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
• [as