November 2006 Archives by author

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Starting: Wed Nov 1 00:11:04 MST 2006
Ending: Thu Nov 30 22:12:45 MST 2006
Messages: 648

• [asterisk-dev] One way audio problems   Asbjørn Zweidorff Kjær
• [asterisk-dev] One way audio problems   Asbjørn Zweidorff Kjær
• [asterisk-dev] One way audio problems   Martin Vít
• [asterisk-dev] IAX2 very CPU hungry   Martin Vít
• [asterisk-dev] SQL_ESC func_odbc.c   Martin Vít
• [asterisk-dev] file: branch 1.2 r47711 - /branches/1.2/channels/chan_local.c   Martin Vít
• [asterisk-dev] sip cancel not working   Mario Fernández Alonso
• [asterisk-dev] Re: Release schedule ?   Tomislav Parčina
• [asterisk-dev] TDM400 Hangup problem (bug?)   Ale
• [asterisk-dev] TDM400 CPC signal   Ale
• [asterisk-dev] TDM400 Hangup problem (bug?)   Ale
• [asterisk-dev] TDM400 CPC signal   Ale
• [asterisk-dev] SIP Multiple endpoints with same id   Volkov Alexei
• [asterisk-dev] SIP ghost calls   Volkov Alexei
• [asterisk-dev] Re: How to busy out PRI channels?   Benny Amorsen
• [asterisk-dev] Re: SIP Multiple endpoints with same id   Benny Amorsen
• [asterisk-dev] Re: chan_sip - performance, and OPTIONS poking for 1000s of peers   Benny Amorsen
• [asterisk-dev] Re: chan_sip - performance, and OPTIONS poking for 1000s of peers   Benny Amorsen
• [asterisk-dev] IAX2 still broken   Anton
• [asterisk-dev] IAX2 still broken   Anton
• [asterisk-dev] IAX2 still broken   Anton
• [asterisk-dev] IAX2 still broken   Anton
• [asterisk-dev] chan_zap - weird behaviour on bridged data calls   Anton
• [asterisk-dev] latest SVN 1.4 chan_h323 compile problem   Anton
• [asterisk-dev] latest SVN 1.4 chan_h323 compile problem   Anton
• [asterisk-dev] Re: [asterisk-users] Re: IAX2 goes "one way audio" when lag gets bad   Anton
• [asterisk-dev] latest SVN 1.4 chan_h323 compile problem   Anton
• [asterisk-dev] HANGUPCAUSE is not passed over IAX2   Anton
• [asterisk-dev] HANGUPCAUSE is not passed over IAX2   Anton
• [asterisk-dev] RFC: internal CLI changes   Anton
• [asterisk-dev] asterisk 1.2.12.1 - segfault in IAX2   Anton
• [asterisk-dev] asterisk 1.2.12.1 - segfault in IAX2   Anton
• [asterisk-dev] RFC: internal CLI changes   Anton
• [asterisk-dev] RFC: internal CLI changes   Anton
• [asterisk-dev] RFC: internal CLI changes   Anton
• [asterisk-dev] IAX2 very CPU hungry   Anton
• [asterisk-dev] IAX2 very CPU hungry   Anton
• [asterisk-dev] IAX2 very CPU hungry   Anton
• [asterisk-dev] IAX2 very CPU hungry   Anton
• [asterisk-dev] SUGGESTION: Attended and blind transfer in the same digit   Anton
• [asterisk-dev] RFC: internal CLI changes   Bob Atkins
• [asterisk-dev] SIP ghost calls   Bob Atkins
• [asterisk-dev] Zaptel Transcoder issues (and possible conflict with PRI cards)   Sebastian Auriol
• [asterisk-dev] Problem with svn.digium.com?   Sebastian Auriol
• [asterisk-dev] Bug id 8233   Dan Austin
• [asterisk-dev] Public SVN borked?   Dan Austin
• [asterisk-dev] Public SVN borked?   Dan Austin
• [asterisk-dev] Re: Help with 240 samples on frames read fromchan_iax   Dan Austin
• [asterisk-dev] Re: Help with 240 samples on frames read fromchan_iax   Dan Austin
• [asterisk-dev] Re: Help with 240 samples on frames readfromchan_iax   Dan Austin
• [asterisk-dev] Re: Help with 240 samples on framesreadfromchan_iax   Dan Austin
• [asterisk-dev] T38 woes, possible answer   Dan Austin
• [asterisk-dev] T38 woes, possible answer   Dan Austin
• [asterisk-dev] Release schedule ?   Dan Austin
• [asterisk-dev] 1.4 open issues (Major and Crash)   Dan Austin
• [asterisk-dev] Release schedule ?   Dan Austin
• [asterisk-dev] understanding ast_codec_get_samples for AMR   Dan Austin
• [asterisk-dev] More T38 questions/fixes   Dan Austin
• [asterisk-dev] More T38 questions/fixes   Dan Austin
• [asterisk-dev] More T38 questions/fixes   Dan Austin
• [asterisk-dev] TLS/SSL futures   Michiel van Baak
• [asterisk-dev] Donwload Branch with Mac OSX   Michiel van Baak
• [asterisk-dev] List main headers changed?   Michiel van Baak
• [asterisk-dev] Donwload Branch with Mac OSX   Mihai Balea
• [asterisk-dev] Video in chan_iax is broken...   Mihai Balea
• [asterisk-dev] Re: [Asterisk-video] Report from Video Task Force in Paris!   Mihai Balea
• [asterisk-dev] VM breakage in trunk r47021   David Bath
• *****SPAM***** [asterisk-dev] dynamic queue members get stuck with status 'on hold'   Dean Bath
• *****SPAM***** RE: [asterisk-dev] dynamic queue members get stuck with status 'onhold'   Dean Bath
• [asterisk-dev] SIP Multiple endpoints with same id   Peter Beckman
• [asterisk-dev] RFC: internal CLI changes   Peter Beckman
• [asterisk-dev] bug 7506 -- This may be a life-changing experience to channel programmers! Please read!   Nic Bellamy
• [asterisk-dev] Clearing pick-up groups on Zap/ channels   Nic Bellamy
• [asterisk-dev] 1.4 open issues (Major and Crash)   Nic Bellamy
• [asterisk-dev] file: branch 1.2 r47711 - /branches/1.2/channels/chan_local.c   Nic Bellamy
• [asterisk-dev] Re: [asterisk-commits] russell: trunk r48019 - in /trunk/include/asterisk: threadstorage.h utils.h   Nic Bellamy
• [asterisk-dev] chanspy error message - 1.2.13   Dov Bigio
• [asterisk-dev] chan_sip3   Marc Blanchet
• [asterisk-dev] cli changes parser   Marc Blanchet
• [asterisk-dev] Need your help to use TDM(Hardware) in Asterisk   Bob
• [asterisk-dev] IAX2 still broken   Greg Boehnlein
• [asterisk-dev] IAX2 still broken   Greg Boehnlein
• [asterisk-dev] Bug marshals - won't you take a look at 8188   Greg Boehnlein
• [asterisk-dev] Asterisk 1.4b3 crashing consistantly - How do I provide helpful info?   Greg Boehnlein
• [asterisk-dev] Asterisk 1.4b3 crashing consistantly - How do I provide helpful info?   Greg Boehnlein
• [asterisk-dev] Release schedule ?   Greg Boehnlein
• [asterisk-dev] Release schedule ?   Greg Boehnlein
• [asterisk-dev] Development :: Staying busy, please have patience   Greg Boehnlein
