485 messages

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

2009 September [All Months]

Page 1 (Messages 1 to 100): 1 2 3 4 5

Re: [asterisk-dev] [Code Review] SIP: peer matchingbycallbackextension - Nick Lewis
Re: [asterisk-dev] - [Iax2 Implementation] Reagrding Attendted Transfer - Kevin P. Fleming
Re: [asterisk-dev] Segmentation Fault on 1.4.24.1 - Leif Madsen
[asterisk-dev] [Code Review] PineTree:: Matching SIP peers beyond SIP proxy - Olle E Johansson
Re: [asterisk-dev] [Code Review] Add mutestream manager action and MUTESTREAM() dialplan function - Russell Bryant
Re: [asterisk-dev] Pinetree :: For Asterisk SIP trunks behind a SIP proxy - Olle E. Johansson
[asterisk-dev] - [Iax2 Implementation] Reagrding Attendted Transfer - Kumar Subramanian
Re: [asterisk-dev] [Code Review] Add mutestream manager action and MUTESTREAM() dialplan function - Olle E Johansson
Re: [asterisk-dev] [Code Review] Add mutestream manager action and MUTESTREAM() dialplan function - Russell Bryant
Re: [asterisk-dev] [Code Review] Add mutestream manager action and MUTESTREAM() dialplan function - Olle E Johansson
Re: [asterisk-dev] [Code Review] PineTree:: Matching SIP peers beyond SIP proxy - Olle E. Johansson
Re: [asterisk-dev] [Code Review] SIP uri parsing cleanup - David Vossel
Re: [asterisk-dev] [Code Review] SIP: Re-send non-100 provisional responses every 60 seconds until a final response is sent - Olle E Johansson
Re: [asterisk-dev] Pinetree :: For Asterisk SIP trunks behind aSIPproxy - Nick Lewis
Re: [asterisk-dev] [Code Review] SIP uri parsing cleanup - Nick Lewis
Re: [asterisk-dev] Peer matching in trunk - matching on contact? - Klaus Darilion
Re: [asterisk-dev] [Code Review] Dynamic parking lots - Michiel van Baak
Re: [asterisk-dev] definition of RTP jitter - potential bug in Asterisk - Leif Madsen
Re: [asterisk-dev] definition of RTP jitter - potential bug in Asterisk - Klaus Darilion
Re: [asterisk-dev] [Code Review] Dynamic parking lots - Sean Bright
Re: [asterisk-dev] please please please open ViewVC for svn.digium.com again - Kevin P. Fleming
Re: [asterisk-dev] please please please open ViewVC for svn.digium.com again - Jeffrey Ollie
Re: [asterisk-dev] [svn-commits] russell: branch 1.2 r216262 - /branches/1.2/doc/IAX2-security.txt - Olle E. Johansson
[asterisk-dev] [Code Review] Optionally build apps in utils/ directory - Sean Bright
Re: [asterisk-dev] please please please open ViewVC for svn.digium.com again - Saúl Ibarra
Re: [asterisk-dev] [Code Review] SIP: peer matching by address with TCP/TLS - Russell Bryant
Re: [asterisk-dev] [Code Review] SIP: peer matching by address with TCP/TLS - David Vossel
[asterisk-dev] [Code Review] SIP: peer matching TCP/TLS 1.6.0 - David Vossel
Re: [asterisk-dev] please please please open ViewVC for svn.digium.com again - Jeffrey Ollie
Re: [asterisk-dev] please please please open ViewVC for svn.digium.com again - Tzafrir Cohen
Re: [asterisk-dev] please please please open ViewVC for svn.digium.com again - Tzafrir Cohen
Re: [asterisk-dev] [Code Review] Optionally build apps in utils/ directory - Tzafrir Cohen
Re: [asterisk-dev] Linksys SPA962 losing registration - Jeff LaCoursiere
Re: [asterisk-dev] MeetMe and kernel module dependency - Jeff Gehlbach
[asterisk-dev] Patch: Manager api, Posibility to send two channels in different direcitons - Håkon Nessjøen
Re: [asterisk-dev] [svn-commits] dvossel: trunk r216594 -/trunk/channels/chan_sip.c - Nick Lewis
Re: [asterisk-dev] [svn-commits] dvossel: trunk r216594 -/trunk/channels/chan_sip.c - Olle E. Johansson
Re: [asterisk-dev] Linksys SPA962 losing registration - Stefan Schmidt
[asterisk-dev] SIP: handling multiple m=video or m= audio lines - David Vossel
Re: [asterisk-dev] SIP: handling multiple m=video or m= audio lines - Kevin P. Fleming
[asterisk-dev] Dahdi compilation error on ARM - Rafael Seste
Re: [asterisk-dev] SIP: handling multiple m=video or m= audio lines - Mark Michelson
Re: [asterisk-dev] SIP: handling multiple m=video or m= audio lines - Kevin P. Fleming
Re: [asterisk-dev] oej: trunk r216805 - /trunk/channels/chan_sip.c - Russell Bryant
Re: [asterisk-dev] Strange behaviour of Incomplete() application - Tilghman Lesher
Re: [asterisk-dev] channel driver question - Martin
Re: [asterisk-dev] oej: trunk r216805 - /trunk/channels/chan_sip.c - Olle E. Johansson
Re: [asterisk-dev] [Asterisk 0014810]: [patch] channel-specific hangupcauses - Olle E. Johansson
Re: [asterisk-dev] SIP: handling multiple m=video or m= audio lines - Olle E. Johansson
[asterisk-dev] Problem with chan->_bridge Pointer in answered Macro - Stefan Schmidt
