4,516 messages

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

2005 July [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 43 44 45 46

[Asterisk-Users] [ANNOUNCE] chan_capi-cm-0.5.4 release - Armin Schindler
[Asterisk-Users] DID + 800 Providers - Nathan E. Pralle
[Asterisk-Users] DID + 800 Providers - BSUM...@aol.com
[Asterisk-Users] DID + 800 Providers - BSUM...@aol.com
[Asterisk-Users] super high bandwidth codec - Dean Collins
[Asterisk-Users] Polycom 600 Ring-Answer (but not ring!) - Billy Dunn
[Asterisk-Users] Zap PRI load testing - Niklas Larsson
[Asterisk-Users] Zap PRI load testing - Adam Goryachev
[Asterisk-Users] VoiceMailMain issue.. - Mauro Zanin
[Asterisk-Users] Uk Caller id - Giorgio Incantalupo
[Asterisk-Users] Voicemail : Unable to create lock file: No such file or directory - Alessio Focardi
[Asterisk-Users] network problem -- echo - Rudolf Ladyzhenskii
[Asterisk-Users] super high bandwidth codec - Eric Wieling aka ManxPower
[Asterisk-Users] Need Advice - Nathan Pralle
[Asterisk-Users] Zap channel configuration problem - Alexis F.
[Asterisk-Users] Re: Polycom IP600 - Flashing clock and date? - Noah Miller
[Asterisk-Users] Need Advice - Chris Thompson
[Asterisk-Users] Re: Marco and Realtime Extension Problem [SOLVED] - Kenige Ho
[Asterisk-Users] Mixed Voice/Data T1 - Chris Mason (Lists)
[Asterisk-Users] Hangups transferring call from Intertel system - cra...@pacelink.com
[Asterisk-Users] Asterisk Configuration - Madhawa Jayanath
[Asterisk-Users] re: realtime caller id extensions matching - Matthew Simpson
[Asterisk-Users] Re: Marco and Realtime Extension Problem [SOLVED] - Tzafrir Cohen
[Asterisk-Users] Should this work? - Kristof Hardy
[Asterisk-Users] A TDM issue.. - chouck
[Asterisk-Users] Call forwarding - Adrian A
[Asterisk-Users] chan_sccp release 20050725 - Sergio Chersovani
[Asterisk-Users] Should this work? - Matt Riddell
[Asterisk-Users] Phone Recommendations - Dan Adams
[Asterisk-Users] 100% CPU with Unicall and * head - Denis Galvão - iSolve
[Asterisk-Users] Some more VOICEMAILMAIN issue... - Mauro Zanin
[Asterisk-Users] best way to dial and connect two users - tom fielding
[Asterisk-Users] Re: why zap call transfer fails? - Michael Jia
[Asterisk-Users] include not working in bristuffed Asterisk 1.0.7 extensions.conf - Giorgio Incantalupo
[Asterisk-Users] Some more VOICEMAILMAIN issue... - Ronald_Wiplinger
[Asterisk-Users] how to config E400P card? - dev2...@126.com
[Asterisk-Users] ABI manager - redirect - Christoph Eicke
[Asterisk-Users] Eyebeam Video+Nat - Guillaume
[Asterisk-Users] How can I use MySQL in the dialplan? - Matthew Boehm
[Asterisk-Users] 7960 SIP Firmware Upgrade Strange Problem - Watkins, Bradley
[Asterisk-Dev] RE: [Asterisk-Users] Zaptel update, Asterisk 1.2 janitor projects - Tzafrir Cohen
[Asterisk-Users] 7960 SIP Firmware Upgrade Strange Problem - Walid Azab
[Asterisk-Users] [Asterisk-Dev] Asterisk 1.2 Release Plans - Kevin P. Fleming
[Asterisk-Users] 7960 SIP Firmware Upgrade Strange Problem - Walid Azab
[Asterisk-Users] Call Quality Reporting - Nathan Pralle
[Asterisk-Users] 100% CPU with Unicall and * head - Denis Galvão - iSolve
[Asterisk-Users] 7960 from SIP to SKINNY - Joseph
[Asterisk-Users] problems with compiling asterisk-oh323-0.6.5 - jonny hashem
[Asterisk-Users] Re: [Asterisk-Dev] Cluecon - Who's going ? - Greg Boehnlein
[Asterisk-Users] Some more VOICEMAILMAIN issue... - Tim Litwiller