• [asterisk-dev] Development :: Staying busy, please have patience   Gregory Boehnlein
• [asterisk-dev] SIP Multiple endpoints with same id   Leo Ann Boon
• [asterisk-dev] SIP Multiple endpoints with same id   Leo Ann Boon
• [asterisk-dev] SIP Multiple endpoints with same id   Bradley
• [asterisk-dev] Queue callback   Alex Brecher
• [asterisk-dev] SUGGESTION: Attended and blind transfer in the same digit   Paco Brufal
• [asterisk-dev] IAX2 still broken   Russell Bryant
• [asterisk-dev] Re: [svn-commits] murf: trunk r46801 - /trunk/res/res_config_odbc.c   Russell Bryant
• [asterisk-dev] VM breakage in trunk r47021   Russell Bryant
• [asterisk-dev] Music on hold deafens in 1.4b3   Russell Bryant
• [asterisk-dev] bug 7506 -- This may be a life-changing experience to channel programmers! Please read!   Russell Bryant
• [asterisk-dev] Music on hold deafens in 1.4b3   Russell Bryant
• [asterisk-dev] <huge>aclocal.m4</huge> ???   Russell Bryant
• [asterisk-dev] delayed responses from -dev list ?   Russell Bryant
• [asterisk-dev] Why -lnsl in IKSEMEL_LIB ?   Russell Bryant
• [asterisk-dev] Re: [svn-commits] kpfleming: trunk r47257 - /trunk/   Russell Bryant
• [asterisk-dev] <huge>aclocal.m4</huge> ???   Russell Bryant
• [asterisk-dev] bug_7506 changes for 1.4, trunk   Russell Bryant
• [asterisk-dev] Who can explain this? chanzap zt_fixup / zt_indicate   Russell Bryant
• [asterisk-dev] <huge>aclocal.m4</huge> ???   Russell Bryant
• [asterisk-dev] Asterisk 1.4b3 crashing consistantly - How do I provide helpful info?   Russell Bryant
• [asterisk-dev] DUNDi transaction ID generation   Russell Bryant
• [asterisk-dev] Release schedule ?   Russell Bryant
• [asterisk-dev] Development :: Staying busy, please have patience   Russell Bryant
• [asterisk-dev] Development :: Staying busy, please have patience   Russell Bryant
• [asterisk-dev] chan_sip.c - struct sip_pvt is very large...   Russell Bryant
• [asterisk-dev] Voicemail argument list limit   Russell Bryant
• [asterisk-dev] Re: [svn-commits] file: trunk r47560 - /trunk/apps/app_meetme.c   Russell Bryant
• [asterisk-dev] (mis)use of threadstorage.h ?   Russell Bryant
• [asterisk-dev] Re: [svn-commits] rizzo: trunk r48084 - /trunk/main/utils.c   Russell Bryant
• [asterisk-dev] better home for ast_dynamic_str and friends ?   Russell Bryant
• [asterisk-dev] Call numbers in IAX2 media frames   Russell Bryant
• [asterisk-dev] Call numbers in IAX2 media frames   Russell Bryant
• [asterisk-dev] Call numbers in IAX2 media frames   Russell Bryant
• [asterisk-dev] Re: [asterisk-commits] russell: trunk r48019 - in /trunk/include/asterisk: threadstorage.h utils.h   Russell Bryant
• [asterisk-dev] Asterisk - GUI   Dome C.
• [asterisk-dev] Sorry for post this message to this list   Dome C.
• [asterisk-dev] In call Codec parameterization in 1.6 ?   Paul Cadach
• [asterisk-dev] Re: why 'o' (preserve original callerid) is notdefault in app_dial.c ?   Paul Cadach
• [asterisk-dev] setting the q.931 "reason for redirecting"?   Paul Cadach
• [asterisk-dev] Voicemail redisign   Brian Capouch
• [asterisk-dev] RFC: internal CLI changes   Brian Capouch
• [asterisk-dev] app_realtime vs. func_realtime   Brian Capouch
• [asterisk-dev] Asterisk 1.2.13 under FC6 - chan_phone.c wants compiler.h   Juan Carlos Castro y Castro
• [asterisk-dev] ISDN gurus - getting isdn call progress ??   Christian
• [asterisk-dev] Adding AMR codec   James Cloos
• [asterisk-dev] Voicemail redisign   Tzafrir Cohen
• [asterisk-dev] Re: [svn-commits] murf: trunk r46801 - /trunk/res/res_config_odbc.c   Tzafrir Cohen
• [asterisk-dev] Zaptel Transcoder issues (and possible conflict with PRI cards)   Tzafrir Cohen
• [asterisk-dev] Zaptel Transcoder issues (and possible conflict with PRI cards)   Tzafrir Cohen
• [asterisk-dev] repost: fixes to zaptel makefile   Tzafrir Cohen
• [asterisk-dev] repost: fixes to zaptel makefile   Tzafrir Cohen
• [asterisk-dev] Clearing pick-up groups on Zap/ channels   Tzafrir Cohen
• [asterisk-dev] TLS/SSL futures   Tzafrir Cohen
• [asterisk-dev] <huge>aclocal.m4</huge> ???   Tzafrir Cohen
• [asterisk-dev] Re: [svn-commits] kpfleming: trunk r47257 - /trunk/   Tzafrir Cohen
• [asterisk-dev] Clearing pick-up groups on Zap/ channels   Tzafrir Cohen
• [asterisk-dev] Clearing pick-up groups on Zap/ channels   Tzafrir Cohen
• [asterisk-dev] optional time formatting   Tzafrir Cohen
• [asterisk-dev] Clearing pick-up groups on Zap/ channels   Tzafrir Cohen
• [asterisk-dev] configuration of chan_zap   Tzafrir Cohen
• [asterisk-dev] Weird errors returned to chan_zap by zaptel   Tzafrir Cohen
• [asterisk-dev] SIP Multiple endpoints with same id   Tzafrir Cohen
• [asterisk-dev] TDM400 Hangup problem (bug?)   Tzafrir Cohen
• [asterisk-dev] Re: Weird errors returned to chan_zap by zaptel   Tzafrir Cohen
• [asterisk-dev] Re: Weird errors returned to chan_zap by zaptel   Tzafrir Cohen
• [asterisk-dev] RFC: internal CLI changes   Tzafrir Cohen
• [asterisk-dev] Need help in how to originate call from c to asterisk ie controlling asterisk cli commands through c.   Tzafrir Cohen
• [asterisk-dev] Release schedule ?   Tzafrir Cohen
• [asterisk-dev] zaptel helper script: the branch   Tzafrir Cohen
• [asterisk-dev] zaptel helper script: the branch   Tzafrir Cohen
• [asterisk-dev] zaptel helper script: the branch   Tzafrir Cohen
• [asterisk-dev] cli changes parser   Tzafrir Cohen
• [asterisk-dev] Neat Application for Text to Speech   Dean Collins
• [asterisk-dev] Microsoft ring cam or round table   Dean Collins
• [asterisk-dev] Sorry for post this message to this list   Dean Collins
• [asterisk-dev] Re: [zaptel-commits] file: trunk r1570 - /trunk/wct4xxp/Kbuild   Joshua Colp
• [asterisk-dev] latest SVN 1.4 chan_h323 compile problem   Joshua Colp
• [asterisk-dev] chan_h323 compile problem in SVN HEAD   Joshua Colp
• [asterisk-dev] SIP Peer matching on Inbound calls (and a feature request, please)   Daniel Corbe
• [asterisk-dev] Re: Help with 240 samples on frames read fromchan_iax   Steven Critchfield
