4,160 messages

com.digium.lists.asterisk-users [All Lists]

2006 June [All Months]

Page 1 (Messages 1 to 100): 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27 28 29 30 31 32 33 34 35 36 37 38 39 40 41 42

[Asterisk-Users] Meetme max users - Matt Florell
[Asterisk-Users] best hardphone for Asterisk? - Doug Crompton
[Asterisk-Users] Asterisk Startups - Douglas Garstang
[Asterisk-Users] News: Asterisk VOIP Jobs Site - Revision 3.0 up! - Douglas Garstang
[Asterisk-Users] Zaptel answering the Line - Tzafrir Cohen
[Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!! - Marco Mouta
[Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!! - Josué Conti
[Asterisk-Users] SE Michigan asterisk users group - Tom Hayden
[Asterisk-Users] Asterisk-1.2.9.1 with Siemens HiPath 4000 - richard Coco
[Asterisk-Users] 1.2.9.1 SIP/Local/Queue behaviours weird - C F
[Asterisk-Users] ASTCC: How to reset periodically all "card in use" flag back? - Ronald Wiplinger
[Asterisk-Users] Re: Voice calls sent to fax extension - Steven
[Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!! - Mike Fedyk
[Asterisk-Users] STUN? - Martin Joseph
[Asterisk-Users] asterisk-stat display problems - Mojo with Horan & Company, LLC
[Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!! - Marco Mouta
[Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!! - Brian Capouch
[Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!! - Francesco Peeters
[Asterisk-Users] "Say" Applications fail - Jon Mosier
[Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!! - Jean-Michel Hiver
[Asterisk-Users] 1.2.9.1 SIP/Local/Queue behaviours weird - Isaac Xiao
[Asterisk-Users] SE Michigan asterisk users group - Carlos Alperin
[Asterisk-Users] ASTCC: How to reset periodically all "card in use" flag back? - JP Carballo
[Asterisk-Users] Oh oh. Micro$oft just noticed VoIP - Francesco Peeters (Asterisk)
[Asterisk-Users] Error in config sample for GoToIf? - Brian Capouch
SV: [Asterisk-Users] Error in config sample for GoToIf? - Brian Capouch
[Asterisk-Users] Asterisk-1.2.9.1 with Siemens HiPath 4000 - richard Coco
[Asterisk-Users] x100p buying advice - Gareth Blades
[Asterisk-Users] siemens pbx and asterisk - richard Coco
[Asterisk-Users] Meetme max users - Kai Ober
[Asterisk-Users] Re: Asterisk x Siemens HiPath 4000 - Josué Conti
[Asterisk-Users] asterisk to mobile phone - Roger Workman
[Asterisk-Users] Call length limitation - Andrew Nowrot
[Asterisk-Users] Callstatus on bridge IAX2 <-> ZAPTEL is always "answer" even if the call fails - John Novack
[Asterisk-Users] isdn-data over iax - DR...@b-w-computer.de
[Asterisk-Users] Re: Asterisk x Siemens HiPath 4000 - Herchi Silviu
[Asterisk-Users] 7960 help: transferring calls - Lacy Moore - Aspendora
[Asterisk-Users] Call length limitation - Andrew Nowrot
[Asterisk-Users] Call length limitation - William Piper
[Asterisk-Users] RE: Asterisk-Users Digest, Vol 23, Issue 182 - Dan Elder
[Asterisk-Users] PRI - Ring requested on channel errors - inbound & outbound stop working. - Tim C. Lewis
[Asterisk-Users] Re: Asterisk x Siemens HiPath 4000 - Mike Lynchfield
[Asterisk-Users] Re: trunk rollover - Steven
[Asterisk-Users] Wierd bug with MD3200 - Luciano Moreira
[Asterisk-Users] Mail loop? - Mike Fedyk
[Asterisk-Users] Mail loop? - Curt Shaffer
[Asterisk-Users] Mail loop? - Steven Ringwald
[Asterisk-Users] transferring calls from ekiga to asterisk - don Paolo Benvenuto
[Asterisk-Users] Most stable Asterisk version - shadowym
