Eric "ManxPower" Wieling
• [asterisk-users] Re: format_wav.c:247 update_header: Unable to find our position   Eric "ManxPower" Wieling
• [asterisk-users] Re: Polycom 601 loop   Eric "ManxPower" Wieling
• [asterisk-users] Re: how to define a pilot number   Eric "ManxPower" Wieling
• [asterisk-users] ztdummy and MOH   Eric "ManxPower" Wieling
• [asterisk-users] Re: Re: Inbound Voice Quality - Speed Change   Eric "ManxPower" Wieling
• [asterisk-users] Using server side phonebook directory with SPA941   Eric "ManxPower" Wieling
• [asterisk-users] Dialplan Streaming   Eric "ManxPower" Wieling
• [asterisk-users] Dialplan Streaming   Eric "ManxPower" Wieling
• [asterisk-users] L options in Dial() dont seem to work....   Eric "ManxPower" Wieling
• [asterisk-users] SIP & NAT   Eric "ManxPower" Wieling
• [asterisk-users] Polycom Power   Eric "ManxPower" Wieling
• [asterisk-users] SIP & NAT   Eric "ManxPower" Wieling
• [asterisk-users] Sipura SPA2000 Transfer Call   Eric "ManxPower" Wieling
• [asterisk-users] switchtype and signalling query   Eric "ManxPower" Wieling
• [asterisk-users] Manager.conf '127.0.0.1 unable to authenticate'   Ken Williams
• [asterisk-users] Manager.conf '127.0.0.1 unable to authenticate'   Ken Williams
• [asterisk-users] GXP-2000 DST Change   Ken Williams
• [asterisk-users] How to match wild card inside a GoToIf?   Ken Williams
• [asterisk-users] How to match wild card inside a GoToIf?   Ken Williams
• [asterisk-users] What happend to voip-info?   Ken Williams
• [asterisk-users] Park & No Announce?   Ken Williams
• [asterisk-users] Park & No Announce?   Ken Williams
• [asterisk-users] Park & No Announce?   Ken Williams
• [asterisk-users] Park & No Announce?   Ken Williams
• [asterisk-users] Park & No Announce?   Ken Williams
• [asterisk-users] FW: Seamless Multi Office Asterisk Deployment   Stephen Wingfield
• [asterisk-users] SER vs Asterisk?   Stephen Wingfield
• [asterisk-users] queue information in mySQL   Thomas Winter
• [asterisk-users] play file and action only stop if one defined key has been pressed   Thomas Winter
• [asterisk-users] play file and action only stop if one defined key has been pressed   Thomas Winter
• [asterisk-users] play file and action only stop if one defined key has been pressed   Thomas Winter
• [asterisk-users] SIP provider did not send BYE if callee is an queue   Thomas Winter
• [asterisk-users] res_musiconhold.c:1243 load_module: No music on hold classes configured   Thomas Winter
• [asterisk-users] Wanted: German to English translator for Asterisk documenation   Stefan Wintermeyer
• [asterisk-users] UK SIP Gateway   Wireless
• [asterisk-users] When to use Echo Cancellation cards?   Wireless
• [asterisk-users] Inbound PSTN CLID irratic with A200   Wireless
• [asterisk-users] DISCONNECT 41 hangup problem on PRI   Tom De Wispelaere
• [asterisk-users] TDM400P/FXS in a HP DL380 G5   Hans Witvliet
• [asterisk-users] Asterisk and T38 ?   Tobias Wolf
• [asterisk-users] accepting a call, macros, and key presses.   Jason Wolfe
• [asterisk-users] Test   Wai Wu
• [asterisk-users] PRI progress codes.   Wai Wu
• [asterisk-users] TC400B   Wai Wu
• [asterisk-users] How many gsm channels   Wai Wu
• [asterisk-users] TC400B   Wai Wu