• [asterisk-dev] Asterisk memory usage   Steven Critchfield
• [asterisk-dev] understanding ast_codec_get_samples for AMR   Steven Critchfield
• [asterisk-dev] Re: How to busy out PRI channels?   Alistair Cunningham
• [asterisk-dev] IAX2 still broken   Loic DIDELOT
• [asterisk-dev] IAX2 still broken   Loic DIDELOT
• [asterisk-dev] IAX2 very CPU hungry   Loic DIDELOT
• [asterisk-dev] Support for TCP in asterisk   Loic DIDELOT
• [asterisk-dev] Support for TCP in asterisk   Loic DIDELOT
• [asterisk-dev] Support for TCP in asterisk   Loic DIDELOT
• [asterisk-dev] Support for TCP in asterisk   Loic DIDELOT
• [asterisk-dev] TLS/SSL futures   Klaus Darilion
• [asterisk-dev] TLS/SSL futures   Klaus Darilion
• [asterisk-dev] Support for TCP in asterisk   Klaus Darilion
• [asterisk-dev] RFC - TLS certificate negotiation   Klaus Darilion
• [asterisk-dev] RFC - TLS certificate negotiation   Klaus Darilion
• [asterisk-dev] SIP ghost calls   Klaus Darilion
• [asterisk-dev] SIP ghost calls   Klaus Darilion
• [asterisk-dev] Bug marshals - won't you take a look at 8188   Stephen Davies
• [asterisk-dev] Re: why 'o' (preserve original callerid) is not default in app_dial.c ?   Stephen Davies
• [asterisk-dev] Bug marshals - won't you take a look at 8188   Stephen Davies
• [asterisk-dev] understanding ast_codec_get_samples for AMR   Stephen Davies
• [asterisk-dev] channel_walk_locked possible bug / possible fix?   Steve Davies
• [asterisk-dev] channel_walk_locked possible bug / possible fix?   Steve Davies
• [asterisk-dev] channel_walk_locked possible bug / possible fix?   Steve Davies
• [asterisk-dev] channel_walk_locked possible bug / possible fix?   Steve Davies
• [asterisk-dev] ERROR message that is not an error?   Steve Davies
• [asterisk-dev] ERROR message that is not an error?   Steve Davies
• [asterisk-dev] Clearing pick-up groups on Zap/ channels   Rod Dorman
• [asterisk-dev] repost: fixes to zaptel makefile   Rod Dorman
• [asterisk-dev] Asterisk 1.4 and Queues RealTime   Gregory Duchatelet
• [asterisk-dev] Release schedule ?   Gregory Duchatelet
• [asterisk-dev] Bug id 8233   Johansson Olle E
• [asterisk-dev] Clearing pick-up groups on Zap/ channels   Johansson Olle E
• [asterisk-dev] Re: SIP Multiple endpoints with same id   Johansson Olle E
• [asterisk-dev] possible memory leak in chan_sip.c ?   Johansson Olle E
• [asterisk-dev] Re: Fwd: Asterisk and Max TNT PRI to SIP Authentication Issue, a little closer   Johansson Olle E
• [asterisk-dev] T38 woes, possible answer   Johansson Olle E
• [asterisk-dev] Unable to make SIP calls   Johansson Olle E
• [asterisk-dev] Release schedule ?   Johansson Olle E
• [asterisk-dev] Release schedule ?   Johansson Olle E
• [asterisk-dev] A chan_sip weekend for 1.2 and 1.4 trunks plus svn trunk   Johansson Olle E
• [asterisk-dev] chan_sip.c - struct sip_pvt is very large...   Johansson Olle E
• [asterisk-dev] sip cancel not working   Johansson Olle E
• [asterisk-dev] chan_sip.c - struct sip_pvt is very large...   Johansson Olle E
• [asterisk-dev] backports site   Johansson Olle E
• [asterisk-dev] backports site   Johansson Olle E
• [asterisk-dev] file: branch 1.2 r47711 - /branches/1.2/channels/chan_local.c   Johansson Olle E
• [asterisk-dev] Support for TCP in asterisk   Johansson Olle E
• [asterisk-dev] chan_sip3   Johansson Olle E
• [asterisk-dev] More T38 questions/fixes   Johansson Olle E
• [asterisk-dev] More T38 questions/fixes   Johansson Olle E
• [asterisk-dev] ERROR message that is not an error?   Johansson Olle E
• [asterisk-dev] IAX2 very CPU hungry   Johansson Olle E
• [asterisk-dev] chan_sip - performance, and OPTIONS poking for 1000s of peers   Johansson Olle E
• [asterisk-dev] Re: chan_sip - performance, and OPTIONS poking for 1000s of peers   Johansson Olle E
• [asterisk-dev] Replacement RTPd?   Johansson Olle E
• [asterisk-dev] Problem with svn.digium.com?   Johansson Olle E
• [asterisk-dev] RFC - TLS certificate negotiation   Johansson Olle E
• [asterisk-dev] RFC - TLS certificate negotiation   Johansson Olle E
• [asterisk-dev] chan_sip - performance, and OPTIONS poking for 1000s of peers   Johansson Olle E
• [asterisk-dev] SIP ghost calls   Johansson Olle E
• [asterisk-dev] SIP ghost calls   Johansson Olle E
• [asterisk-dev] RFC: internal CLI changes   Steve Edwards
• [asterisk-dev] Pthread wrapper updates   SF Markus Elfring
• [asterisk-dev] Pthread wrapper updates   SF Markus Elfring
• [asterisk-dev] Pthread wrapper updates   SF Markus Elfring
• [asterisk-dev] expression checking implementation update   SF Markus Elfring
• [asterisk-dev] function ast_merge_contexts_and_delete   Garavaglia Fabio
• [asterisk-dev] Invalid conference number   Song Zhi Feng
• [asterisk-dev] MeetMe and silence suppression   ISAC Flavius
• [asterisk-dev] Strange code in chan_zap.c ?   Kevin P. Fleming
• [asterisk-dev] Music on hold deafens in 1.4b3   Kevin P. Fleming
• [asterisk-dev] Re: why 'o' (preserve original callerid) is notdefault in app_dial.c ?   Kevin P. Fleming
• [asterisk-dev] Re: why 'o' (preserve original callerid) is notdefault in app_dial.c ?   Kevin P. Fleming
• [asterisk-dev] Music on hold deafens in 1.4b3   Kevin P. Fleming
• [asterisk-dev] Re: [asterisk-commits] rizzo: trunk r47132 - in /trunk: include/asterisk/manager.h main/manager.c   Kevin P. Fleming
• [asterisk-dev] Re: [asterisk-commits] rizzo: trunk r47132 - in /trunk: include/asterisk/manager.h main/manager.c   Kevin P. Fleming
• [asterisk-dev] Public SVN borked?   Kevin P. Fleming
• [asterisk-dev] Zaptel Transcoder issues (and possible conflict with PRI cards)   Kevin P. Fleming
• [asterisk-dev] Subversion repository maintenance and upgrades are completed   Kevin P. Fleming
• [asterisk-dev] repost: fixes to zaptel makefile   Kevin P. Fleming
• [asterisk-dev] Re: [asterisk-commits] oej: trunk r47194 - /trunk/channels/chan_sip.c   Kevin P. Fleming
• [asterisk-dev] <huge>aclocal.m4</huge> ???   Kevin P. Fleming