[Asterisk-Users] Meetme + Sangoma issue? - Matt Florell
[Asterisk-Users] dial if - Miles Scruggs
[Asterisk-Users] Addon-ooh323 install problem - Tetsuya Yamamoto
[Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!! - olivier.taylor
[Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!! - trixter aka Bret McDanel
[Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!! - trixter aka Bret McDanel
[Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!! - Marco Mouta
[Asterisk-Users] Trixbox maunual configuration - Jordan Novak
[Asterisk-Users] Realtime: how to use column setvar? - Ronald Wiplinger
[Asterisk-Users] point to point T hookup? - Sean Cook
[Asterisk-Users] point to point T hookup? - Sean Cook
[Asterisk-Users] Re: Changing standard Voicemail behavior - Leah Newmark
[Asterisk-Users] Mysql Trixbox - Wasif
[Asterisk-Users] asterisk -> my cell phone's voicemail sound problems - Cory Forsyth
[Asterisk-Users] Asterisk 1.2.7.1 with Polycom 501 on SIP -> ConfCalling - Jerry Jones
[Asterisk-Users] Echo Cancellation - Douglas Garstang
[Asterisk-Users] Suggested Phone - Daniel Salama
[Asterisk-Users] Suggested Phone - Cory Andrews
[Asterisk-Users] asterisk shutdown - William Piper
[Asterisk-Users] WebPhone - Erick Perez
[Asterisk-Users] Help with incoming SIP routing - Christopher Aloi
[Asterisk-Users] IAX2 Destroying channel to avoid deadlock - Jordan Novak
[Asterisk-Users] SNOM Softphone on windows 2000 - Stefan-Michael. Guenther (in-put GbR)
[Asterisk-Users] Call length limitation - Andrew Nowrot
[Asterisk-Users] app_sms not working anymore - stoffell
[Asterisk-Users] Realtime SIP Registrations - Aaron Daniel
[Asterisk-Users] Cisco 7905G SIP firmware needed - Andrea Frigo
[Asterisk-Users] Cisco 7905G SIP firmware needed - Ryan Amos
[Asterisk-Users] Ztdummy and Debian on Intel Macmini - Martin Joseph
[Asterisk-Users] Avaya 4610sw SIP setup problem - Henk
[Asterisk-Users] No Sounds - Bernhard Janetzki
[Asterisk-Users] Sangoma A104D is dropping DTMF digits during IVR - Jay Dutt
[Asterisk-Users] need help troubleshooting clipping and garbled VOIP calls - Rich Adamson
[Asterisk-Users] Sangoma A104D is dropping DTMF digits, during IVR - Andre Courchesne - Consultant
[Asterisk-Users] Queue errors when phones are down, and possible solution - Franklin Webb
[Asterisk-Users] ISDN (E1) Hardware Echo Cancellation - Avi Miller
[Asterisk-Users] dlink wifi dph-540 and text messaging - Jerry Geis
[Asterisk-Users] Very bad quality with AVMFritz!cardPCIandchan_capi - James Harper
[Asterisk-Users] Digium Hardware Reliability - Philippe Lindheimer
[Asterisk-Users] voting,suggestiuon,your input needed to all - Mike Lynchfield
[Asterisk-Users] need help troubleshooting clipping and garbled VOIP calls - Thomas Kenyon
[Asterisk-Users] OH323 issue on AT320 Phones - aste...@frameweb.it
[Asterisk-Users] IAX jitter / clocking problem - Doug Lytle
[Asterisk-Users] Integrate asterisk with Database - Vidura Senadeera
Re: [Asterisk-Users] ISDN: 3° incoming call - Armin Schindler
[Asterisk-Users] Asterisk -x option in 1.2.9.1 - Douglas Garstang
[Asterisk-Users] asterisk to mobile phone - Woodoo People .pGa!
[Asterisk-Users] IAX jitter / clocking problem - Pavel Jezek
[Asterisk-Users] Auto answer an IAXY how - Alexander Lopez
[Asterisk-Users] SIP qualify time - best practices? - Bryan Field-Elliot
[Asterisk-Users] SOLVED: IAX jitter / clocking problem - Pavel Jezek

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