April 2003 Archives by author

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Starting: Tue Apr 1 01:18:13 MST 2003
Ending: Wed Apr 30 16:07:42 MST 2003
Messages: 315

• [Asterisk-Dev] Asterisk crashes on voicemail timeout   Ask Bjørn Hansen
• [Asterisk-Dev] Noisy TDM20   Ask Bjørn Hansen
• [Asterisk-Dev] H323 processing limit   Xara Developement
• [Asterisk-Dev] CVS-04/09/03-13:12:31 SIP Calls cut off   Mikael Andersson
• [Asterisk-Dev] CVS-04/09/03-13:12:31 SIP Calls cut off   Mikael Andersson
• [Asterisk-Dev] CVS-04/09/03-13:12:31 SIP Calls cut off   Mikael Andersson
• [Asterisk-Dev] Problem with meetme   Mikael Andersson
• [Asterisk-Dev] adsi phone codes?   Joe Antkowiak
• [Asterisk-Dev] adsi phone codes?   Joe Antkowiak
• [Asterisk-Dev] adsi phone codes?   Joe Antkowiak
• [Asterisk-Dev] DNID   Asterisk
• [Asterisk-Dev] correction   Asterisk
• [Asterisk-Dev] Voicetronix driver for V6PCI/V12PCI - first draft   Paul Bagyenda
• [Asterisk-Dev] Voicetronix Openswitch driver - draft 2   Paul Bagyenda
• [Asterisk-Dev] Changes to SIP   Michael Baird
• [Asterisk-Dev] Channel bank w/ bri card and Zaptel   George Bean
• [Asterisk-Dev] best approach for a new isdn driver mainly for asterisk.. ?!   George Bean
• [Asterisk-Dev] best approach for a new isdn driver mainly for asterisk.. ?!   George Bean
• [Asterisk-Dev] best approach for a new isdn driver mainly for asterisk.. ?!   George Bean
• [Asterisk-Dev] best approach for a new isdn driver mainly for asterisk.. ?!   George Bean
• [Asterisk-Dev] unique cdr id per call   Michiel Betel
• [Asterisk-Dev] unique cdr id per call   Michiel Betel
• [Asterisk-Dev] Documentation bug (I think)   Michiel Betel
• [Asterisk-Dev] SIP-to-IAX[2] problems   Michael Bielicki
• [Asterisk-Dev] unresolved symbols with latest openh323 and chan_h323   Michael Bielicki
• [Asterisk-Dev] cvs coredump   Andre Bierwirth
• [Asterisk-Dev] Asterisk 0.4 Seg Faulted after 4 days   Matteo Brancaleoni
• R: [Asterisk-Dev] persistent SIP registration   Matteo Brancaleoni
• [Asterisk-Dev] Channel bank w/ bri card and Zaptel   Steve Brown
• [Asterisk-Dev] Channel bank w/ bri card and Zaptel   Steve Brown
• [Asterisk-Dev] zttool channel identification   Steve Brown
• [Asterisk-Dev] Time stamps   Steve Brown
• [Asterisk-Dev] Time stamps   Steve Brown
• [Asterisk-Dev] Time stamps   Steve Brown
• [Asterisk-Dev] zaptel fcshdlc extension - need advice   Steve Brown
• [Asterisk-Dev] best approach for a new isdn driver mainly for asterisk.. ?!   Steve Brown
• [Asterisk-Dev] best approach for a new isdn driver mainly for asterisk.. ?!   Steve Brown
• [Asterisk-Dev] Hangup detection alternatives   Brian Capouch
• [Asterisk-Dev] Attempted to delete non-existant schedule entry   Brian Capouch
• [Asterisk-Dev] Attempted to delete non-existant schedule entry   Brian Capouch
• [Asterisk-Dev] CVS-04/09/03-13:12:31 SIP Calls cut off   Brian Capouch
• [Asterisk-Dev] Vonage as FXO?   Brian Capouch
• [Asterisk-Dev] Asterisk crashes on voicemail timeout   Brian Capouch
• [Asterisk-Dev] Noisy TDM20   Brian Capouch
• [Asterisk-Dev] Noisy TDM20   Brian Capouch