• [asterisk-dev] TLS/SSL futures   Kevin P. Fleming
• [asterisk-dev] Subversion repository maintenance and upgrades are completed   Kevin P. Fleming
• [asterisk-dev] Re: [svn-commits] kpfleming: trunk r47257 - /trunk/   Kevin P. Fleming
• [asterisk-dev] repost: fixes to zaptel makefile   Kevin P. Fleming
• [asterisk-dev] delayed responses from -dev list ?   Kevin P. Fleming
• [asterisk-dev] Clearing pick-up groups on Zap/ channels   Kevin P. Fleming
• [asterisk-dev] Why -lnsl in IKSEMEL_LIB ?   Kevin P. Fleming
• [asterisk-dev] <huge>aclocal.m4</huge> ???   Kevin P. Fleming
• [asterisk-dev] <huge>aclocal.m4</huge> ???   Kevin P. Fleming
• [asterisk-dev] <huge>aclocal.m4</huge> ???   Kevin P. Fleming
• [asterisk-dev] Re: [zaptel-commits] file: trunk r1570 - /trunk/wct4xxp/Kbuild   Kevin P. Fleming
• [asterisk-dev] Clearing pick-up groups on Zap/ channels   Kevin P. Fleming
• [asterisk-dev] Why -lnsl in IKSEMEL_LIB ?   Kevin P. Fleming
• [asterisk-dev] <huge>aclocal.m4</huge> ???   Kevin P. Fleming
• [asterisk-dev] SIP Multiple endpoints with same id   Kevin P. Fleming
• [asterisk-dev] Why -lnsl in IKSEMEL_LIB ?   Kevin P. Fleming
• [asterisk-dev] SIP Multiple endpoints with same id   Kevin P. Fleming
• [asterisk-dev] Asterisk 1.4b3 crashing consistantly - How do I provide helpful info?   Kevin P. Fleming
• [asterisk-dev] SIP Multiple endpoints with same id   Kevin P. Fleming
• [asterisk-dev] Synchronized queue between threads   Kevin P. Fleming
• [asterisk-dev] TLS/SSL futures   Kevin P. Fleming
• [asterisk-dev] possible memory leak in chan_sip.c ?   Kevin P. Fleming
• [asterisk-dev] possible memory leak in chan_sip.c ?   Kevin P. Fleming
• [asterisk-dev] possible memory leak in chan_sip.c ?   Kevin P. Fleming
• [asterisk-dev] Re: Asterisk and Max TNT SIP Authentication Issue, WORKING   Kevin P. Fleming
• [asterisk-dev] RFC: internal CLI changes   Kevin P. Fleming
• [asterisk-dev] ISDN gurus - getting isdn call progress ??   Kevin P. Fleming
• [asterisk-dev] Release schedule ?   Kevin P. Fleming
• [asterisk-dev] Release schedule ?   Kevin P. Fleming
• [asterisk-dev] Re: Release schedule ?   Kevin P. Fleming
• [asterisk-dev] Release schedule ?   Kevin P. Fleming
• [asterisk-dev] Release schedule ?   Kevin P. Fleming
• [asterisk-dev] Re: [asterisk-commits] rizzo: trunk r47652 - in /trunk: include/asterisk/cli.h main/cli.c   Kevin P. Fleming
• [asterisk-dev] RFC: internal CLI changes   Kevin P. Fleming
• [asterisk-dev] RFC: internal CLI changes   Kevin P. Fleming
• [asterisk-dev] channel_walk_locked possible bug / possible fix?   Kevin P. Fleming
• [asterisk-dev] channel_walk_locked possible bug / possible fix?   Kevin P. Fleming
• [asterisk-dev] file: branch 1.2 r47711 - /branches/1.2/channels/chan_local.c   Kevin P. Fleming
• [asterisk-dev] channel_walk_locked possible bug / possible fix?   Kevin P. Fleming
• [asterisk-dev] 1.4 svn configure looks to Qt in wrong Fedora places   Kevin P. Fleming
• [asterisk-dev] translate.c:88 powerof: Powerof 0: No power??   Kevin P. Fleming
• [asterisk-dev] Re: [asterisk-commits] russell: trunk r48019 - in /trunk/include/asterisk: threadstorage.h utils.h   Kevin P. Fleming
• [asterisk-dev] Re: [asterisk-commits] russell: trunk r48019 - in /trunk/include/asterisk: threadstorage.h utils.h   Kevin P. Fleming
• [asterisk-dev] Re: [asterisk-commits] russell: trunk r48019 - in /trunk/include/asterisk: threadstorage.h utils.h   Kevin P. Fleming
• [asterisk-dev] Release schedule ?   Kevin P. Fleming
• [asterisk-dev] Problem with svn.digium.com?   Kevin P. Fleming
• [asterisk-dev] app_dictate and AMI   Kevin P. Fleming
• [asterisk-dev] Call numbers in IAX2 media frames   Kevin P. Fleming
• [asterisk-dev] extend IAX2 IE proposal   Kevin P. Fleming
• [asterisk-dev] Clearing pick-up groups on Zap/ channels   Matthew Fredrickson
• [asterisk-dev] Re: Weird errors returned to chan_zap by zaptel   Matthew Fredrickson
• [asterisk-dev] ISDN gurus - getting isdn call progress ??   Matthew Fredrickson
• [asterisk-dev] Queues and Flash/SendDTMF in hybrid PBX   Matthew Fredrickson
• RESOLVED: [asterisk-dev] Fedora Core 6 (FC6) and Asterisk-1.2.13 andZaptel-1.2.10 compile problems   Michael J. Tubby B.Sc. G8TIC
• [asterisk-dev] Who can explain this? chanzap zt_fixup / zt_indicate   Paulo Garcia
• [asterisk-dev] Asterisk memory usage   Paulo Garcia
• [asterisk-dev] Asterisk memory usage   Paulo Garcia
• [asterisk-dev] TLS/SSL futures   Scott Gifford
• [asterisk-dev] flash transfer problem in asterisk integration with old PBX   Andrea Giuliani
• [asterisk-dev] Adding AMR codec   Ian Hailey
• [asterisk-dev] understanding ast_codec_get_samples for AMR   Ian Hailey
• [asterisk-dev] Re: How to busy out PRI channels?   Freddi Hansen
• [asterisk-dev] Code Support request for IAX provider sending registration refreshes of 0   Paul Hewlett
• [asterisk-dev] Code Support request for IAX provider sending registration refreshes of 0   Paul Hewlett
• [asterisk-dev] Code Support request for IAX provider sending registration refreshes of 0   Paul Hewlett
• [asterisk-dev] ERROR message that is not an error?   Paul Hewlett
• [asterisk-dev] App_SMS: UDH (EMS) bug and fix   Anselm Martin Hoffmeister
• [asterisk-dev] Re: App_SMS: UDH (EMS) bug and fix   Anselm Martin Hoffmeister
• [asterisk-dev] spandsp's T.38 gateway in Asterisk yet?   Henning Holtschneider
• [asterisk-dev] Digium Card - E1 signalling give weird error   Cheow Han Hoong
• [asterisk-dev] Re: CDR-csv records   Mark Hulber
• [asterisk-dev] CDR-csv records   Mark Hulber
• [asterisk-dev] app_dictate and AMI   F. Mitchell Felling II
• [asterisk-dev] app_dictate and AMI   F. Mitchell Felling II
• [asterisk-dev] Dialplan code to simulate calls   Diego Iastrubni
• [asterisk-dev] Queues and Flash/SendDTMF in hybrid PBX   Vieri (Antonio in Istambul)
• [asterisk-dev] Re: [asterisk-users] Re: IAX2 goes "one way audio" when lag gets bad   Pavel Jezek
