October 2003 Archives by thread
      
      Starting: Wed Oct  1 01:36:51 MST 2003
         Ending: Fri Oct 31 12:11:31 MST 2003
         Messages: 338
     
- [Asterisk-Dev] non-root asterisk install
 
duncan
- [Asterisk-Dev] chan_alsa broken ?
 
Conroy, Lawrence (SMTP)
- [Asterisk-Dev] non-root asterisk install
 
TC
- [Asterisk-Dev] RE: Asterisk license (fwd)
 
costas 
- [Asterisk-Dev] GnoPhone CVS compile error
 
Peter Nixon
- [Asterisk-Dev] Gastman patch to make it compile on SuSE 8.2
 
Peter Nixon
- [Asterisk-Dev] oss Errors
 
James Coberly
- [Asterisk-Dev] SIP transfer.
 
Micke Andersson
- [Asterisk-Dev] Asterisk and RADIUS
 
Peter Nixon
- [Asterisk-Dev] Voice detection
 
Brad Waite
- [Asterisk-Dev] Cisco 7960 w/DTMF Patch
 
Robert
- [Asterisk-Dev] double dtmfs
 
duncan
- [Asterisk-Dev] Problem with sip show channels
 
Alex Lopez
- [Asterisk-Dev] Job Opening at Digium
 
Mark Spencer
- [Asterisk-Dev] FreeBSD
 
Magnus Benngard
- [Asterisk-Dev] Asterisk for 2.6.0 series kernels?
 
Brian Capouch
- [Asterisk-Dev] DEBUG[1116949808]: File channel.c, Line 380 (ast_queue_frame):
 Dropping voice to exceptionally long queue on IAX2[iaxtel@iaxtel]/7
 
Eric Wieling
- [Asterisk-Dev] Speech Coding Information
 
Eric Wieling
- [Asterisk-Dev] Asterisk crash a box when log is size  2147483647
 
James Sizemore
- [Asterisk-Dev] DATETIME Variable or TIME in general
 
Nicholas Romero
- [Asterisk-Dev] Anything wrong with sleep()ing in pbx_spool.c
 
James Sharp
- [Asterisk-Dev] Anyone need snom 100 phone?
 
Steve Radich
- [Asterisk-Dev] Anyone need snom 100 phone?
 
Wade J. Weppler
- [Asterisk-Dev] opteron anyone ?
 
Michael Bielicki
- [Asterisk-Dev] DATETIME Variable or TIME in general
 
Ben Miller
- [Asterisk-Dev] asterisk-oh323 Bugs (perhaps)
 
Rattana BIV
- [Asterisk-Dev] SCTP support in asterisk?
 
Roy Sigurd Karlsbakk
- [Asterisk-Dev] Core Dump
 
Eric Wieling
- [Asterisk-Dev] [Asterisk-Users] Directory for Cisco 7960
 
Jared Smith
- [Asterisk-Dev] IAX Trunking between 2 Asterisks
 
Kang.ChenJi at c3smail.monmouth.army.mil
- [Asterisk-Dev] IAX Trunking between 2 Asterisks
 
Kang.ChenJi at c3smail.monmouth.army.mil
- [Asterisk-Dev] status of Registration
 
Kang.ChenJi at c3smail.monmouth.army.mil
- [Asterisk-Dev] SIP Connection on
 
Kang.ChenJi at c3smail.monmouth.army.mil
- [Asterisk-Dev] Postgres and Asterisk
 
Peter Nixon
- [Asterisk-Dev] ast_openstream bug?
 
Tais M. Hansen
- [Asterisk-Dev] Real billing time in CDR
 
Sistemas - ANALITICA MD
- [Asterisk-Dev] logger.conf
 
Tais M. Hansen
- [Asterisk-Dev] Fw: chan_h323 - Segmentation fault (core dumped)
 
CW_ASN
- [Asterisk-Dev] Re: [Asterisk-Users] Queues and max time in queue timeout?
 
