2,471 messages

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

2007 March [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

[asterisk-users] [Announce] Web-MeetMe V3.0.1 released - Dan Austin
[asterisk-users] Doorphone vs. Grandstream BT101 - Steve Totaro
[asterisk-users] HUD Lite server on Debian - Dovid B
[asterisk-users] TDM analog cards, volume, echo, fxotune, ztmonitor and HPEC - Tzafrir Cohen
[asterisk-users] the age old telephone tree... why re-invent the wheel? - Steve Totaro
[asterisk-users] Problem with ztdummy - Alan Chandler
[asterisk-users] Fedora + Linux Kernel 2.6 for Zaptel/AsteriskInstallation - George C. Attopany
[asterisk-users] FW: Seamless Multi Office Asterisk Deployment - kjcsb
[asterisk-users] how to check and set D-channel status - Farooq Ahmed
[asterisk-users] How to resolve CallerID from AudioCodes FXO - Angel Heart
[asterisk-users] Failure acknowledgement time - cimsi
[asterisk-users] cutting hash in dial app - René Enskat
[asterisk-users] how to check and set D-channel status - younss azzayani
[asterisk-users] Moving from Bristuff to mISDN - Giorgio Incantalupo
[asterisk-users] Re: Two or More Bri Cards - Edoardo Serra
[asterisk-users] Re: Two or More Bri Cards - Stelios Koroneos
Fwd: [asterisk-users] Multi-registration ? - Drew Gibson
[asterisk-users] Asterisk incoming caller id problem - Gordon Henderson
[asterisk-users] Emergency chan_sip issue - Chris Bagnall
[asterisk-users] SRTP vs ZRTP in Asterisk - Michael Graves
[asterisk-users] Re: Polycom 601 loop - Nathan Bell
[asterisk-users] SIP REFER - kall...@genion.de
[asterisk-users] how to define a pilot number - Lito Lampitoc
[asterisk-users] Limit call duration - Rizwan Hisham
[asterisk-users] Re: how to define a pilot number - Lito Lampitoc
[asterisk-users] cisco 7905 - Shaikh Jallaluddin
[asterisk-users] Doorphone - Ray Wadkins
[asterisk-users] TDM400p reliability - Joe Acquisto
[asterisk-users] Re: SIP/IAX peers UNREACHABLE and audio loss - Edoardo Serra
[asterisk-users] TDM400p reliability - Ira
[asterisk-users] TDM400p reliability - Chris Bagnall
[asterisk-users] Re: asterisk-users Digest, Vol 32, Issue 106 - shadowym
[asterisk-users] TDM400p reliability - Dave Fullerton
[asterisk-users] Park & No Announce? - Matt
[asterisk-users] Inbound Voice Quality - Speed Change - Lacy Moore - Aspendora
[asterisk-users] Re: Inbound Voice Quality - Speed Change - Travis Schafer
[asterisk-users] ztdummy and MOH - Lacy Moore - Aspendora
[asterisk-users] ztdummy and MOH - Klaverstyn, David C
[asterisk-users] Counting callers - Suity Zsolt
[asterisk-users] UK BT PRI - younss azzayani
[asterisk-users] Re: How is this feature called ? - Tomislav Parcina
[asterisk-users] wireless desktop phones - Jordan Novak
[asterisk-users] Cisco 30VIP Phone - Derek Whitten
[asterisk-users] PoE - IEEE 802.3af - Mike
[asterisk-users] PoE - IEEE 802.3af - Bruce Reeves
[asterisk-users] PoE - IEEE 802.3af - Dave Fullerton
[asterisk-users] Transfering not working - how to debug? - Alan Chandler
[asterisk-users] Multi-line phones - Asterisk uses wrong callerid - Matt
[asterisk-users] Polycom and Asterisk - Darryl Dunkin
RES: [asterisk-users] Development of new features in Asterisk Manager - Steve Murphy