• [asterisk-dev] latest SVN 1.4 chan_h323 compile problem   Pavel Jezek
• [asterisk-dev] oh323 faststart with cisco gateways question   Pavel Jezek
• Fw: [asterisk-dev] oh323 faststart with cisco gateways   Pavel Jezek
• [asterisk-dev] TLS/SSL futures   Olle E Johansson
• [asterisk-dev] Clearing pick-up groups on Zap/ channels   Olle E Johansson
• [asterisk-dev] Development :: Staying busy, please have patience   Olle E Johansson
• [asterisk-dev] ast_channel_masquerade fails   Olle E Johansson
• [asterisk-dev] Video in chan_iax is broken...   Olle E Johansson
• [asterisk-dev] Report from Video Task Force in Paris!   Olle E Johansson
• [asterisk-dev] understanding ast_codec_get_samples for AMR   Steve Kann
• [asterisk-dev] IAX2 very CPU hungry   Steve Kann
• [asterisk-dev] DO_CRASH question   Roy Sigurd Karlsbakk
• [asterisk-dev] setting the q.931 "reason for redirecting"?   Roy Sigurd Karlsbakk
• [asterisk-dev] setting the q.931 "reason for redirecting"?   Roy Sigurd Karlsbakk
• [asterisk-dev] setting the q.931 "reason for redirecting"?   Roy Sigurd Karlsbakk
• [asterisk-dev] setting the q.931 "reason for redirecting"?   Roy Sigurd Karlsbakk
• [asterisk-dev] setting the q.931 "reason for redirecting"?   Roy Sigurd Karlsbakk
• [asterisk-dev] RDNIS not set on diverted incoming calls?   Roy Sigurd Karlsbakk
• [asterisk-dev] setting the q.931 "reason for redirecting"?   Roy Sigurd Karlsbakk
• [asterisk-dev] setting the q.931 "reason for redirecting"?   Roy Sigurd Karlsbakk
• [asterisk-dev] Release schedule ?   Roy Sigurd Karlsbakk
• [asterisk-dev] backports site   Roy Sigurd Karlsbakk
• [asterisk-dev] Release schedule ?   Roy Sigurd Karlsbakk
• [asterisk-dev] error setting the RDNIS   Roy Sigurd Karlsbakk
• [asterisk-dev] error setting the RDNIS   Roy Sigurd Karlsbakk
• [asterisk-dev] developer documentation   Faruk Kasumovic
• [asterisk-dev] ISDN gurus - getting isdn call progress ??   Steve Kennedy
• [asterisk-dev] ISDN gurus - getting isdn call progress ??   Steve Kennedy
• [asterisk-dev] Adding AMR codec   Steve Kennedy
• [asterisk-dev] Voicemail redisign   Kristian Kielhofner
• [asterisk-dev] Voicemail redisign   Kristian Kielhofner
• [asterisk-dev] Voicemail redisign   Kristian Kielhofner
• [asterisk-dev] Voicemail redisign   Kristian Kielhofner
• [asterisk-dev] Support for TCP in asterisk   Satya Kishore
• [asterisk-dev] Clearing pick-up groups on Zap/ channels   Gil Kloepfer
• [asterisk-dev] why 'o' (preserve original callerid) is not defaultin app_dial.c ?   Koopmann, Jan-Peter
• [asterisk-dev] auto hangup problem   Kaloyan Kovachev
• [asterisk-dev] Ringback tones in asterisk?   Kaloyan Kovachev
• [asterisk-dev] Moh stops immediately   Kaloyan Kovachev
• [asterisk-dev] Re: why 'o' (preserve original callerid) is not default in app_dial.c ?   John Lange
• [asterisk-dev] Voicemail redisign   John Lange
• [asterisk-dev] Music on hold deafens in 1.4b3   John Lange
• [asterisk-dev] Re: why 'o' (preserve original callerid) is not default in app_dial.c ?   John Lange
• [asterisk-dev] Music on hold deafens in 1.4b3   John Lange
• [asterisk-dev] Music on hold deafens in 1.4b3   John Lange
• [asterisk-dev] Music on hold deafens in 1.4b3   John Lange
• [asterisk-dev] Music on hold deafens in 1.4b3   John Lange
• [asterisk-dev] Asterisk 1.4b3 crashing consistantly - How do I provide helpful info?   John Lange
• [asterisk-dev] Asterisk 1.4b3 crashing consistantly - How do I provide helpful info?   John Lange
• [asterisk-dev] SIP Multiple endpoints with same id   John Lange
• [asterisk-dev] SIP Multiple endpoints with same id   John Lange
• [asterisk-dev] SIP Multiple endpoints with same id   John Lange
• [asterisk-dev] Asterisk 1.4b3 crashing consistantly - How do I provide helpful info?   John Lange
• [asterisk-dev] Release schedule ?   John Lange
• [asterisk-dev] List main headers changed?   John Lange
• [asterisk-dev] zaptel helper script: the branch   John Lange
• [asterisk-dev] SIP Peer matching on Inbound calls (and a featurerequest, please)   Steve Langstaff
• [asterisk-dev] SIP NOTIFY routing problem   Steve Langstaff
• [asterisk-dev] chan_sip - performance, and OPTIONS poking for 1000s of peers   Steve Langstaff
• [asterisk-dev] Development :: Staying busy, please have patience   Andrew Latham
• [asterisk-dev] Voicemail redisign   Chris Lee
• [asterisk-dev] asterisk disappeared! any thoughts?   Tilghman Lesher
• [asterisk-dev] Music on hold deafens in 1.4b3   Tilghman Lesher
• [asterisk-dev] DO_CRASH question   Tilghman Lesher
• [asterisk-dev] Re: Help with 240 samples on frames read from chan_iax   Tilghman Lesher
• [asterisk-dev] Re: Help with 240 samples on frames read fromchan_iax   Tilghman Lesher
• [asterisk-dev] optional time formatting   Tilghman Lesher
• [asterisk-dev] Re: Weird errors returned to chan_zap by zaptel   Tilghman Lesher
• [asterisk-dev] TDM400 CPC signal   Tilghman Lesher
• [asterisk-dev] SQL_ESC func_odbc.c   Tilghman Lesher
• [asterisk-dev] comments on this one ? (macros for parsing config file ?)   Tilghman Lesher
• [asterisk-dev] One way audio problems   Tilghman Lesher
• [asterisk-dev] CUT() default argument problem   Tilghman Lesher
• [asterisk-dev] RFC: internal CLI changes   Tilghman Lesher
• [asterisk-dev] RFC: internal CLI changes   Tilghman Lesher
• [asterisk-dev] app_realtime vs. func_realtime   Tilghman Lesher
• [asterisk-dev] Need to support about Asterisk Manager API   Ly, Phung My
• [asterisk-dev] Need help in how to get notified of events from client   Ly, Phung My
• [asterisk-dev] ISDN gurus - getting isdn call progress ??   Julian Lyndon-Smith
• [asterisk-dev] Release schedule ?   Leif Madsen
• [asterisk-dev] Release schedule ?   Leif Madsen
• [asterisk-dev] Re: [svn-commits] file: trunk r47560 - /trunk/apps/app_meetme.c   Leif Madsen
• [asterisk-dev] Re: chan_sip - performance, and OPTIONS poking for 1000s of peers   Leif Madsen
• [asterisk-dev] chan_sip - performance, and OPTIONS poking for 1000s of peers   Leif Madsen