James Sizemore
- [Asterisk-Dev] VoIP hardphone development
 
John Todd
- [Asterisk-Dev] AGI script looping not killed by Asterisk
 
Areski
- [Asterisk-Dev] (E)AGI Threads
 
Ousmane Wilane
- [Asterisk-Dev] Notify a channel and call transfer
 
astdev at netscape.net
- [Asterisk-Dev] README.channels bugnote needs your help
 
John Todd
- [Asterisk-Dev] chan_skinny core dump
 
CW_ASN
- [Asterisk-Dev] Possible bug in the Queue Application and feature request
 
Alex Lopez
- [Asterisk-Dev] Retrieve Message from Voicemail
 
Kang.ChenJi at c3smail.monmouth.army.mil
- [Asterisk-Dev] Possible bug in the Queue Application and
 featurerequest
 
TC
- [Asterisk-Dev] Possible bug in the Queue Application and featurerequest
 
Alex Lopez
- [Asterisk-Dev] README.channels bugnote needs your help
 
James Golovich
- [Asterisk-Dev] Makefile problem
 
Peter Nixon
- [Asterisk-Dev] Retrieve Message from Voicemail
 
Kang.ChenJi at c3smail.monmouth.army.mil
- [Asterisk-Dev] Makefile patch
 
Jared Mauch
- [Asterisk-Dev] fixed makefile patch
 
Jared Mauch
- [Asterisk-Dev] Problem with asterisk/codecs/gsm/Makefile asterisk/codecs/mp3
 
Scott Lambert
- [Asterisk-Dev] D/120JCT-LS Setup issue.
 
James T. Boylan
- [Asterisk-Dev] [PATCH] app_agi.c
 
Steven Critchfield
- [Asterisk-Dev] Meetme Conference Configuration
 
Kang.ChenJi at c3smail.monmouth.army.mil
- [Asterisk-Dev] sip nat autodetect?
 
Jared Mauch
- [Asterisk-Dev] Music Onhold Configuration
 
Kang.ChenJi at c3smail.monmouth.army.mil
- [Asterisk-Dev] Meetme Conference Configuration
 
Kang.ChenJi at c3smail.monmouth.army.mil
- [Asterisk-Dev] [bug] sizeof buf in pbx.c:handle_show_application() is to small
 
Steven Critchfield
- [Asterisk-Dev] Extensions not working after a sip Dial() ?
 
James Sizemore
- [Asterisk-Dev] Freebsd - timer
 
Olle E. Johansson
- [Asterisk-Dev] Fastman aka Flash-Astman + String Terminations
 
Brancaleoni Matteo
- [Asterisk-Dev] Some questions of heavy * deployment and stability.
 
Anton Tinchev
- [Asterisk-Dev] PSQL logging bug with CAPI ?
 
Florian Overkamp
- [Asterisk-Dev] Problems with making calls from one Gnophone to another through the local Asterisk Server
 
sheebaaggarwal at hfcl.com
- [Asterisk-Dev] SIP question
 
Rattana BIV
- [Asterisk-Dev] TDM40P FXO Modules
 
Eric Wieling
- [Asterisk-Dev] Third Party VoiceMail Interface
 
Jonathan.Hopper at kaz-group.com
- [Asterisk-Dev] Problems with making calls from one Gnophone to another through the local Asterisk Server
 
Sheeba Aggarwal
- [Asterisk-Dev] Problems with making calls from one Gnophone to another through the local Asterisk Server
 
Sheeba Aggarwal
- [Asterisk-Dev] Problems with making calls from one Gnophone to another through the local Asterisk Server
 
Sheeba Aggarwal
- [Asterisk-Dev] Problems with making calls from one Gnophone to another through the local Asterisk Server
 
Sheeba Aggarwal
- [Asterisk-Dev] Problems with making calls from one Gnophone to another through the local Asterisk Server
 
Sheeba Aggarwal
- [Asterisk-Dev] IAX with dynamic echo cancellation - what do you think?
 
John Harragin
- [Asterisk-Dev] Minimal Installation
 
Nathan Littlepage
- [Asterisk-Dev] Jitter Buffer Settings
 
Eric Wieling
- [Asterisk-Dev] [Asterisk-Users] wcfxo fails to load
 
Chris Albertson
- [Asterisk-Dev] type=peer and context=
 
Eric Wieling
- [Asterisk-Dev] app_festival doesn't use Zaptel timeing
 
Eric Wieling
- [Asterisk-Dev] MGCP error for Cisco 7750 FXO card
 
aaron rohde
- [Asterisk-Dev] Status of chan_dialogic Licensing.
 
jboylan at akila.com
- [Asterisk-Dev] Status of chan_dialogic Licensing.
 