• [asterisk-users] Coaching in asterisk   Wai Wu
• [asterisk-users] RE: Coaching in asterisk   Wai Wu
• [asterisk-users] RE: Coaching in asterisk   Wai Wu
• [asterisk-users] 1.4 compile issue   Wai Wu
• [asterisk-users] 1.4 compile issue   Wai Wu
• [asterisk-users] RE: Coaching in asterisk   Wai Wu
• [asterisk-users] disable client side hangup after dialing 911   Wai Wu
• [asterisk-users] RE: Coaching in asterisk   Wai Wu
• [asterisk-users] RE: Coaching in asterisk   Wai Wu
• [asterisk-users] Create meetme conference rooms on the flight.   Wai Wu
• [asterisk-users] One way dtmf tone on IAX   Wai Wu
• [asterisk-users] App_RXFax Problem.   John Wulter
• [asterisk-users] Need comparison between PBXtra, Trixbox, Thirdlane, Druid, Aheeva etc.   Zeeshan Zakaria
• [asterisk-users] Re: Help: CallerID Name not being sent on outbound PRI trunk   Zeeshan Zakaria
• [asterisk-users] Which VoIP router and switch to use for medium size business   Zeeshan Zakaria
• [asterisk-users] When to use Echo Cancellation cards?   Zeeshan Zakaria
• [asterisk-users] Which VoIP router and switch to use for medium size business   Zeeshan Zakaria
• [asterisk-users] Filter IDENT(Port 113) on Linksys router puts remote extensions to one way audio   Zeeshan Zakaria
• [asterisk-users] Re: Filter IDENT(Port 113) on Linksys router puts remote extensions to one way audio   Zeeshan Zakaria
• [asterisk-users] MP3Player   Dominik Zalewski
• [asterisk-users] Issue with Hamlet ISDN PCI card(Cologne Chipset)   Mauro Zanin
• [asterisk-users] vzaphfc installation...   Mauro Zanin
• [asterisk-users] vzaphfc installation?   Mauro Zanin
• [asterisk-users] Asterisk 1.2.15 chan_vpb with vpb-driver 4.0   Yifan Zhang
• [asterisk-users] zaptel 1.4 on fedora core 6 with dell pe 2850   Ma Zhiyong
• [asterisk-users] zaptel 1.4 on fedora core 6 with dell pe 2850   Ma Zhiyong
• [asterisk-users] about bluetooth channel   Iban Lopetegi Zinkunegi
• [asterisk-users] Call Back   Ivo Zivkov
• [asterisk-users] Call Back   Ivo Zivkov
• [asterisk-users] Call Back   Ivo Zivkov
• [asterisk-users] FAX using T38   Zoa
• [asterisk-users] Multiple simultaneous calls   Zoa
• [asterisk-users] IAX best practices   Zoa
• [asterisk-users] sip: failed the authenticate on INVITE   Michael Zoller
• [asterisk-users] sip: failed the authenticate on INVITE   Michael Zoller
• [asterisk-users] Limit call duration   Suity Zsolt
• [asterisk-users] Limit call duration   Suity Zsolt
• [asterisk-users] Limit call duration   Suity Zsolt
• [asterisk-users] Limit call duration   Suity Zsolt
• [asterisk-users] Counting callers   Suity Zsolt
• [asterisk-users] Counting callers   Suity Zsolt
• [asterisk-users] Asterisk and Fax   --[ UxBoD ]--
• [asterisk-users] Asterisk and Fax   --[ UxBoD ]--
• [asterisk-users] Asterisk 1.4.0 Installation error on Red Hat Linux 9.0-Urgent   --[ UxBoD ]--
• [asterisk-users] SNOM 360   --[ UxBoD ]--
• [asterisk-users] SNOM 360   --[ UxBoD ]--
• [asterisk-users] SNOM 360   --[ UxBoD ]--
• [asterisk-users] Question on Priorities   --[ UxBoD ]--
• [asterisk-users] Earliest dial tone, after boot up.   joe a.
• [asterisk-users] TDM400p, no CLI activity   joe a.
• [asterisk-users] polycom random reboots   joe a.