• [asterisk-dev] Asterisk 1.2.13 under FC6 - chan_phone.c wants compiler.h   Leif Madsen
• [asterisk-dev] SIP Multiple endpoints with same id   Mikael Magnusson
• [asterisk-dev] Support for TCP in asterisk   Mikael Magnusson
• [asterisk-dev] Celliax LiveCD 0.0.32 released (let Asterisk manage GSM and CDMA cellular phones, and Skype calls to/from cellphones, via chan_celliax)   Giovanni Maruzzelli
• [asterisk-dev] Re: How to busy out PRI channels?   Josh McAllister
• [asterisk-dev] Ringback tones in asterisk?   Sherwood McGowan
• [asterisk-dev] IAX2 still broken   Jeremy McNamara
• [asterisk-dev] Code Support request for IAX provider sending registration refreshes of 0   Jeremy McNamara
• [asterisk-dev] Need your help to use TDM(Hardware) in Asterisk   Khelik Mikhail
• [asterisk-dev] Need your help to use TDM(Hardware) in Asterisk   Khelik Mikhail
• [asterisk-dev] ISDN gurus - getting isdn call progress ??   Kai Militzer
• [asterisk-dev] RFC: internal CLI changes   Oriz L. Miller
• [asterisk-dev] RFC: internal CLI changes   Oriz L. Miller
• [asterisk-dev] chan_sip - performance, and OPTIONS poking for 1000s of peers   Ryan Mitchell
• [asterisk-dev] Re: chan_sip - performance, and OPTIONS poking for 1000s of peers   Ryan Mitchell
• [asterisk-dev] chan_sip - performance, and OPTIONS poking for 1000s of peers   Ryan Mitchell
• [asterisk-dev] chan_sip - performance, and OPTIONS poking for 1000s of peers   Ryan Mitchell
• [asterisk-dev] Re: How to busy out PRI channels?   Tony Mountifield
• [asterisk-dev] Re: why 'o' (preserve original callerid) is not default in app_dial.c ?   Tony Mountifield
• [asterisk-dev] Re: oej: branch 1.2 r46899 - /branches/1.2/channels/chan_sip.c   Tony Mountifield
• [asterisk-dev] Re: Help with 240 samples on frames read from chan_iax   Tony Mountifield
• [asterisk-dev] Re: Help with 240 samples on frames read from chan_iax   Tony Mountifield
• [asterisk-dev] Weird errors returned to chan_zap by zaptel   Tony Mountifield
• [asterisk-dev] Re: Weird errors returned to chan_zap by zaptel   Tony Mountifield
• [asterisk-dev] Re: Weird errors returned to chan_zap by zaptel   Tony Mountifield
• [asterisk-dev] Re: Weird errors returned to chan_zap by zaptel   Tony Mountifield
• [asterisk-dev] Re: Release schedule ?   Tony Mountifield
• [asterisk-dev] Re: Weird errors returned to chan_zap by zaptel   Tony Mountifield
• [asterisk-dev] Re: 1.4 open issues (Major and Crash)   Tony Mountifield
• [asterisk-dev] Re: Need to support about Asterisk Manager API   Tony Mountifield
• [asterisk-dev] Re: comments on this one ? (macros for parsing config file ?)   Tony Mountifield
• [asterisk-dev] Problem with svn.digium.com?   Tony Mountifield
• [asterisk-dev] Experiment: Dialplan size vs. Speed   Steve Murphy
• [asterisk-dev] bug 7506 -- This may be a life-changing experience to channel programmers! Please read!   Steve Murphy
• [asterisk-dev] bug_7506 changes for 1.4, trunk   Steve Murphy
• [asterisk-dev] CUT() default argument problem   Steve Murphy
• [asterisk-dev] Release schedule ?   Matthew Nicholson
• [asterisk-dev] Voicemail redisign   Matt O'Gorman
• [asterisk-dev] Voicemail redisign   Matt O'Gorman
• [asterisk-dev] Voicemail redisign   Matt O'Gorman
• [asterisk-dev] Voicemail redisign   Matt O'Gorman
• [asterisk-dev] Voicemail redisign   Matt O'Gorman
• [asterisk-dev] Voicemail redisign   Matt O'Gorman
• [asterisk-dev] Why -lnsl in IKSEMEL_LIB ?   Matt O'Gorman
• [asterisk-dev] Voicemail redisign   Carlton O'Riley
• [asterisk-dev] setting the q.931 "reason for redirecting"?   Kai Ober
• [asterisk-dev] setting the q.931 "reason for redirecting"?   Kai Ober
• [asterisk-dev] setting the q.931 "reason for redirecting"?   Kai Ober
• [asterisk-dev] setting the q.931 "reason for redirecting"?   Kai Ober
• [asterisk-dev] setting the q.931 "reason for redirecting"?   Kai Ober
• [asterisk-dev] setting the q.931 "reason for redirecting"?   Kai Ober
• [asterisk-dev] BUGid=7435 WAS: Re: Development :: Staying busy, please have patience   Kai Ober
• [asterisk-dev] blindtransfer and initiator hangup   Vitaly Oborsky
• [asterisk-dev] Support for TCP in asterisk   Sergey Okhapkin
• [asterisk-dev] Support for TCP in asterisk   Sergey Okhapkin
• [asterisk-dev] Support for TCP in asterisk   Sergey Okhapkin
• [asterisk-dev] <huge>aclocal.m4</huge> ???   Jeffrey C. Ollie
• [asterisk-dev] <huge>aclocal.m4</huge> ???   Jeffrey C. Ollie
• [asterisk-dev] SIP Multiple endpoints with same id   Jeffrey C. Ollie
• [asterisk-dev] SIP Multiple endpoints with same id   Jeffrey C. Ollie
• [asterisk-dev] SIP Multiple endpoints with same id   Jeffrey C. Ollie
• [asterisk-dev] Asterisk -> Mobile Phone quality issues   Stephane PAQUET
• [asterisk-dev] In call Codec parameterization in 1.6 ?   Tim Panton
• [asterisk-dev] IAX2 still broken   Tim Panton
• [asterisk-dev] In call Codec parameterization in 1.6 ?   Tim Panton
• [asterisk-dev] Pthread wrapper updates   Tim Panton
• [asterisk-dev] Code Support request for IAX provider sending registration refreshes of 0   Tim Panton
• [asterisk-dev] Pthread wrapper updates   Tim Panton
• [asterisk-dev] Adding AMR codec   Tim Panton
• [asterisk-dev] IAX2 very CPU hungry   Tim Panton
• [asterisk-dev] Call numbers in IAX2 media frames   Tim Panton
• [asterisk-dev] Call numbers in IAX2 media frames   Tim Panton
• [asterisk-dev] Queue callback   Tim Panton
• [asterisk-dev] Donwload Branch with Mac OSX   David Parcerisa
• [asterisk-dev] Re: [svn-commits] kpfleming: trunk r47257 - /trunk/   Jason Parker
• [asterisk-dev] Development :: Staying busy, please have patience   Jason Parker
• [asterisk-dev] CLI builtins commands ?   Jason Parker
• [asterisk-dev] app_realtime vs. func_realtime   Trevor Peirce
• [asterisk-dev] <huge>aclocal.m4</huge> ???   Oron Peled
• [asterisk-dev] Unable to make SIP calls   zen Perry
• [asterisk-dev] Ringback tones in asterisk?   zen Perry