• [Asterisk-Dev] Hangup detection using ATA186/Vonage for "backhaul?"   Brian Capouch
• [Asterisk-Dev] sigh. . iconnecthere (behind NAT) appears broken again. .   Brian Capouch
• [Asterisk-Dev] Asterisk no register in openh323gk   Lubomir Christov
• [Asterisk-Dev] Channel bank w/ bri card and Zaptel   Steven Critchfield
• [Asterisk-Dev] Channel bank w/ bri card and Zaptel   Steven Critchfield
• [Asterisk-Dev] zaptel rpm submission   Steven Critchfield
• [Asterisk-Dev] fast-forward & rewind   Steven Critchfield
• [Asterisk-Dev] mp3player related segfaults   Steven Critchfield
• [Asterisk-Dev] single bit T1 errors (Zhone again)   Steven Critchfield
• [Asterisk-Dev] single bit T1 errors (Zhone again)   Steven Critchfield
• [Asterisk-Dev] FAX Subsystem discussion.   Steven Critchfield
• [Asterisk-Dev] FAX Subsystem discussion.   Steven Critchfield
• [Asterisk-Dev] best approach for a new isdn driver mainly for asterisk.. ?!   Steven Critchfield
• [Asterisk-Dev] best approach for a new isdn driver mainly for asterisk.. ?!   Steven Critchfield
• [Asterisk-Dev] em_w signalling problems   Steven Critchfield
• [Asterisk-Dev] em_w signalling problems   Steven Critchfield
• [Asterisk-Dev] best approach for a new isdn driver mainly for asterisk.. ?!   Steven Critchfield
• [Asterisk-Dev] Patch to add Record-Route handling for SIP, also honour Connect: properly   Stephen Davies
• [Asterisk-Dev] Patch to add Record-Route handling for SIP, also honour Connect: properly: TRY 2   Stephen Davies
• [Asterisk-Dev] SIP CANCEL/BYE authorization patch   Stephen Davies
• [Asterisk-Dev] SIP: Problem in recent Contact changes - results in peer context being ignored   Stephen Davies
• [Asterisk-Dev] Don't understand authentication with SIP?   Stephen Davies
• [Asterisk-Dev] PATCH: Fix code to find peer in check_user   Stephen Davies
• [Asterisk-Dev] Proposal for better matching of incoming calls to known sip_users   Stephen Davies
• [Asterisk-Dev] PATCH: Ability to override "From" for calls to a peer, also fix inaddrcmp sense   Stephen Davies
• [Asterisk-Dev] Re: PATCH: Ability to override "From" for calls to a peer, also fix inaddrcmp sense   Stephen Davies
• [Asterisk-Dev] CVS-04/09/03-13:12:31 SIP Calls cut off   Stephen Davies
• [Asterisk-Dev] Feature request: auto-guessing NAT status based on From: IP address   Stephen Davies
• [Asterisk-Dev] Registrations SHOULD use same Call-ID. Packet8 seems to care   Stephen Davies
• [Asterisk-Dev] Re: Registrations SHOULD use same Call-ID. Packet8 seems to care   Stephen Davies
• [Asterisk-Dev] Re: Registrations SHOULD use same Call-ID. Packet8 seems to care   Stephen Davies
• [Asterisk-Dev] best approach for a new isdn driver mainly for asterisk.. ?!   Stephen Davies
• [Asterisk-Dev] best approach for a new isdn driver mainly for asterisk.. ?!   Stephen Davies
• [Asterisk-Dev] IAX deadlock (?)   Stephen Davies
• [Asterisk-Dev] Changes to SIP   James Dennis
• [Asterisk-Dev] Registration patch   James Dennis
• [Asterisk-Dev] SIP CANCEL/BYE authorization patch   James Dennis
• [Asterisk-Dev] SIP CANCEL/BYE authorization patch   James Dennis
• [Asterisk-Dev] SIP CANCEL/BYE authorization patch   James Dennis