• [asterisk-users] Dell poweredge 860 acceptable forofficeenvironment ?   joe a.
• [asterisk-users] Earliest dial tone, after boot up.   joe acquisto
• [asterisk-users] TDM400p, no CLI activity   joe acquisto
• [asterisk-users] Unicall Brazil   andreunimed
• [asterisk-users] Tesco Internet Phone   asterisk
• [asterisk-users] Choppy sound with chan_capi + Fritz Card USB   asterisk-users at christophrothe.de
• [asterisk-users] Paid support offered   asterisk at 911networks.com
• [asterisk-users] blieve i my TE110P or My teleco provider ??   younss azzayani
• [asterisk-users] Problem with TE212P   younss azzayani
• [asterisk-users] Running Fax on E1 line   younss azzayani
• [asterisk-users] blieve i my TE110P or My teleco provider ??   younss azzayani
• [asterisk-users] Configurations Files of TE110P   younss azzayani
• [asterisk-users] Configurations Files of TE110P   younss azzayani
• [asterisk-users] Configurations Files of TE110P   younss azzayani
• [asterisk-users] Configurations Files of TE110P   younss azzayani
• [asterisk-users] Configurations Files of TE110P   younss azzayani
• [asterisk-users] Configurations Files of TE110P   younss azzayani
• [asterisk-users] Rx+,Rx-,Tx+,Tx- of TE110P   younss azzayani
• [asterisk-users] Rx+,Rx-,Tx+,Tx- of TE110P   younss azzayani
• [asterisk-users] French PRI channel - exact signaling used   younss azzayani
• [asterisk-users] Freepbx Incoming call's configuration   younss azzayani
• [asterisk-users] Freepbx Incoming call's configuration   younss azzayani
• [asterisk-users] SIP phone supporting more than 10 extension with a call transfer command   younss azzayani
• [asterisk-users] Dell poweredge 860 acceptable for office environment ?   younss azzayani
• [asterisk-users] SIP phone supporting more than 10 extension with a call transfer command   younss azzayani
• [asterisk-users] Dell poweredge 860 acceptable for office environment ?   younss azzayani
• [asterisk-users] Dell poweredge 860 acceptable for office environment ?   younss azzayani
• [asterisk-users] SMS Integration and SMS commands   younss azzayani
• [asterisk-users] make calls with different phone numbers   younss azzayani
• [asterisk-users] Configuring Faxs any help :)   younss azzayani
• [asterisk-users] Configuring Faxs any help :)   younss azzayani
• [asterisk-users] Configuring Faxs any help :)   younss azzayani
• [asterisk-users] Configuring Faxs any help :)   younss azzayani
• [asterisk-users] Configuring Faxs any help :)   younss azzayani
• [asterisk-users] ses ActiveDirectory and also Ldap and Kerberos.   younss azzayani
• [asterisk-users] how to check and set D-channel status   younss azzayani
• [asterisk-users] UK BT PRI   younss azzayani
• [asterisk-users] Hearing noise after 1min of calling   younss azzayani
• [asterisk-users] Tesco Internet Phone   bails
• [asterisk-users] Tesco Internet Phone   bails
• [asterisk-users] Tesco Internet Phone   bails
• [asterisk-users] SNOM 360   bails
• [asterisk-users] Re: Asterisknow with video and X-Lite not quite working   dave cantera
• [asterisk-users] Re: Asterisknow with video and X-Lite not quite working   dave cantera
• [asterisk-users] Linksys not Ringing   dave cantera
• [asterisk-users] Single sign on PC + phone?   dave cantera
• [asterisk-users] proposal: a new mailing list for asterisk 1.4, why not?   dave cantera
• [asterisk-users] T1 cable for Digium T1/E1 Cards   dave cantera