• [asterisk-dev] Moh stops immediately   zen Perry
• [asterisk-dev] Push to Talk settings.   Jonson Player
• [asterisk-dev] ISDN gurus - getting isdn call progress ??   Daniel Pocock
• [asterisk-dev] Fwd: Asterisk and Max TNT PRI to SIP Authentication Issue, a little closer   JR Richardson
• [asterisk-dev] Re: Fwd: Asterisk and Max TNT PRI to SIP Authentication Issue, a little closer   JR Richardson
• [asterisk-dev] Re: Asterisk and Max TNT SIP Authentication Issue, WORKING   JR Richardson
• [asterisk-dev] RE: SIP Peer matching on Inbound calls   JR Richardson
• [asterisk-dev] Synchronized queue between threads   Tim Ringenbach
• [asterisk-dev] Re: why 'o' (preserve original callerid) is not default in app_dial.c ?   Luigi Rizzo
• [asterisk-dev] Re: why 'o' (preserve original callerid) is notdefault in app_dial.c ?   Luigi Rizzo
• [asterisk-dev] Re: why 'o' (preserve original callerid) is notdefault in app_dial.c ?   Luigi Rizzo
• [asterisk-dev] Re: [asterisk-commits] rizzo: trunk r47132 - in /trunk: include/asterisk/manager.h main/manager.c   Luigi Rizzo
• [asterisk-dev] Re: [asterisk-commits] oej: trunk r47194 - /trunk/channels/chan_sip.c   Luigi Rizzo
• [asterisk-dev] <huge>aclocal.m4</huge> ???   Luigi Rizzo
• [asterisk-dev] TLS/SSL futures   Luigi Rizzo
• [asterisk-dev] Why -lnsl in IKSEMEL_LIB ?   Luigi Rizzo
• [asterisk-dev] <huge>aclocal.m4</huge> ???   Luigi Rizzo
• [asterisk-dev] TLS/SSL futures   Luigi Rizzo
• [asterisk-dev] delayed responses from -dev list ?   Luigi Rizzo
• [asterisk-dev] Why -lnsl in IKSEMEL_LIB ?   Luigi Rizzo
• [asterisk-dev] Why -lnsl in IKSEMEL_LIB ?   Luigi Rizzo
• [asterisk-dev] <huge>aclocal.m4</huge> ???   Luigi Rizzo
• [asterisk-dev] Why -lnsl in IKSEMEL_LIB ?   Luigi Rizzo
• [asterisk-dev] SIP Multiple endpoints with same id   Luigi Rizzo
• [asterisk-dev] SIP Multiple endpoints with same id   Luigi Rizzo
• [asterisk-dev] SIP Multiple endpoints with same id   Luigi Rizzo
• [asterisk-dev] SIP Multiple endpoints with same id   Luigi Rizzo
• [asterisk-dev] SIP Multiple endpoints with same id   Luigi Rizzo
• [asterisk-dev] possible memory leak in chan_sip.c ?   Luigi Rizzo
• [asterisk-dev] possible memory leak in chan_sip.c ?   Luigi Rizzo
• [asterisk-dev] possible memory leak in chan_sip.c ?   Luigi Rizzo
• [asterisk-dev] possible memory leak in chan_sip.c ?   Luigi Rizzo
• [asterisk-dev] SIP Multiple endpoints with same id   Luigi Rizzo
• [asterisk-dev] possible memory leak in chan_sip.c ?   Luigi Rizzo
• [asterisk-dev] Release schedule ?   Luigi Rizzo
• [asterisk-dev] Release schedule ?   Luigi Rizzo
• [asterisk-dev] Release schedule ?   Luigi Rizzo
• [asterisk-dev] RE: SIP Peer matching on Inbound calls   Luigi Rizzo
• [asterisk-dev] chan_sip.c - struct sip_pvt is very large...   Luigi Rizzo
• [asterisk-dev] chan_sip.c - struct sip_pvt is very large...   Luigi Rizzo
• [asterisk-dev] RFC: internal CLI changes   Luigi Rizzo
• [asterisk-dev] Re: [asterisk-commits] oej: trunk r47611 - /trunk/include/asterisk/cli.h   Luigi Rizzo
• [asterisk-dev] RFC: internal CLI changes   Luigi Rizzo
• [asterisk-dev] comments on this one ? (macros for parsing config file ?)   Luigi Rizzo
• [asterisk-dev] Re: comments on this one ? (macros for parsing config file ?)   Luigi Rizzo
• [asterisk-dev] RFC: internal CLI changes   Luigi Rizzo
• [asterisk-dev] Re: [asterisk-commits] rizzo: trunk r47652 - in /trunk: include/asterisk/cli.h main/cli.c   Luigi Rizzo
• [asterisk-dev] CLI builtins commands ?   Luigi Rizzo
• [asterisk-dev] RFC: internal CLI changes   Luigi Rizzo
• [asterisk-dev] (mis)use of threadstorage.h ?   Luigi Rizzo
• [asterisk-dev] Re: [asterisk-commits] russell: trunk r48019 - in /trunk/include/asterisk: threadstorage.h utils.h   Luigi Rizzo
• [asterisk-dev] RFC - TLS certificate negotiation   Luigi Rizzo
• [asterisk-dev] better home for ast_dynamic_str and friends ?   Luigi Rizzo
• [asterisk-dev] Re: [svn-commits] rizzo: trunk r48084 - /trunk/main/utils.c   Luigi Rizzo
• [asterisk-dev] Re: [asterisk-commits] russell: trunk r48103 - /trunk/main/utils.c   Luigi Rizzo
• [asterisk-dev] Re: [asterisk-commits] oej: branch 1.4 r48166 - in /branches/1.4: ./ channels/chan_sip.c   Luigi Rizzo
• [asterisk-dev] Re: How to busy out PRI channels?   Adam Robins
• [asterisk-dev] Looking to hire Experienced Asterisk and PHP Developer   A. Rose
• [asterisk-dev] IAX2 very CPU hungry   Mark Rounds
• [asterisk-dev] Release schedule ?   Matthew Rubenstein
• [asterisk-dev] Replacement RTPd?   Matthew Rubenstein
• [asterisk-dev] timelimit for calls bridged by queue app.   Rajkumar S
• [asterisk-dev] Synchronized queue between threads   Corrado Santoro
• [asterisk-dev] Synchronized queue between threads   Corrado Santoro
• [asterisk-dev] translate.c:88 powerof: Powerof 0: No power??   Hans Petter Selasky
• [asterisk-dev] translate.c:88 powerof: Powerof 0: No power??   Hans Petter Selasky
• [asterisk-dev] SIP ghost calls   Hans Petter Selasky
• [asterisk-dev] SIP ghost calls   Hans Petter Selasky
• [asterisk-dev] SIP ghost calls   Hans Petter Selasky
• [asterisk-dev] Help with 240 samples on frames read from chan_iax   Moises Silva
• [asterisk-dev] Re: Help with 240 samples on frames read from chan_iax   Moises Silva
• [asterisk-dev] Re: Help with 240 samples on frames read from chan_iax   Moises Silva
• [asterisk-dev] Re: Help with 240 samples on frames read from chan_iax   Moises Silva
• [asterisk-dev] Re: Help with 240 samples on frames readfromchan_iax   Moises Silva
• [asterisk-dev] Re: Help with 240 samples on frames readfromchan_iax   Moises Silva
• [asterisk-dev] Need help in how to get notified of events from client   Moises Silva
• [asterisk-dev] OriginateSuccess and OriginateError incomplete (in 1.4 beta3)   Moises Silva
• [asterisk-dev] Re: Help with 240 samples on frames readfromchan_iax   Denis Smirnov
• [asterisk-dev] Bug marshals - won't you take a look at 8188   Jared Smith