[Asterisk-Users] What does pbx-wilcalu.so do and why does it keep crashing my * box? - Mark Phillips
[Asterisk-Users] mpg123 - two processes - Brian West
[Asterisk-Users] ASTCC: different incriments - Rusty Shackleford
[Asterisk-Users] mpg123 - two processes - El Flynn
[Asterisk-Users] super high bandwidth codec - Andrew C. Brown
[Asterisk-Users] super high bandwidth codec - Tzafrir Cohen
[Asterisk-Users] TE110P Cable Pin Out - Miloš Kocbek
[Asterisk-Users] Call Monitoring - Ian Bert Tusil
[Asterisk-Users] Why is sip saying NO NAT - Chris Mason (Lists)
[Asterisk-Users] IAX over HTTP - James Cloos
[Asterisk-Users] Zaptel error: Unable to create channel op type 'Zap' - Peter Raaijmakers
[Asterisk-Users] Zaptel error: Unable to create channel op type 'Zap' - Altus Snyman
[Asterisk-Users] Port Restricted CONE NAT Error - Afzaal Mirza
R: [Asterisk-Users] Sound Quality Problems - Yousef Herzallah
[Asterisk-Users] Voice mailbox on the fly? - Tzafrir Cohen
R: [Asterisk-Users] Zaptel error: Unable to create channel op type'Zap' - Damon Brown
[Asterisk-Users] Music on Hold: CPU Intensive Monster - Matthew Boehm
[Asterisk-Users] spandsp - dorn hetzel
[Asterisk-Users] cdr_mysql does not write to mysql db - Carlos Chavez
[Asterisk-Users] Random Behavior on Trunk Lines with TDM Card - Heath Bowlin
R: [Asterisk-Users] Zaptel error: Unable to create channel op type'Zap' - Derek Whitten
[Asterisk-Users] automated response - Karthikeyan
[Asterisk-Users] Latest CVS HEAD and the new wct4xxp card - Hall, Eric M.
[Asterisk-Users] Read Back Caller ID Using Number Announcement in Digital Receptionist - Peter Bowyer
[Asterisk-Users] Call Monitoring - Dan Littlejohn
[Asterisk-Users] RE: Asterisk fax problems with SPANDSP - Allan Mak
[Asterisk-Users] announce to caller in queues (asterisk for art!) - Hans-Christoph Steiner
[Asterisk-Users] sip+oh323 - no voice at sip side - bar...@datacomsa.pl
[Asterisk-Users] Strange PRI lockup - Johann
[Asterisk-Users] Suggested System Specs - 20 ext, 8 Incoming Lines - Doug Logan
[Asterisk-Users] dialplan defenition - David Koski
[Asterisk-Users] A problem with queues - Jorge Alayon
[Asterisk-Users] SIP WEB Phone (Wanna implement Click to Call) - Guillermo Salas M
[Asterisk-Users] Can you caculate with me? - Geoff Manning
[Asterisk-Users] strange dial problem with polycom 501 - Tom Hayden
[Asterisk-Users] Querying Nagios users... - Jeremy Melanson
[Asterisk-Users] Broadvoice- 404 not Found - Bernie Courtney
[Asterisk-Users] Sound Cards, ALSA, and Asterisk - Robert Christian
[Asterisk-Users] most stable linux to build business - snacktime
[Asterisk-Users] grandstream budget tone 100 ip-phone just one call - David Romero
[Asterisk-Users] help Windows messenger configuaration - someshwarak
[Asterisk-Users] SIP phone procedural question - Angus Comber
[Asterisk-Users] new TDM04B - Jerry Geis
[Asterisk-Users] X100P/Caller ID: clidtest works, * complains - Jon Whitear
[Asterisk-Users] OT Skype almost being sold - Dean Collins
[Asterisk-Users] Using AGI, how do you clear a variable? - James Moore
[Asterisk-Users] Sound Card help - Bashir Ullah
[Asterisk-Users] ASTCC: different incriments - Ade Agbero
[Asterisk-Users] Questions on Asterisk and CallerID - Ganbold Tsagaankhuu
[Asterisk-Users] Voicemail envelope time is 4 hours ahead(?) - Frank Tarczynski

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