• [asterisk-users] Fedora + Linux Kernel 2.6 for Zaptel/Asterisk Installation   dave cantera
• [asterisk-users] ses ActiveDirectory and also Ldap and Kerberos.   dave cantera
• [asterisk-users] Asterisk 1.4.2   dave cantera
• [asterisk-users] Linksys/Sipura SPA-942 phones in larger deployments   dave cantera
• [asterisk-users] Linksys/Sipura SPA-942 phones in largerdeployments   dave cantera
• [asterisk-users] Linksys/Sipura SPA-942 phones in larger deployments   dave cantera
• [asterisk-users] Asterisk 1.4.2   dave cantera
• [asterisk-users] the age old telephone tree... why re-invent the wheel?   dave cantera
• [asterisk-users] 1.4 - IAX2 - No registration for peer   dave cantera
• [asterisk-users] Multi-registration ?   dave cantera
• [asterisk-users] Polycom 601 loop   dave cantera
• [asterisk-users] Re: Polycom 601 loop   dave cantera
• [asterisk-users] Re: Polycom 601 loop   dave cantera
• [asterisk-users] OT Vonage V-Phone Adapter (Possible Hack)   cb
• [asterisk-users] voip-info.org status update   cb
• [asterisk-users] SRTP testers needed   marek cervenka
• [asterisk-users] SRTP testers needed   marek cervenka
• [asterisk-users] Failure acknowledgement time   cimsi
• [asterisk-users] Failure acknowledgement time   cimsi
• [asterisk-users] Problem configuring voice conference   John covici
• [asterisk-users] Problems with Voice conferencing   John covici
• [asterisk-users] Re: Problems with Voice conferencing   John covici
• [asterisk-users] Odd MeetMe bahaviour with MoH ...   John covici
• [asterisk-users] Patton 1400   Jean-Louis curty
• [asterisk-users] AsteriskNow Beta 4 - zaptel.conf / zapata.conf problems   demuel at thephinix.org
• [asterisk-users] error, install freePbx   dima
• [asterisk-users] Configuring Faxs any help :)   dima
• [asterisk-users] Configuring Faxs any help :)   dima
• [asterisk-users] SRTP vs ZRTP in Asterisk   dubrowin.44628913 at bloglines.com
• [asterisk-users] Cost of Branded Equipment for Voip Provider Implementation   ecleofe7 at yahoo.com
• [asterisk-users] outdial to phone for new VM notification   end1r
• [asterisk-users] SIP/IAX peers UNREACHABLE and audio loss   f6hqz-m at hamwlan.net
• [asterisk-users] Issue with Hamlet ISDN PCI card(Cologne Chipset)   f6hqz-m at hamwlan.net
• [asterisk-users] Re: RE : SIP/IAX peers UNREACHABLE and audio loss   f6hqz-m at hamwlan.net
• [asterisk-users] Two or More Bri Cards   f6hqz-m at hamwlan.net
• [asterisk-users] Asterisk 1.4 and chan_misdn   f6hqz-m at hamwlan.net
• [asterisk-users] wireless desktop phones   f6hqz-m at hamwlan.net
• [asterisk-users] wireless desktop phones   f6hqz-m at hamwlan.net
• [asterisk-users] Re: asterisk-users Digest, Vol 32, Issue 128   fb at lucyde.com
• [asterisk-users] Re: asterisk-users Digest, Vol 32, Issue 129   fb at lucyde.com
• [asterisk-users] Can I generate random SIP traffic?   gabriele.pinzauti at witech.it
• [asterisk-users] Asterisk distributed deployment   ggonzalez at telviso.com.ar
• [asterisk-users] Re: Asterisk distributed deployment   ggonzalez at telviso.com.ar
• [asterisk-users] Which IP Phones have buttons can be assigned to functions with Asterisk   bilal ghayyad
• [asterisk-users] How to resolve CallerID from AudioCodes FXO   hind habaoui
• [asterisk-users] FW: Microsoft buys Tellme   wendell hamilton
• [asterisk-users] FW: Microsoft buys Tellme   wendell hamilton
• [asterisk-users] Shoretel integration into Salesforce.com   wendell hamilton
• [asterisk-users] T1 cable for Digium T1/E1 Cards   jacobso1
