May 2003 Archives by author

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Starting: Thu May 1 06:53:40 MST 2003
Ending: Sat May 31 23:14:36 MST 2003
Messages: 265

• [Asterisk-Dev] Asterisk channel status is "down"   Petr Michálek
• [Asterisk-Dev] Some build fixes for h323 plugin   Paweł Gołaszewski
• [Asterisk-Dev] Some build fixes for h323 plugin   Paweł Gołaszewski
• [Asterisk-Dev] DESTDIR capability   Paweł Gołaszewski
• [Asterisk-Dev] DESTDIR capability   Paweł Gołaszewski
• [Asterisk-Dev] DESTDIR capability   Paweł Gołaszewski
• [Asterisk-Dev] linking and libs fixes for chan_h323   Paweł Gołaszewski
• [Asterisk-Dev] Gastman fixes   Paweł Gołaszewski
• [Asterisk-Dev] modified DESTDIR patch   Paweł Gołaszewski
• [Asterisk-Dev] chan_sip   Paweł Gołaszewski
• [Asterisk-Dev] Any chance at Asterisk supporting Speex?   Paweł Gołaszewski
• [Asterisk-Dev] mpg123 - buffering problem   Garry Adkins
• [Asterisk-Dev] IAX over 1394   Chris Albertson
• [Asterisk-Dev] Voicemenu - directory   Mikael Andersson
• [Asterisk-Dev] Komodo Fone 300 / ATA-182   Mikael Andersson
• [Asterisk-Dev] Komodo Fone 300 / ATA-182   Mikael Andersson
• [Asterisk-Dev] Komodo Fone 300 / ATA-182   Mikael Andersson
• [Asterisk-Dev] note on music on hold and new mpg123   Joe Antkowiak
• [Asterisk-Dev] IAX over 1394   Joe Antkowiak
• [Asterisk-Dev] CAC Adit 600 FXO-8 Cards   Joe Antkowiak
• pt480 too Re: [Asterisk-Dev] Aastra PT390 ADSI Issues (w/ Patch)   Joe Antkowiak
• [Asterisk-Dev] ADSI FDN for voicemail app?   Joe Antkowiak
• [Asterisk-Dev] API asterisk   Rattana BIV
• [Asterisk-Dev] -- CPE does not support Call Waiting Caller*ID.   Brad Bergman
• [Asterisk-Dev] Anyone doing QOS routing on Linux for SIP/RTP?   Alberto Bertogli
• [Asterisk-Dev] Anyone doing QOS routing on Linux for SIP/RTP?   Alberto Bertogli
• [Asterisk-Dev] enum.c -> HEADER ??   Michael Bielicki
• [Asterisk-Dev] some thoughts and some bugs :)   Michael Bielicki
• [Asterisk-Dev] 2nd part of my last mail   Michael Bielicki
• [Asterisk-Dev] Any chance at Asterisk supporting Speex?   Michael Bielicki
• [Asterisk-Dev] Patch to fix moh not closing properly   Sam Bingner
• [Asterisk-Dev] FW: hdlc link down errors   David P. Boswell
• [Asterisk-Dev] FW: hdlc link down errors   David P. Boswell
• [Asterisk-Dev] Anyone doing QOS routing on Linux for SIP/RTP?   Brian Capouch
• [Asterisk-Dev] -- CPE does not support Call Waiting Caller*ID.   Brian Capouch
• [Asterisk-Dev] -- CPE does not support Call Waiting Caller*ID.   Brian Capouch
• [Asterisk-Dev] Patch for voicemail date/time   Brian Capouch
• [Asterisk-Dev] Voicemail waiting dialtone woes   Brian Capouch
• [Asterisk-Dev] vmail.cgi hosed in recent CVS?   Brian Capouch
• [Asterisk-Dev] IAX vs. IAX2 - problems and notes   Brian Capouch
• [Asterisk-Dev] Asterisk SIP behind NAT. Possible solution.   Jamie Carl