• [Asterisk-Dev] SIP: tag fix   James Dennis
• [Asterisk-Dev] Re: PATCH: Ability to override "From" for calls to a peer, also fix inaddrcmp sense   James Dennis
• [Asterisk-Dev] Re: PATCH: Ability to override "From" for calls to a peer, also fix inaddrcmp sense   James Dennis
• [Asterisk-Dev] CVS-04/09/03-13:12:31 SIP Calls cut off   James Dennis
• [Asterisk-Dev] FAX Subsystem discussion.   Bruce Ferrell
• [Asterisk-Dev] Changes to SIP   Christoph Frei
• [Asterisk-Dev] Changes to SIP   Christoph Frei
• [Asterisk-Dev] SIP CANCEL/BYE authorization patch   Christoph Frei
• [Asterisk-Dev] SIP CANCEL/BYE authorization patch   Christoph Frei
• [Asterisk-Dev] Problems with ${EXTEN}   Christoph Frei
• [Asterisk-Dev] Problems with ${EXTEN}   Christoph Frei
• [Asterisk-Dev] best approach for a new isdn driver mainly for asterisk.. ?!   Gary
• [Asterisk-Dev] MGCP and ChanIsAvail problem   James Golovich
• [Asterisk-Dev] unique cdr id per call   James Golovich
• [Asterisk-Dev] A few points   Adam Goryachev
• [Asterisk-Dev] Documentation bug (I think)   Adam Goryachev
• [Asterisk-Dev] Documentation bug (I think) - Another one   Adam Goryachev
• [Asterisk-Dev] fast-forward & rewind   Jim Gottlieb
• [Asterisk-Dev] SIP-to-IAX[2] problems   Jim Gottlieb
• [Asterisk-Dev] no audio after many transfers   Jim Gottlieb
• [Asterisk-Dev] no audio after many transfers   Jim Gottlieb
• [Asterisk-Dev] no audio after many transfers   Jim Gottlieb
• [Asterisk-Dev] no audio after many transfers   Jim Gottlieb
• [Asterisk-Dev] Asterisk crashes on voicemail timeout   Jim Gottlieb
• [Asterisk-Dev] Hangup detection using ATA186/Vonage for "backhaul?"   Jim Gottlieb
• [Asterisk-Dev] Patch for SIP and no callerID   Michael Graff
• [Asterisk-Dev] g723 (probably again...)   Michael Graff
• [Asterisk-Dev] g723 (probably again...)   Michael Graff
• [Asterisk-Dev] dialogic card   Clay Graner
• [Asterisk-Dev] Adding Unique CallID to ast_channel structure   Levent Guendogdu
• [Asterisk-Dev] H323 processing limit   Abdul Hakeem
• [Asterisk-Dev] FAX Subsystem discussion.   John Harragin
• [Asterisk-Dev] FAX Subsystem discussion.   John Harragin
• [Asterisk-Dev] FAX Subsystem discussion.   John Harragin
• [Asterisk-Dev] no audio after many transfers   John Harragin
• [Asterisk-Dev] Asterisk on RedHat 9.0 - fix   David Hollis
• [Asterisk-Dev] dialogic card   Stephen Hui
• [Asterisk-Dev] compile error   Tomaz Izanc
• [Asterisk-Dev] compile error   Tomaz Izanc
• [Asterisk-Dev] compile error   Tomaz Izanc
• [Asterisk-Dev] compile error   Tomaz Izanc
• [Asterisk-Dev] compile error   Tomaz Izanc
• [Asterisk-Dev] zaptel rpm submission   Carey Jung
• [Asterisk-Dev] zaptel rpm submission   Carey Jung
• [Asterisk-Dev] Modules doesn't free from asterisk   Miroslaw KLABA
• [Asterisk-Dev] persistent SIP registration   Miroslaw KLABA
• [Asterisk-Dev] Modules doesn't free from asterisk   Miroslaw KLABA
• [Asterisk-Dev] Modules doesn't free from asterisk   Miroslaw KLABA
• [Asterisk-Dev] Modules doesn't free from asterisk   Miroslaw KLABA
• [Asterisk-Dev] Problem with H.323   Miroslaw KLABA