• [asterisk-users] Set(CALLERID(all) not working with 'unknown' call?   jan.sarin at securia.se
• SV: [asterisk-users] Set(CALLERID(all) not working with 'unknown'call?   jan.sarin at securia.se
• [asterisk-users] Asterisk incoming caller id problem   johnny_xing
• [asterisk-users] shutdown   josanchez at unicauca.edu.co
• [asterisk-users] cmd pickup Problem   jroesch at bsradio.de
• [asterisk-users] SIP REFER   kalle.odenthal at genion.de
• [asterisk-users] SIP RTP Tunnel   kalle.odenthal at genion.de
• [asterisk-users] SIP RTP Tunnel   kalle.odenthal at genion.de
• [asterisk-users] SIP RTP Tunnel   kalle.odenthal at genion.de
• [asterisk-users] SIP REFER ans Dialplan   kalle at esprITschmiede.de
• [asterisk-users] Interconnexion d'un serveur Asterisk à des PABX LG ( IP LDK)   khawla khawla
• [asterisk-users] FW: Seamless Multi Office Asterisk Deployment   kjcsb
• [asterisk-users] Asterisk hangs up SIP call after 6 200 retransmits   kjcsb
• [asterisk-users] Asterisk hangs up SIP call after 6 200 retransmits   kjcsb
• [asterisk-users] Call center manager for Asterisk (Release 0.3)   lenz
• [asterisk-users] asterisk log analyzer   lenz
• [asterisk-users] x100p.com   Mail list
• OT: Re: [asterisk-users] voip-info.org status update   Mail list
• [asterisk-users] Need feedback on vitelity   Mail list
• [asterisk-users] A request for your input.   lmth at deakin.edu.au
• [asterisk-users] Newbie Question   mail-lists
• [asterisk-users] Re: Problem converting a Cisco 7960 to SIP   mail-lists
• [asterisk-users] Can't hear any sound (This time in plain text)   map
• [asterisk-users] Complicated callback solution   marcelobiz at comcast.net
• [asterisk-users] Complicated callback solution   marcelobiz at comcast.net
• [asterisk-users] Asterisk and Databases   marcelobiz at comcast.net
• [asterisk-users] Doorphone vs. Grandstream BT101   marcotasto
• [asterisk-users] AsteriskNow Beta 4 - zaptel.conf / zapata.conf problems   mark.d.rowe at bt.com
• [asterisk-users] Linksys/Sipura SPA-942 phones in larger deployments   andrew matthews
• [asterisk-users] zaptel 1.4 on fedora core 6 with dell pe 2850   mazhiyong at sohu.com
• [asterisk-users] While the VoIP-Info.org site is down...   mitcheloc
• [asterisk-users] voip-info.org is back!   mitcheloc
• [asterisk-users] Jajah.com like script?   mitcheloc
• [asterisk-users] Microsoft launches first PABX   mitcheloc
• [asterisk-users] Asterisk Automated Outbound Messaging   mitcheloc
• [asterisk-users] Issues with a Linksys SPA 2102 and asterisk   nigma nigmus
• [asterisk-users] Re: queue information into db   nik600
• [asterisk-users] Re: queue information into db   nik600
• [asterisk-users] Re: queue information into db   nik600
• [asterisk-users] asterisk and ssl   nik600
• [asterisk-users] Re: queue information into db   nik600
• [asterisk-users] Re: queue information into db   nik600
• [asterisk-users] Re: queue information into db   nik600
• [asterisk-users] Asterisk and Databases   nik600
• [asterisk-users] Call center manager for Asterisk (Release 0.3)   nik600
• [asterisk-users] Re: Call center manager for Asterisk (Release 0.3)   nik600
• [asterisk-users] Call center manager for Asterisk (Release 0.3)   nik600
• [asterisk-users] Call center manager for Asterisk (Release 0.3)   nik600
• [asterisk-users] Call center manager for Asterisk (Release 0.3)   nik600
• [asterisk-users] Unicall/R2 for Asterisk 1.4 Available for TESTING   nivlekch