• [Asterisk-Dev] More IAX tests with different codecs   Steven Critchfield
• [Asterisk-Dev] asterisk crashing   Steven Critchfield
• [Asterisk-Dev] asterisk crashing   Steven Critchfield
• [Asterisk-Dev] -- CPE does not support Call Waiting Caller*ID.   Steven Critchfield
• [Asterisk-Dev] IAX over 1394   Steven Critchfield
• [Asterisk-Dev] Meetme changes   Steven Critchfield
• [Asterisk-Dev] zapata, zaptel, GPL etc   Steven Critchfield
• [Asterisk-Dev] zapata, zaptel, GPL etc   Steven Critchfield
• [Asterisk-Dev] Withdrawling from the list.....   Steven Critchfield
• [Asterisk-Dev] enum.c -> HEADER ??   Stephen Davies
• [Asterisk-Dev] enum.c -> HEADER ??   Stephen Davies
• [Asterisk-Dev] enum.c -> HEADER ??   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] PATCH: Little change so acl.c will compile on my older glibc   Stephen Davies
• [Asterisk-Dev] Anyone doing QOS routing on Linux for SIP/RTP?   Stephen Davies
• [Asterisk-Dev] More IAX tests with different codecs   Stephen Davies
• [Asterisk-Dev] Compilation warning   Alex Feldman
• [Asterisk-Dev] IAX vs. IAX2 - problems and notes   Dan Fernandez
• [Asterisk-Dev] Komodo Fone 300 / ATA-182   Jim Flagg
• [Asterisk-Dev] Any chance at Asterisk supporting Speex?   Jim Flagg
• [Asterisk-Dev] Any chance at Asterisk supporting Speex?   Jim Flagg
• [Asterisk-Dev] ${RDNIS} from SIP-GW   Christoph Frei
• [Asterisk-Dev] ${RDNIS} from SIP-GW   Christoph Frei
• [Asterisk-Dev] ACD - Sorry, posted to wrong list.   Jim Friedeck
• [Asterisk-Dev] Polarity Reversal ?   Gary
• [Asterisk-Dev] Anyone doing QOS routing on Linux for SIP/RTP?   Andrew Gillham
• [Asterisk-Dev] Bug in chan_sip.c and "realod"   Andrew Gillham
• [Asterisk-Dev] ${RDNIS} from SIP-GW   James Golovich
• [Asterisk-Dev] R2/DTMF on Asterisk? How???   James Golovich
• [Asterisk-Dev] E1, crc-4 errors   Konrad Gorski
• [Asterisk-Dev] E1, crc-4 errors   Konrad Gorski
• [Asterisk-Dev] E1, crc-4 errors   Konrad Gorski
• [Asterisk-Dev] E1, crc-4 errors   Konrad Gorski
• [Asterisk-Dev] E1, crc-4 errors   Konrad Gorski
• [Asterisk-Dev] E1, crc-4 errors   Konrad Gorski
• [Asterisk-Dev] Anyone doing QOS routing on Linux for SIP/RTP?   Adam Goryachev
• [Asterisk-Dev] Anyone doing QOS routing on Linux for SIP/RTP?   Adam Goryachev
• [Asterisk-Dev] Anyone doing QOS routing on Linux for SIP/RTP?   Adam Goryachev
• [Asterisk-Dev] Patch for voicemail date/time   Jim Gottlieb
• [Asterisk-Dev] follow-on calls   Jim Gottlieb
• [Asterisk-Dev] follow-on calls   Jim Gottlieb
• [Asterisk-Dev] follow-on calls   Jim Gottlieb
• [Asterisk-Dev] Patches for caller-id on SIP?   Michael Graff
• [Asterisk-Dev] callerid toward X101P   Michael Graff
• [Asterisk-Dev] Bug in chan_sip.c and "realod"   Michael Graff
• [Asterisk-Dev] Meetme changes   Michael Graff
• [Asterisk-Dev] link browsing not working in GnoPhone   Manoj Kr. Gupta