• [Asterisk-Dev] Memory usage?   Miroslaw KLABA
• [Asterisk-Dev] Anyone interested in sharing PRI/T1 capacity in US (Virginia)?   Jim Kunkel
• [Asterisk-Dev] Changes to SIP   Gregg Lebovitz
• [Asterisk-Dev] SIP CANCEL/BYE authorization patch   Gregg Lebovitz
• [Asterisk-Dev] SIP CANCEL/BYE authorization patch   Gregg Lebovitz
• [Asterisk-Dev] iconnect hangup works   Gregg Lebovitz
• [Asterisk-Dev] Time stamps   Tilghman Lesher
• [Asterisk-Dev] Feature request: auto-guessing NAT status based on From: IP address   Tilghman Lesher
• [Asterisk-Dev] GotoIfTime patch   Tilghman Lesher
• [Asterisk-Dev] no audio after many transfers   Tilghman Lesher
• [Asterisk-Dev] persistent SIP registration   Howard Lin
• [Asterisk-Dev] persistent SIP registration   Howard Lin
• [Asterisk-Dev] persistent SIP registration   Howard Lin
• [Asterisk-Dev] MGCP and ChanIsAvail problem   Pavel Litvinenko
• [Asterisk-Dev] mgcp call features patch   Pavel Litvinenko
• [Asterisk-Dev] Some makefile changes   Thorsten Lockert
• [Asterisk-Dev] Duplicate symbol   Thorsten Lockert
• [Asterisk-Dev] (no subject)   Thorsten Lockert
• [Asterisk-Dev] ENODATA is not standard   Thorsten Lockert
• [Asterisk-Dev] strdupa not common   Thorsten Lockert
• [Asterisk-Dev] Dynamic loader changes   Thorsten Lockert
• [Asterisk-Dev] Missing ranlib call, better detection   Thorsten Lockert
• [Asterisk-Dev] IPPROTO_IP is more common than SOL_IP   Thorsten Lockert
• [Asterisk-Dev] setpriority on non-Linux system   Thorsten Lockert
• [Asterisk-Dev] More include fixes   Thorsten Lockert
• [Asterisk-Dev] __PMT use in db1-ast   Thorsten Lockert
• [Asterisk-Dev] Some select(2) usage Linux-specific   Thorsten Lockert
• [Asterisk-Dev] Blocking open(2) does not work with some thread implementations   Thorsten Lockert
• [Asterisk-Dev] Shared module symbols leak   Thorsten Lockert
• [Asterisk-Dev] More include, IP TOS fixes   Thorsten Lockert
• [Asterisk-Dev] Some select(2) usage Linux-specific   Thorsten Lockert
• [Asterisk-Dev] Some select(2) usage Linux specific, take 2   Thorsten Lockert
• [Asterisk-Dev] Makefile changes for more platforms   Thorsten Lockert
• [Asterisk-Dev] Reorder tests in editline/configure{,.in}   Thorsten Lockert
• [Asterisk-Dev] Different patch for __PMT in db1-ast   Thorsten Lockert
• [Asterisk-Dev] More portable pthread initializers   Thorsten Lockert
• [Asterisk-Dev] strdupa not common, take 2   Thorsten Lockert
• [Asterisk-Dev] Shared module symbols leak, take 2   Thorsten Lockert
• [Asterisk-Dev] Anyone interested in sharing PRI/T1 capacity in US (Virginia)?   C. Maj
• [Asterisk-Dev] X100P doesn`t call   Michael Manousos
• [Asterisk-Dev] H323 processing limit   Michael Manousos
• [Asterisk-Dev] Patch: SIP accountcode, amaflags   Brancaleoni Matteo
• [Asterisk-Dev] compile error   Brancaleoni Matteo
• [Asterisk-Dev] cvs coredump   Jared Mauch
• [Asterisk-Dev] H323 processing limit   Jeremy McNamara
• [Asterisk-Dev] compile error   Jeremy McNamara
• [Asterisk-Dev] no audio after many transfers   Jeremy McNamara
• [Asterisk-Dev] unresolved symbols with latest openh323 and chan_h323   Jeremy McNamara