CW_ASN - Gus
- [Asterisk-Dev] sql storage for voicemail
 
Steve Creel
- [Asterisk-Dev] potential problem with IAX.
 
Steven Critchfield
- [Asterisk-Dev] Test, please ignore
 
Eric Wieling
- [Asterisk-Dev] [PATCH] for apps/app_agi.c
 
Steven Critchfield
- [Asterisk-Dev] internationalisation of voice-prompts
 
Michiel Betel
- [Asterisk-Dev] Patch: Fix for  SIP bug #116
 
Bisker, Scott (7805)
- [Asterisk-Dev] Asterisk for 2.6.0 series kernels?
 
Dorian Gray
- [Asterisk-Dev] Behaviour on retransmission of SIP 407 responses
 
Ginés Gómez
- [Asterisk-Dev] Re: Asterisk and RADIUS
 
RAD Development
- [Asterisk-Dev] important feature missing?!
 
Payam Shabanian
- [Asterisk-Dev] alpha/numeric paging
 
ASTERISK
- [Asterisk-Dev] app_privacy
 
Olle E. Johansson
- [Asterisk-Dev] possible Bug in "Originate" manager command
 
Thomas Haeger
- [Asterisk-Dev] Re: Asterisk and RADIUS
 
RAD Development
- [Asterisk-Dev] chan_capi core dump
 
Roy Sigurd Karlsbakk
- [Asterisk-Dev] DTMF detection on modem(ISDN)
 
Tomaz Izanc
- [Asterisk-Dev] Java based AGI
 
Johanna Kangas
- [Asterisk-Dev] conf bridge/nat solution
 
Jared Mauch
- [Asterisk-Dev] setvar(SIP_CODEC) bug?
 
Luis Benavente
- [Asterisk-Dev] Pluggable authentication for assorted channels
 
James Sharp
- [Asterisk-Dev] [Asterisk-Users] Upcoming Major CVS Changes
 
Mark Spencer
- [Asterisk-Dev] who added silence detection to app_agi.c?
 
Steven Critchfield
- [Asterisk-Dev] Re: [Asterisk-Users] SIP behind NAT, workaround to make W Snel's very welcome fix work both for inside *and* outside clients
 
Chris Albertson
- [Asterisk-Dev] X100P/ATA-186
 
Phillip Jackson, President & CEO
- [Asterisk-Dev] e100p errors
 
Roy Sigurd Karlsbakk
- [Asterisk-Dev] Re: callerid= in iax.conf does not work when incoming caller id is blank
 
Linus Surguy
- [Asterisk-Dev] probs with loading tor2 and wcusb
 
Thomas Haeger
- [Asterisk-Dev] Re: [Asterisk-Users] Call transfering, conferencing - From Users list
 
Nicholas Romero
- [Asterisk-Dev] [Asterisk-Users] Call transfering, conferencing - From Users list
 
Nicholas Romero
- [Asterisk-Dev] IAX on SNOM Source
 
Nicholas Romero
- [Asterisk-Dev] Trace / Log Transfers
 
Markus Mayer
- [Asterisk-Dev] PATCH: rtp.c, passing g723.1, infinite loop
 
Bill Leckey
- [Asterisk-Dev] Directory Dialing Plan
 
Ryan R. Fligg
- [Asterisk-Dev] NAT patch for chan_sip.c
 
William Waites
- [Asterisk-Dev] Echo and Channel Monitor revisited
 
Nicholas Romero
- [Asterisk-Dev] Question about quad E1
 
Micke Andersson
- [Asterisk-Dev] two NAT patches and STUN
 
Chris Albertson
- [Asterisk-Dev] [BUG] [SIMPLE FIX] wct1xxp.c bug
 
Steven Critchfield
    
      Last message date: 
       Fri Oct 31 12:11:31 MST 2003
    Archived on: Tue Sep  5 14:26:46 MST 2006
    
   
     
     
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