Re: [asterisk-dev] [Asterisk 0014810]: [patch] channel-specific hangupcauses - Tim Ringenbach
[asterisk-dev] Opening temp file in action_command - Milad Rastian
[asterisk-dev] Bug in SendFax/ReceiveFax ? - Håkon Nessjøen
Re: [asterisk-dev] DAHDI_CHECK_HOOKSTATE and setting rxisoffhook - Tzafrir Cohen
[asterisk-dev] A Curious Question Cisco IAD - Michial Thompson
Re: [asterisk-dev] [dahdi-commits] tzafrir: tools/trunk r7134 - /tools/trunk/ - Kevin P. Fleming
Re: [asterisk-dev] [Code Review] Fix race condition that may lead to a crash in ast_hangup() procedure. - Matthew Nicholson
[asterisk-dev] [Code Review] Format change when queue still has frames of the old format - Tilghman Lesher
Re: [asterisk-dev] [Code Review] Fix race condition that may lead to a crash in ast_hangup() procedure. - Russell Bryant
Re: [asterisk-dev] [Code Review] Fix race condition that may lead to a crash in ast_hangup() procedure. - Matthew Nicholson
Re: [asterisk-dev] [Code Review] Fix race condition that may lead to a crash in ast_hangup() procedure. - Russell Bryant
Re: [asterisk-dev] [Code Review] Fix race condition that may lead to a crash in ast_hangup() procedure. - Russell Bryant
Re: [asterisk-dev] MeetMe in Macro - Miguel Molina
Re: [asterisk-dev] [Code Review] Fix race condition that may lead to a crash in ast_hangup() procedure. - Russell Bryant
Re: [asterisk-dev] [Code Review] Fix race condition that may lead to a crash in ast_hangup() procedure. - Matthew Nicholson
Re: [asterisk-dev] [Code Review] Document Asterisk open source issue tracker workflow - Russell Bryant
Re: [asterisk-dev] [Code Review] SIP: INVITE w/Replaces deadlock fix - Russell Bryant
Re: [asterisk-dev] [Code Review] AST-33: Create a list of channel variables to be posted within AMI call events - Tilghman Lesher
Re: [asterisk-dev] [Code Review] AST-33: Create a list of channel variables to be posted within AMI call events - Tilghman Lesher
Re: [asterisk-dev] [Code Review] SIP: INVITE w/Replaces deadlock fix - Olle E Johansson
Re: [asterisk-dev] [Code Review] Format change when queue still has frames of the old format - Tilghman Lesher
[asterisk-dev] Help me test a new feature for AMI Redirect() command - Håkon Nessjøen
Re: [asterisk-dev] [Code Review] New application JabberReceive, implement SendText in chan_gtalk and chan_jingle - Jeff Peeler
Re: [asterisk-dev] Looking for a DID provider in Manila, Philippines - Matt D
Re: [asterisk-dev] Looking for a DID provider in Manila, Philippines - VoIP er
Re: [asterisk-dev] Dahdi-linux embedded - Tzafrir Cohen
Re: [asterisk-dev] [Code Review] SIP: port configuration - David Vossel
Re: [asterisk-dev] make -j - Sean Bright
Re: [asterisk-dev] [Code Review] New application JabberReceive - Olle E. Johansson
Re: [asterisk-dev] [Code Review] tcptls_session memory leak - Russell Bryant
Re: [asterisk-dev] make -j - Sean Bright
Re: [asterisk-dev] [svn-commits] mvanbaak: branch 1.4 r220027 - /branches/1.4/build_tools/mkpkgconfig - Sean Bright
Re: [asterisk-dev] ISDN Calling/Caller/Redirecting Subaddress support work. - Alec Davis
Re: [asterisk-dev] ISDN Calling/Caller/Redirecting Subaddress support work. - Richard Mudgett
Re: [asterisk-dev] make -j - Sean Bright
Re: [asterisk-dev] [Code Review] AST-33: Create a list of channel variables to be posted within AMI call events - Olle E. Johansson
Re: [asterisk-dev] make -j - Klaus Darilion
[asterisk-dev] [Code Review] Finish implementaton of astobj2 OBJ_MULTIPLE, and convert ast_channel_iterator to use it. - Kevin Fleming
[asterisk-dev] Inquiry:How to get remote access via Asterisk? - hadi motamedi
Re: [asterisk-dev] [Code Review] AST-33: Create a list of channel variables to be posted within AMI call events - Stefan Reuter
Re: [asterisk-dev] [Code Review] AST-33: Create a list of channel variables to be posted within AMI call events - Olle E. Johansson
Re: [asterisk-dev] AST-XXX issues ? - Tilghman Lesher
Re: [asterisk-dev] Inquiry:How to get remote access via Asterisk? - Alec Davis
[asterisk-dev] Review request(s) - Brent Thomson
Re: [asterisk-dev] Review request(s) - Russell Bryant
[asterisk-dev] Destinatio operator recognition by LNP Distinctive tone during "DIAL" - Isamar Maia
Re: [asterisk-dev] [Code Review] Added ability to perform SRV lookups for AGI URIs - Jared Smith
Re: [asterisk-dev] [Code Review] Added ability to perform SRV lookups for AGI URIs - Olle E Johansson
Re: [asterisk-dev] SIP TLS handshake needs a timeout - Olle E. Johansson
Re: [asterisk-dev] reasoning behind char[0] - Kevin P. Fleming

Page 1 (Messages 1 to 100): 1 2 3 4 5