• [Asterisk-Dev] g723 (probably again...)   Jeremy McNamara
• [Asterisk-Dev] best approach for a new isdn driver mainly for asterisk.. ?!   Pauline Middelink
• [Asterisk-Dev] best approach for a new isdn driver mainly for asterisk.. ?!   Pauline Middelink
• [Asterisk-Dev] best approach for a new isdn driver mainly for asterisk.. ?!   Pauline Middelink
• [Asterisk-Dev] persistent SIP registration   Benjamin Miller
• [Asterisk-Dev] FAX Subsystem discussion.   Florian Overkamp
• [Asterisk-Dev] format of voicemail files and extensions problem   Dave Packham
• [Asterisk-Dev] Noisy TDM20   Dave Packham
• [Asterisk-Dev] Noisy TDM20   Dave Packham
• [Asterisk-Dev] best approach for a new isdn driver mainly for asterisk.. ?!   Patrick
• [Asterisk-Dev] best approach for a new isdn driver mainly for asterisk.. ?!   Patrick
• [Asterisk-Dev] best approach for a new isdn driver mainly for asterisk.. ?!   Patrick
• [Asterisk-Dev] best approach for a new isdn driver mainly for asterisk.. ?!   Patrick
• [Asterisk-Dev] best approach for a new isdn driver mainly for asterisk.. ?!   Patrick
• [Asterisk-Dev] Documentation bug (I think)   Francisco Perez-Landaeta
• [Asterisk-Dev] single bit T1 errors (Zhone again)   Don Pobanz
• [Asterisk-Dev] single bit T1 errors (Zhone again)   Jon Pounder
• [Asterisk-Dev] single bit T1 errors (Zhone again)   Jon Pounder
• [Asterisk-Dev] single bit T1 errors (Zhone again)   Jon Pounder
• [Asterisk-Dev] FAX Subsystem discussion.   Jon Pounder
• [Asterisk-Dev] FAX Subsystem discussion.   Jon Pounder
• [Asterisk-Dev] setpriority on non-Linux system   Jon Pounder
• [Asterisk-Dev] Memory usage?   Jon Pounder
• [Asterisk-Dev] adsi phone codes?   Jon Pounder
• [Asterisk-Dev] adsi phone codes?   Jon Pounder
• [Asterisk-Dev] zaptel Makefile cleanup   Steven Pritchard
• [Asterisk-Dev] mgcp call features patch   Karl Putland
• [Asterisk-Dev] mgcp call features cont.   Karl Putland
• [Asterisk-Dev] chan_mgcp patch ready for beta testing   Karl Putland
• [Asterisk-Dev] patch for addmailbox   Karl Putland
• [Asterisk-Dev] Asterisk on RedHat 9.0 - fix   Martin Pycko
• [Asterisk-Dev] Problems with ${EXTEN}   Martin Pycko
• [Asterisk-Dev] Modules doesn't free from asterisk   Martin Pycko
• [Asterisk-Dev] compile error   Martin Pycko
• [Asterisk-Dev] Anyone interested in sharing PRI/T1 capacity i n US (Virginia)?   Steve Radich
• [Asterisk-Dev] SIP DTMF causing voice quality loss?   Steve Radich
• [Asterisk-Dev] fast-forward & rewind   Mike Reiling
• [Asterisk-Dev] fast-forward & rewind   Mike Reiling
• [Asterisk-Dev] Problem with IAX   Sergio Serrano Revuelto
• [Asterisk-Dev] Problem with IAX   Sergio Serrano Revuelto
• [Asterisk-Dev] Asterisk no register in openh323gk   Sergio Serrano Revuelto
• [Asterisk-Dev] Not Voice only noise   Sergio Serrano Revuelto
• [Asterisk-Dev] Not Voice only noise   Sergio Serrano Revuelto
• [Asterisk-Dev] X100P doesn`t call   Sergio Serrano Revuelto
• [Asterisk-Dev] X100P doesn`t call   Sergio Serrano Revuelto
• [Asterisk-Dev] X100P doesn`t call   Sergio Serrano Revuelto
• [Asterisk-Dev] X100P doesn`t call   Sergio Serrano Revuelto
• [Asterisk-Dev] Problem with H.323   Sergio Serrano Revuelto