• [asterisk-dev] App_SMS: UDH (EMS) bug and fix   Jared Smith
• [asterisk-dev] Need help in how to originate call from c to asterisk ie controlling asterisk cli commands through c.   Jared Smith
• [asterisk-dev] channel_walk_locked possible bug / possible fix?   Jared Smith
• [asterisk-dev] IAX2 very CPU hungry   Jared Smith
• [asterisk-dev] OriginateSuccess and OriginateError incomplete (in 1.4 beta3)   Jared Smith
• [asterisk-dev] IAX2 very CPU hungry   Derek Smithies
• [asterisk-dev] IAX2 very CPU hungry   Derek Smithies
• [asterisk-dev] extend IAX2 IE proposal   Derek Smithies
• [asterisk-dev] Codec translation??   Mahmoud Sohrabi
• [asterisk-dev] ast_channel_make_compatible   Mahmoud Sohrabi
• [asterisk-dev] app_realtime vs. func_realtime   Mark Spencer
• [asterisk-dev] Voicemail argument list limit   Donald Stahl
• [asterisk-dev] Voicemail argument list limit   Donald Stahl
• [asterisk-dev] Application for streaming a live audio   Steven
• [asterisk-dev] chan_h323 compile problem in SVN HEAD   Di-Shi Sun
• [asterisk-dev] minor chan_h323 compile problems in SVN HEAD   Di-Shi Sun
• [asterisk-dev] oh323 faststart with cisco gateways question   Di-Shi Sun
• Fw: [asterisk-dev] oh323 faststart with cisco gateways   Di-Shi Sun
• [asterisk-dev] extend IAX2 IE proposal   Di-Shi Sun
• [asterisk-dev] Zaptel 1.2.11 released   Asterisk Development Team
• [asterisk-dev] Looking to hire Experienced Asterisk and PHP Developer   Amit at Telepele
• [asterisk-dev] CQDRs - Call Quality records (was: Bug marshals - won't you take a look...)   John Todd
• [asterisk-dev] chan_sip - performance, and OPTIONS poking for 1000s of peers   John Todd
• [asterisk-dev] Re: How to busy out PRI channels?   Steve Totaro
• [asterisk-dev] Re: How to busy out PRI channels?   Steve Totaro
• [asterisk-dev] Application for streaming a live audio   Steve Totaro
• [asterisk-dev] Application for streaming a live audio   Steve Totaro
• [asterisk-dev] Video in chan_iax is broken...   Tristan
• [asterisk-dev] Code Support request for IAX provider sending registration refreshes of 0   James Trix
• [asterisk-dev] Code Support request for IAX provider sending registration refreshes of 0   James Trix
• [asterisk-dev] Code Support request for IAX provider sending registration refreshes of 0   James Trix
• [asterisk-dev] Code Support request for IAX provider sending registration refreshes of 0   James Trix
• [asterisk-dev] Code Support request for IAX provider sending registration refreshes of 0   James Trix
• [asterisk-dev] Code Support request for IAX provider sending registration refreshes of 0   James Trix
• [asterisk-dev] Code Support request for IAX provider sending registration refreshes of 0   James Trix
• [asterisk-dev] need help with development   Wojciech Tryc
• [asterisk-dev] RFC: internal CLI changes   Dylan VanHerpen
• [asterisk-dev] Queues and Flash/SendDTMF in hybrid PBX   Vieri
• [asterisk-dev] Queues and Flash/SendDTMF in hybrid PBX   Vieri
• [asterisk-dev] Need your help to use TDM(Hardware) in Asterisk   Viswanathan
• [asterisk-dev] SIP Multiple endpoints with same id   Alexei Volkov
• [asterisk-dev] endianness in chan_phone   Yoann WALTHER
• [asterisk-dev] app_dictate   Michael Welter
• [asterisk-dev] dynamic queue members get stuck with status 'on hold'   BJ Weschke
• [asterisk-dev] Bug marshals - won't you take a look at 8188   BJ Weschke
• [asterisk-dev] Release schedule ?   Howard White
• [asterisk-dev] Re: How to busy out PRI channels?   Eric "ManxPower" Wieling
• [asterisk-dev] Re: why 'o' (preserve original callerid) is notdefault in app_dial.c ?   Eric "ManxPower" Wieling
• [asterisk-dev] Clearing pick-up groups on Zap/ channels   Eric "ManxPower" Wieling
• [asterisk-dev] SIP Multiple endpoints with same id   Eric "ManxPower" Wieling
• [asterisk-dev] SIP Multiple endpoints with same id   Eric "ManxPower" Wieling
• [asterisk-dev] SIP Multiple endpoints with same id   Eric "ManxPower" Wieling
• [asterisk-dev] SIP Multiple endpoints with same id   Eric "ManxPower" Wieling
• [asterisk-dev] SIP Multiple endpoints with same id   Eric "ManxPower" Wieling
• [asterisk-dev] Re: Asterisk and Max TNT SIP Authentication Issue, WORKING   Eric "ManxPower" Wieling
• [asterisk-dev] SUGGESTION: Attended and blind transfer in the same digit   Eric "ManxPower" Wieling
• [asterisk-dev] Re: chan_sip - performance, and OPTIONS poking for 1000s of peers   Eric "ManxPower" Wieling
• [asterisk-dev] Recording synchronization fails   Guillermo Winkler
• [asterisk-dev] asterisk disappeared! any thoughts?   Ramu Yadav
• [asterisk-dev] TD - stops receiving characters   Warrick Zedi
• [asterisk-dev] auto hangup problem   Ma Zhiyong
• [asterisk-dev] Re: auto hangup problem   Ma Zhiyong
• [asterisk-dev] SIP ghost calls   Zoa
• [asterisk-dev] app_queue: Out of order call delivery when agents are member of multiple queues   Ardjan Zwartjes
• [asterisk-dev] Fwd: Asterisk and Max TNT PRI to SIP Authentication Issue, a little closer   alex at pilosoft.com
• [asterisk-dev] Application for streaming a live audio   anuroop.r at aricent.com
• [asterisk-dev] Receiving calls on Yahoo & MSN Messengers?   asterisk at teladdict.com
• [asterisk-dev] ast_channel_masquerade fails   dmb
• [asterisk-dev] OriginateSuccess and OriginateError incomplete (in 1.4 beta3)   dmb
• [asterisk-dev] Writing Module for Asterisk   Elkeir ismail
• [asterisk-dev] Howto extend the interface of Asterisk   levin lee
• [asterisk-dev] [asterisk-users] Queues: roundrobin w/ reset ("circular call distribution")   lists.digium.com at tgice.com
• [asterisk-dev] Users.conf   mrdlnf
• [asterisk-dev] 1.4 svn configure looks to Qt in wrong Fedora places   sean
• [asterisk-dev] Need help in how to originate call from c to asterisk ie controlling asterisk cli commands through c.   rajesh singh

Last message date: Thu Nov 30 22:12:45 MST 2006
Archived on: Thu Nov 30 22:12:33 MST 2006

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