• [asterisk-users] Re: zapata with Tiger3XX compilation error   pedro noticioso
• [asterisk-users] Tesco Internet Phone   phil.dawson at marnock.com
• [asterisk-users] Too Many Open Files, Hung SIP Sessions, Can I Increase File Count?   phil.dawson at marnock.com
• [asterisk-users] hardware spec   phil.dawson at marnock.com
• [asterisk-users] switches   phil.dawson at marnock.com
• [asterisk-users] bugetone 200's   phil.dawson at marnock.com
• [asterisk-users] A New Phone Service - www.virtualphoneline.com   rehan at supertec.com
• [asterisk-users] Off Topic: Open Source USB Softphone   laurent schweizer
• [asterisk-users] Queue Announcements for Operators   scott
• [asterisk-users] Need comparison between PBXtra, Trixbox, Thirdlane, Druid, Aheeva etc.   shadowym
• [asterisk-users] Re: Asterisk distributed deployment   shadowym
• [asterisk-users] Zap Channel Deadlocks   shadowym
• [asterisk-users] When to use Echo Cancellation cards?   shadowym
• [asterisk-users] asterisk on mini-itx   shadowym
• [asterisk-users] Rebooting all Aastra phones   shadowym
• [asterisk-users] voip-info.org status update   shadowym
• [asterisk-users] voip-info.org status update   shadowym
• [asterisk-users] voip-info.org status update   shadowym
• [asterisk-users] While the VoIP-Info.org site is down...   shadowym
• [asterisk-users] voip-info.org status update   shadowym
• [asterisk-users] voip-info.org status update   shadowym
• [asterisk-users] Microsoft launches first PABX   shadowym
• [asterisk-users] Need comparison between PBXtra, Trixbox, Thirdlane, Druid, Aheeva etc.   shadowym
• [asterisk-users] Microsoft launches first PABX   shadowym
• [asterisk-users] Microsoft launches first PABX   shadowym
• [asterisk-users] Re: asterisk-users Digest, Vol 32, Issue 106   shadowym
• [asterisk-users] wireless desktop phones   shadowym
• [asterisk-users] Looking for a terminations provider (carrier grade)   sip
• [asterisk-users] Looking for a terminations provider (carrier grade)   sip
• [asterisk-users] Queue - Call delivery problems - Asterisk 1.4 - autofill=yes   equis software
• [asterisk-users] Queue App - Free agent and waiting calls   equis software
• [asterisk-users] Queue App - Free agent and waiting calls   equis software
• [asterisk-users] Queue App - Free agent and waiting calls   equis software
• [asterisk-users] Multiple simultaneous calls   stefano.totaro at transport.alstom.com
• [asterisk-users] Multiple simultaneous calls   stefano.totaro at transport.alstom.com
• Rif: Re: [asterisk-users] Multiple simultaneous calls   stefano.totaro at transport.alstom.com
• [asterisk-users] No Audio when integrating openSER and Asterisk , in NAT   raviprakash sunkara
• [asterisk-users] No Audio when integrating with openSER and Asterisk in the SAME LAN ,   raviprakash sunkara
• [asterisk-users] Need help to strip variable   van Leen, Phil
• [asterisk-users] Asterisk 1.4.1 Released   voiplist
• [asterisk-users] SIP remote crash bug   voiplist
• [asterisk-users] asterisk on mini-itx   voiplist
• [asterisk-users] Is there any variable for Voicemail Password in Asterisk   jamshed zaidi
• [asterisk-users] How to access Voicemail Password in Asterisk without using VoicemailMain????????   jamshed zaidi
• [asterisk-users] How to access Voicemail Password in Asteriskwithout using V   jamshed zaidi
• [asterisk-users] How to access Voicemail Passwordin Asteriskwithout using V   jamshed zaidi