• [Asterisk-Dev] OH323-FAX   Niclas Gustafsson
• [Asterisk-Dev] OpenH323 channel driver, Q931 Calling party number   Niclas Gustafsson
• [Asterisk-Dev] Snappier dialing   John Harragin
• [Asterisk-Dev] Snappier dialing   John Harragin
• [Asterisk-Dev] Is Asterisk-User list working?   John Harragin
• [Asterisk-Dev] Make question   John Harragin
• [Asterisk-Dev] Patch: fix writing to constant storage regression   Luke Howard
• [Asterisk-Dev] PostgreSQL CDR Support   Ray Russell Reese III
• [Asterisk-Dev] working mpg321 patch   Tomaz Izanc
• [Asterisk-Dev] compile error - cdr_mysql.so   Tomaz Izanc
• [Asterisk-Dev] compile error - cdr_mysql.so   Tomaz Izanc
• [Asterisk-Dev] Anyone doing QOS routing on Linux for SIP/RTP?   Bill Jennings
• [Asterisk-Dev] FW: hdlc link down errors   Carey Jung
• [Asterisk-Dev] FW: hdlc link down errors   Carey Jung
• [Asterisk-Dev] FW: hdlc link down errors   Carey Jung
• [Asterisk-Dev] Using linux RTC with ztdummy   Klaus-Peter Junghanns
• [Asterisk-Dev] Using linux RTC with ztdummy   Klaus-Peter Junghanns
• [Asterisk-Dev] Using linux RTC with ztdummy   Klaus-Peter Junghanns
• [Asterisk-Dev] chan_capi version 0.2.0 released   Klaus-Peter Junghanns
• [Asterisk-Dev] DESTDIR capability   Klaus-Peter Junghanns
• [Asterisk-Dev] DESTDIR capability   Klaus-Peter Junghanns
• [Asterisk-Dev] Zombie channel doesn't free   Miroslaw KLABA
• [Asterisk-Dev] libpri LAPD T200 timer   Liam Kenny
• [Asterisk-Dev] Komodo Fone 300 / ATA-182   Hemant Kumar
• [Asterisk-Dev] Hooking E100P straight to telco exchange without modems.   Hemant Kumar
• [Asterisk-Dev] zapata, zaptel, GPL etc   Hemant Kumar
• [Asterisk-Dev] zapata, zaptel, GPL etc   Hemant Kumar
• [Asterisk-Dev] Answering Machine Detection with WildCard?   Francois Lambert
• [Asterisk-Dev] depmod fails for zaptel   Gregg Lebovitz
• [Asterisk-Dev] depmod fails for zaptel   Gregg Lebovitz
• [Asterisk-Dev] depmod fails for zaptel   Gregg Lebovitz
• [Asterisk-Dev] asterisk crashing   Gregg Lebovitz
• [Asterisk-Dev] asterisk crashing   Gregg Lebovitz
• [Asterisk-Dev] asterisk still crashing   Gregg Lebovitz
• [Asterisk-Dev] Patch: SIP bindaddr not always used   Tilghman Lesher
• [Asterisk-Dev] Patch for voicemail date/time   Tilghman Lesher
• [Asterisk-Dev] Patch for voicemail date/time   Tilghman Lesher
• [Asterisk-Dev] Patch for voicemail date/time   Tilghman Lesher
• [Asterisk-Dev] strdupa in app_voicemail2.c   Tilghman Lesher
• [Asterisk-Dev] Patch for voicemail date/time   Tilghman Lesher
• [Asterisk-Dev] Patch for voicemail date/time   Tilghman Lesher
• [Asterisk-Dev] gastman hostname patch   Tilghman Lesher
• [Asterisk-Dev] gastman hostname patch   Tilghman Lesher
• [Asterisk-Dev] gastman hostname patch   Tilghman Lesher
• [Asterisk-Dev] Answering Machine Detection with WildCard?   Tilghman Lesher