[asterisk-users] Polycom and Asterisk - Mike Hammett
[asterisk-users] Dialplan Streaming - Doug Garstang
[asterisk-users] Dialplan Streaming - Doug Garstang
[asterisk-users] Polycom SoundPoint 501 - Darryl Dunkin
[asterisk-users] Polycom SoundPoint 501 - Paolo Supino
[asterisk-users] Voice mail - Khayelihle Mbona
[asterisk-users] Voice mail - Tzafrir Cohen
[asterisk-users] wireless desktop phones - Steve Langstaff
[asterisk-users] sip: failed the authenticate on INVITE - Giorgio Incantalupo
[asterisk-users] "Couldn't load variables.txt?aldope=xxxxx " - Carlos Jerónimo
[asterisk-users] error in FreePBX - Remco Barendse
[asterisk-users] Re: Re: Inbound Voice Quality - Speed Change - Jim Duda
[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] Asterisk does not reINVITE after 302Redirect & 401Unauthorized - Mush...@3com.com
[asterisk-users] Scratchy Audio with Asterisk 1.2.4 over IAX on FreeBSD? - Benoit Panizzon
[asterisk-users] Multi-line phones - Asterisk uses wrong callerid - Drew Gibson
[asterisk-users] help - UNSUBSCRIBE - Jerric
[asterisk-users] UK PRI and outgoing CLI FYI - Steve Kennedy
RES: RES: RES: [asterisk-users] Development of new featuresin AsteriskManager - Richard Lyman
[asterisk-users] Asterisk hangs up SIP call after 6 200 retransmits - Raj Jain
[asterisk-users] Re: Problem converting a Cisco 7960 to SIP - Ron Wellsted
[asterisk-users] Off Topic: Open Source USB Softphone - Gordon Henderson
[asterisk-users] UK PRI and outgoing CLI FYI - Julian Lyndon-Smith
[asterisk-users] error in FreePBX - Francesco Peeters (Asterisk)
[asterisk-users] CallerID + Name - Trevor Peirce
[asterisk-users] Re: Problem converting a Cisco 7960 to SIP - Salvatore Giudice
[asterisk-users] Interconnexion d'un serveur Asterisk à des PABX LG ( IP LDK - Yuan LIU
[asterisk-users] Asterisk 1.4 with Digium B410P - Timing problem - Marco Mouta
[asterisk-users] web based sip phone - Tim Panton
[asterisk-users] web based sip phone - Pezhman Lali
[asterisk-users] FAX & mISDN - LKS GMAIL
[asterisk-users] FAX & mISDN - LKS GMAIL
[asterisk-users] Linksys SPA 3102 causing me problems - Alan Chandler
Fwd: [asterisk-users] Multi-registration ? - Yuan LIU
[asterisk-users] web based sip phone - Yuan LIU
[asterisk-users] cutting hash in dial app - Yuan LIU
[asterisk-users] Which IP Phones have buttons can be assigned to functions with Asterisk - bilal ghayyad
[asterisk-users] xten web phone - Pezhman Lali
[asterisk-users] SNOM 360 - --[ UxBoD ]--
[asterisk-users] unconditionally redirecting incoming calls by 302 Moved Temporarily messages doing right accounting - Ricardo Carvalho
[asterisk-users] forwarding loop not detected - Bill Gibbs
[asterisk-users] SNOM 360 - --[ UxBoD ]--
[asterisk-users] Lucent TNT - ring timer - Brent
[asterisk-users] Redirect failed, channel not up. - Nathan Bell
[asterisk-users] switchtype and signalling query - Doug Lytle
[asterisk-users] Indicating agent status on the phone - Philipp Kempgen
[asterisk-users] Sponsored development - Monodirectional audio handling - Edoardo Serra
[asterisk-users] Re: Sponsored development - Monodirectional audio handling - Edoardo Serra
[asterisk-users] Setting a call to be recorded before Xfer? - Dean Collins

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