• [Asterisk-Dev] compile error   Sergio Serrano Revuelto
• [Asterisk-Dev] Low level of sounds in ISDN   Sergio Serrano Revuelto
• [Asterisk-Dev] H.323 Question   Sergio Serrano Revuelto
• [Asterisk-Dev] cisco 2620 to pri card   Skuse, Phil
• [Asterisk-Dev] compile error   Jared Smith
• [Asterisk-Dev] Sending ANI via MF T-1 Trunks   Mark Spencer
• [Asterisk-Dev] zttool channel identification   Mark Spencer
• [Asterisk-Dev] Changes to SIP   Mark Spencer
• [Asterisk-Dev] Changes to SIP   Mark Spencer
• [Asterisk-Dev] zaptel rpm submission   Mark Spencer
• [Asterisk-Dev] SIP CANCEL/BYE authorization patch   Mark Spencer
• [Asterisk-Dev] Attempted to delete non-existant schedule entry   Mark Spencer
• [Asterisk-Dev] Re: PATCH: Ability to override "From" for calls to a peer, also fix inaddrcmp sense   Mark Spencer
• [Asterisk-Dev] Documentation bug (I think)   Mark Spencer
• [Asterisk-Dev] X100P doesn`t call   Mark Spencer
• [Asterisk-Dev] Documentation bug (I think) - Another one   Mark Spencer
• [Asterisk-Dev] Voicetronix driver for V6PCI/V12PCI - first draft   Mark Spencer
• [Asterisk-Dev] mp3player related segfaults   Mark Spencer
• [Asterisk-Dev] Feature request: reading in parts of config files via URI   Mark Spencer
• [Asterisk-Dev] single bit T1 errors (Zhone again)   Mark Spencer
• [Asterisk-Dev] single bit T1 errors (Zhone again)   Mark Spencer
• [Asterisk-Dev] Feature request: auto-guessing NAT status based on From: IP address   Mark Spencer
• [Asterisk-Dev] Feature request: auto-guessing NAT status based on From: IP address   Mark Spencer
• [Asterisk-Dev] Problems with ${EXTEN}   Mark Spencer
• [Asterisk-Dev] Problems with ${EXTEN}   Mark Spencer
• [Asterisk-Dev] SIP registration now fails ?   Mark Spencer
• [Asterisk-Dev] SIP registration now fails ?   Mark Spencer
• [Asterisk-Dev] Re: Registrations SHOULD use same Call-ID. Packet8 seems to care   Mark Spencer
• [Asterisk-Dev] Re: Registrations SHOULD use same Call-ID. Packet8 seems to care   Mark Spencer
• [Asterisk-Dev] Some select(2) usage Linux-specific   Mark Spencer
• [Asterisk-Dev] compile error   Mark Spencer
• [Asterisk-Dev] IAX deadlock (?)   Mark Spencer
• [Asterisk-Dev] no audio after many transfers   Mark Spencer
• [Asterisk-Dev] Asterisk crashes on voicemail timeout   Mark Spencer
• [Asterisk-Dev] Noisy TDM20   Mark Spencer
• [Asterisk-Dev] best approach for a new isdn driver mainly for asterisk.. ?!   Mark Spencer
• [Asterisk-Dev] Noisy TDM20   Mark Spencer
• [Asterisk-Dev] Asterisk crashes on voicemail timeout   Mark Spencer
• [Asterisk-Dev] Hangup detection using ATA186/Vonage for "backhaul?"   Mark Spencer
• [Asterisk-Dev] Adding Unique CallID to ast_channel structure   Mark Spencer
• [Asterisk-Dev] New CVS issues with Remote UNIX connection warnings in CLI   Mark Spencer
• [Asterisk-Dev] SIP registration now fails ?   Iain Stevenson
• [Asterisk-Dev] SIP registration now fails ?   Iain Stevenson
• [Asterisk-Dev] best approach for a new isdn driver mainly for asterisk.. ?!   Iain Stevenson
• [Asterisk-Dev] Feature request: reading in parts of config files via URI   John Todd