• [Asterisk-Dev] IAX vs. IAX2 - problems and notes   Tilghman Lesher
• [Asterisk-Dev] Withdrawling from the list.....   Tilghman Lesher
• [Asterisk-Dev] More include fixes   Thorsten Lockert
• [Asterisk-Dev] strdupa in app_voicemail2.c   Thorsten Lockert
• [Asterisk-Dev] Resolver issues with ENUM on non-Linux systems   Thorsten Lockert
• [Asterisk-Dev] mkdep uses bash   Thorsten Lockert
• [Asterisk-Dev] strdupa in app_voicemail2.c   Thorsten Lockert
• [Asterisk-Dev] DTMF and Transfer Call Problem   Michael Manousos
• [Asterisk-Dev] DTMF and Transfer Call Problem   Michael Manousos
• [Asterisk-Dev] DTMF and Transfer Call Problem   Michael Manousos
• [Asterisk-Dev] DTMF and Transfer Call Problem   Michael Manousos
• [Asterisk-Dev] Some build fixes for h323 plugin   Michael Manousos
• [Asterisk-Dev] asterisk-oh323, new version 0.5.2   Michael Manousos
• [Asterisk-Dev] OH323-FAX   Michael Manousos
• [Asterisk-Dev] Using linux RTC with ztdummy   Brancaleoni Matteo
• [Asterisk-Dev] Phonecore package   Brancaleoni Matteo
• [Asterisk-Dev] IAX vs. IAX2 - problems and notes   Brancaleoni Matteo
• [Asterisk-Dev] App voicemail2 compile error   Brancaleoni Matteo
• [Asterisk-Dev] Bug in app_directory.c - bad hardcoded filename path   Brancaleoni Matteo
• [Asterisk-Dev] linking and libs fixes for chan_h323   Jeremy McNamara
• [Asterisk-Dev] compile error - cdr_mysql.so   Jeremy McNamara
• [Asterisk-Dev] IAX over 1394   Jeremy McNamara
• [Asterisk-Dev] Anyone doing QOS routing on Linux for SIP/RTP?   Jeremy McNamara
• [Asterisk-Dev] CAC Adit 600 FXO-8 Cards   Jeremy McNamara
• [Asterisk-Dev] Meetme changes   Jeremy McNamara
• [Asterisk-Dev] zapata, zaptel, GPL etc   Jeremy McNamara
• [Asterisk-Dev] zapata, zaptel, GPL etc   Jeremy McNamara
• [Asterisk-Dev] H323 segfault .... PPC   Jeremy McNamara
• [Asterisk-Dev] H323 segfault .... PPC   Jeremy McNamara
• [Asterisk-Dev] follow-on calls   Jeremy McNamara
• [Asterisk-Dev] follow-on calls   Jeremy McNamara
• [Asterisk-Dev] follow-on calls   Jeremy McNamara
• [Asterisk-Dev] chan_zap bug with hidecallerid?   Pauline Middelink
• [Asterisk-Dev] E1, crc-4 errors   Pauline Middelink
• [Asterisk-Dev] Audio Problem in CVS and new "immediate" pri behavior   Benjamin Miller
• [Asterisk-Dev] Configuration/Management   Filip Olsson
• [Asterisk-Dev] callerid toward X101P   Florian Overkamp
• [Asterisk-Dev] ${RDNIS} from SIP-GW   Florian Overkamp
• [Asterisk-Dev] Anyone doing QOS routing on Linux for SIP/RTP?   Florian Overkamp
• [Asterisk-Dev] chan_zap bug with hidecallerid?   Lorenzo Pallara
• [Asterisk-Dev] Hardware advice required   Juan Perez
• [Asterisk-Dev] Polarity Reversal ?   Jon Pounder
• [Asterisk-Dev] working mpg321 patch   Steven Pritchard
• [Asterisk-Dev] ALSA problems   Emanuele Pucciarelli
• [Asterisk-Dev] final mgcp patch   Karl Putland
• [Asterisk-Dev] More IAX tests with different codecs   Karl Putland
• [Asterisk-Dev] question about ast_queue_frame in channel.c   Karl Putland
• [Asterisk-Dev] final mgcp patch   Karl Putland
• [Asterisk-Dev] mgcp inband dtmf tweaks   Karl Putland
• [Asterisk-Dev] mgcp inband dtmf tweaks   Karl Putland
• [Asterisk-Dev] final mgcp patch   Karl Putland
• [Asterisk-Dev] Patch for Bug and memory leak in chan_iax[2].c dialplan caching   Karl Putland
• [Asterisk-Dev] chan_zap bug with hidecallerid?   Martin Pycko
• [Asterisk-Dev] chan_zap bug with hidecallerid?   Martin Pycko
• [Asterisk-Dev] E1, crc-4 errors   Martin Pycko
• [Asterisk-Dev] -- CPE does not support Call Waiting Caller*ID.   Martin Pycko
• [Asterisk-Dev] zapata, zaptel, GPL etc   Martin Pycko
• [Asterisk-Dev] Anyone doing QOS routing on Linux for SIP/RTP?   Steve Radich
• [Asterisk-Dev] DTMF and Transfer Call Problem   Sergio Serrano Revuelto
• [Asterisk-Dev] DTMF and Transfer Call Problem   Sergio Serrano Revuelto
• [Asterisk-Dev] DTMF and Transfer Call Problem   Sergio Serrano Revuelto
• [Asterisk-Dev] DTMF and Transfer Call Problem   Sergio Serrano Revuelto
• [Asterisk-Dev] FAX configuration   Sergio Serrano Revuelto
• [Asterisk-Dev] Any chance at Asterisk supporting Speex?   Sergio Serrano Revuelto
• [Asterisk-Dev] ${RDNIS} from SIP-GW   Adam Roach
• [Asterisk-Dev] Polarity Reversal ?   Vinod Sankar
• [Asterisk-Dev] Cable adapter for digium cards for use in data mode   Vinod Sankar
• [Asterisk-Dev] Polarity Reversal ?   Vinod Sankar
• [Asterisk-Dev] R2/DTMF on Asterisk? How???   Vinod Sankar
• [Asterisk-Dev] zapata, zaptel, GPL etc   Vinod Sankar
• [Asterisk-Dev] zapata, zaptel, GPL etc   Vinod Sankar
• [Asterisk-Dev] final mgcp patch   Tycho Schenkeveld
• [Asterisk-Dev] final mgcp patch   Tycho Schenkeveld
• [Asterisk-Dev] final mgcp patch   Tycho Schenkeveld
• [Asterisk-Dev] final mgcp patch   Tycho Schenkeveld
• [Asterisk-Dev] asterisk process killed by devlabel   Steven Smith
• [Asterisk-Dev] High performance on Network interface -   Steven Smith
• [Asterisk-Dev] enum.c -> HEADER ??   Mark Spencer
• [Asterisk-Dev] Using linux RTC with ztdummy   Mark Spencer
• [Asterisk-Dev] enum.c -> HEADER ??   Mark Spencer
• [Asterisk-Dev] Is ECHO_CAN_MARK3 i386 only?   Mark Spencer
• [Asterisk-Dev] final mgcp patch   Mark Spencer
• [Asterisk-Dev] question about ast_queue_frame in channel.c   Mark Spencer
• [Asterisk-Dev] mgcp inband dtmf tweaks   Mark Spencer
• [Asterisk-Dev] E1, crc-4 errors   Mark Spencer
• [Asterisk-Dev] E1, crc-4 errors   Mark Spencer
• [Asterisk-Dev] ${RDNIS} from SIP-GW   Mark Spencer
• [Asterisk-Dev] Patch: SIP bindaddr not always used   Mark Spencer
• [Asterisk-Dev] Bug in chan_sip.c and "realod"   Mark Spencer
• [Asterisk-Dev] Bug in SIP code - OPTIONS requests don't die   Mark Spencer
• [Asterisk-Dev] Meetme changes   Mark Spencer