• [Asterisk-Dev] Feature request: reading in parts of config files via URI   John Todd
• [Asterisk-Dev] Feature request: auto-guessing NAT status based on From: IP address   John Todd
• [Asterisk-Dev] Feature request: auto-guessing NAT status based on From: IP address   John Todd
• [Asterisk-Dev] Feature request: auto-guessing NAT status based on From: IP address   John Todd
• [Asterisk-Dev] SIP DTMF causing voice quality loss?   John Todd
• [Asterisk-Dev] GotoIfTime patch   John Todd
• [Asterisk-Dev] Error in "iax2 show channels"   John Todd
• [Asterisk-Dev] IAX2 problems   John Todd
• [Asterisk-Dev] Porting to Windows   Massimo Vellucci
• [Asterisk-Dev] best approach for a new isdn driver mainly for asterisk.. ?!   Andrea Venturi
• [Asterisk-Dev] best approach for a new isdn driver mainly for asterisk.. ?!   Andrea Venturi
• [Asterisk-Dev] best approach for a new isdn driver mainly for asterisk.. ?!   Andrea Venturi
• [Asterisk-Dev] Asterisk on RedHat 9.0 - fix   DUSTIN WILDES
• [Asterisk-Dev] Asterisk on RedHat 9.0 - fix   Wade Weppler
• [Asterisk-Dev] single bit T1 errors (Zhone again)   Wade Weppler
• [Asterisk-Dev] FAX Subsystem discussion.   Chad Wicker
• [Asterisk-Dev] Interesting SIP document   Eric Wieling
• [Asterisk-Dev] Feature request: auto-guessing NAT status based on From: IP address   Eric Wieling
• [Asterisk-Dev] Problems with ${EXTEN}   Eric Wieling
• [Asterisk-Dev] Problems with ${EXTEN}   Eric Wieling
• [Asterisk-Dev] g723 (probably again...)   Eric Wieling
• [Asterisk-Dev] g723 (probably again...)   Eric Wieling
• [Asterisk-Dev] FAX Subsystem discussion.   Roger Wrethman
• [Asterisk-Dev] FAX Subsystem discussion.   Roger Wrethman
• [Asterisk-Dev] Sending ANI via MF T-1 Trunks   Tom Zimnicki
• [Asterisk-Dev] Sending ANI via MF T-1 Trunks   Tom Zimnicki
• [Asterisk-Dev] Changes to SIP   alex at pilosoft.com
• [Asterisk-Dev] Anyone interested in sharing PRI/T1 capacity i n US (Virginia)?   alex at pilosoft.com
• [Asterisk-Dev] Feature request: auto-guessing NAT status based on From: IP address   ask at develooper.com
• [Asterisk-Dev] unique cdr id per call   asterisk at billheckel.com
• [Asterisk-Dev] Adding Unique CallID to ast_channel structure   asterisk at billheckel.com
• [Asterisk-Dev] New CVS issues with Remote UNIX connection warnings in CLI   asterisk at billheckel.com
• [Asterisk-Dev] Changes to SIP   asterisk at jdennis.net
• [Asterisk-Dev] cisco 2620 to pri card   ayaz
• [Asterisk-Dev] any experience with PacketCable NCS 1.0 or SGCP 1.x???   flying_sushi at SAFe-mail.net
• [Asterisk-Dev] !!! chan_h323 does not seem to work properly !!!!   flying_sushi at SAFe-mail.net
• [Asterisk-Dev] single bit T1 errors (Zhone again)   howwhi at vcch.com
• [Asterisk-Dev] Documentation bug (I think)   stephen jarusinsky
• [Asterisk-Dev] Anyone interested in sharing PRI/T1 capacity in US (Virginia)?   stephen jarusinsky
• [Asterisk-Dev] best approach for a new isdn driver mainly for asterisk.. ?!   tmassey at obscorp.com

Last message date: Wed Apr 30 16:07:42 MST 2003
Archived on: Tue Sep 5 14:26:39 MST 2006

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