• [Asterisk-Dev] Is ECHO_CAN_MARK3 i386 only?   Iain Stevenson
• [Asterisk-Dev] 'goodbye' in voicemail2 not quite there yet   Iain Stevenson
• [Asterisk-Dev] H323 segfault .... PPC   Iain Stevenson
• [Asterisk-Dev] H323 segfault .... PPC   Iain Stevenson
• [Asterisk-Dev] H.323 lack of progress report and a patch for minor bugs   Iain Stevenson
• [Asterisk-Dev] More IAX tests with different codecs   TC
• [Asterisk-Dev] Bug in chan_sip.c and "realod"   TC
• [Asterisk-Dev] zapata, zaptel, GPL etc   TC
• [Asterisk-Dev] Withdrawling from the list.....   Tener, Stuart B., IT3 , USNR-R
• [Asterisk-Dev] IAX vs. IAX2 - problems and notes   John Todd
• [Asterisk-Dev] More IAX tests with different codecs   John Todd
• [Asterisk-Dev] Anyone doing QOS routing on Linux for SIP/RTP?   John Todd
• [Asterisk-Dev] Anyone doing QOS routing on Linux for SIP/RTP?   John Todd
• [Asterisk-Dev] Extremely minor MeetMe bug   John Todd
• [Asterisk-Dev] Bug in SIP code - OPTIONS requests don't die   John Todd
• [Asterisk-Dev] Bug in app_directory.c - bad hardcoded filename path   John Todd
• [Asterisk-Dev] IAX vs. IAX2 - problems and notes   John Todd
• [Asterisk-Dev] depmod fails for zaptel   The Traveller
• [Asterisk-Dev] E1, crc-4 errors   The Traveller
• [Asterisk-Dev] E1, crc-4 errors   The Traveller
• [Asterisk-Dev] (patch) PlayDialtone   The Traveller
• [Asterisk-Dev] Aastra PT390 ADSI Issues (w/ Patch)   Jayson Vantuyl
• [Asterisk-Dev] chan_zap bug with hidecallerid?   Andrea Venturi
• [Asterisk-Dev] Bug Report - astman missing caller-id info   John Vozza
• [Asterisk-Dev] FW: hdlc link down errors   DUSTIN WILDES
• [Asterisk-Dev] Cable adapter for digium cards for use in data mode   Dave Weis
• [Asterisk-Dev] Anyone doing QOS routing on Linux for SIP/RTP?   Wade Weppler
• [Asterisk-Dev] ${RDNIS} from SIP-GW   Eric Wieling
• [Asterisk-Dev] Anyone doing QOS routing on Linux for SIP/RTP?   Eric Wieling
• [Asterisk-Dev] Anyone doing QOS routing on Linux for SIP/RTP?   Eric Wieling
• [Asterisk-Dev] Anyone doing QOS routing on Linux for SIP/RTP?   Adam Tauno Williams
• [Asterisk-Dev] Error parsing "response" in check_auth()   Wilhelm Wimmreuter
• [Asterisk-Dev] Meetme changes   Charles E. Youse
• [Asterisk-Dev] Using linux RTC with ztdummy   asterisk at billheckel.com
• [Asterisk-Dev] zapata, zaptel, GPL etc   asterisk at billheckel.com
• [Asterisk-Dev] R2/DTMF on Asterisk? How???   cziom
• [Asterisk-Dev] Patch: SIP bindaddr not always used   matt at overloaded.net
• [Asterisk-Dev] H245 tunneling / Q-SIG module   slime2k at t-online.de
• [Asterisk-Dev] Anyone doing QOS routing on Linux for SIP/RTP?   steve
• [Asterisk-Dev] Anyone doing QOS routing on Linux for SIP/RTP?   wasim at convergence.com.pk

Last message date: Sat May 31 23:14:36 MST 2003
Archived on: Tue Sep 5 14:26:41 MST 2006

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