Archived Mails Search Asterisk Developers Goto page 1, 2, 3, 4, 5, 6  Next Archived Mails Forum Index » Asterisk Developers  Topics   Replies   Author   Views   Last Post  realtime engine 0 Bhrugu Mehta 133 Mon Jan 18, 2010 1:53 pm Bhrugu Mehta fallback to audio faxing when T.38 INVITE fail with 488/606 Not acceptable 3 Kristijan Vrban 133 Mon Jan 18, 2010 12:48 pm Klaus Darilion Manager "sipchannel" command 0 Olle E. Johansson 56 Mon Jan 18, 2010 12:15 pm Olle E. Johansson [Code Review] CDR: Add Calling and Called Subaddress fields to CDR record 3 Alec Davis 123 Mon Jan 18, 2010 12:12 pm Alec Davis Regarding asterisk source code contribution 1 Sherin Baby Thomas 59 Mon Jan 18, 2010 12:04 pm David Ruggles Multiple authorization header for a SIP message. More info to the bug report. 1 Eduardo Ferro 88 Mon Jan 18, 2010 7:27 am Olle E. Johansson signal 0 Bhrugu Mehta 53 Mon Jan 18, 2010 6:30 am Bhrugu Mehta thread debug 4 Bhrugu Mehta 159 Mon Jan 18, 2010 4:56 am Bhrugu Mehta SIP invite to localhost/loopback 8 Geoffrey Mina 272 Mon Jan 18, 2010 12:45 am Matt Riddell chan_datacard and RTCP 5 Artem Makhutov 168 Sun Jan 17, 2010 8:26 pm Artem Makhutov lmadsen: trunk r239834 - /trunk/configs/extensions.conf.sample 0 Russell Bryant 92 Fri Jan 15, 2010 11:25 pm Russell Bryant dvossel: trunk r239712 - in /trunk: ./ apps/ main/ 0 Russell Bryant 75 Fri Jan 15, 2010 11:23 pm Russell Bryant tilghman: trunk r239000 - in /trunk: Makefile main/asterisk.c 0 Russell Bryant 81 Fri Jan 15, 2010 10:57 pm Russell Bryant Requiring session timers and call drop 0 ê¹€ì�˜ê²½ 82 Fri Jan 15, 2010 9:10 pm ê¹€ì�˜ê²½ [Code Review] Added ability to perform SRV lookups for AGI URIs 0 Tilghman Lesher 75 Fri Jan 15, 2010 7:26 pm Tilghman Lesher [Code Review] High Resolution Call Time for Billsec and Duration 2 Tilghman Lesher 110 Fri Jan 15, 2010 7:24 pm Tilghman Lesher SIP NOTIFY with 183 Ringing Body stays after call is tranfered 0 Karsten Asche 79 Fri Jan 15, 2010 5:35 pm Karsten Asche [Code Review] Add notification of interrupted file and fix bug 2 thedavidfactor 94 Fri Jan 15, 2010 5:15 pm thedavidfactor Asterisk 1.6.2.1 Now Available 0 Asterisk Development Team 78 Fri Jan 15, 2010 3:30 pm Asterisk Development Team Asterisk 1.6.1.13 Now Available 0 Asterisk Development Team 79 Fri Jan 15, 2010 3:29 pm Asterisk Development Team Asterisk 1.6.0.21 Now Available 0 Asterisk Development Team 73 Fri Jan 15, 2010 3:28 pm Asterisk Development Team Asterisk 1.4.29 Now Available 0 Asterisk Development Team 83 Fri Jan 15, 2010 3:28 pm Asterisk Development Team Ast_channel question 2 Salvatore Frandina 93 Fri Jan 15, 2010 8:26 am Salvatore Frandina SIP CHANNEL(rtpqos,audio,...) values 0 Kirill 'Big K' Katsnelson 70 Fri Jan 15, 2010 6:26 am Kirill 'Big K' Katsnelson [review-request] T.30 <-> T.38 Gateway application 0 Daniel Ferenci 73 Thu Jan 14, 2010 3:16 pm Daniel Ferenci Adding SendDTMF support to ExternalIVR 0 David Ruggles 75 Wed Jan 13, 2010 8:04 pm David Ruggles ast_rtp->altthem 2 Olle E. Johansson 103 Wed Jan 13, 2010 7:01 pm Mark Michelson [Code Review] Add notificiation of intrupted file and fix bug 1 thedavidfactor 79 Wed Jan 13, 2010 5:55 pm thedavidfactor [Code Review] Provide method for testing GotoIfTime without changing localtime 6 Tilghman Lesher 174 Wed Jan 13, 2010 3:46 pm David Vossel ISDN Cause codes for unanswered calls 1 Steve Moran 89 Wed Jan 13, 2010 2:55 pm Håkon Nessjøen Problem FOP 1 Naudit007 86 Wed Jan 13, 2010 2:53 pm Nick Lewis Moving call from channel x to channel x 5 Håkon Nessjøen 329 Wed Jan 13, 2010 2:49 pm Håkon Nessjøen Pbx Developer 1 Gary Cardozo 74 Wed Jan 13, 2010 12:38 pm Bhrugu Mehta fallback to audio faxing when T.38 INVITE fail with 488/606 Not acceptable 1 Kevin P. Fleming 69 Wed Jan 13, 2010 12:32 pm Olle E. Johansson IMAP Invalid system flag UNSEEN 1 Hoggins! 73 Wed Jan 13, 2010 10:23 am Nick Lewis Muffins for new peer types 6 Nick Lewis 190 Wed Jan 13, 2010 10:16 am Nick Lewis Unavailable peer not show in BLF 0 Fabio Terrile 65 Wed Jan 13, 2010 9:45 am Fabio Terrile [Code Review] Missing new-message notification for urgent messages 0 Tilghman Lesher 91 Tue Jan 12, 2010 11:57 pm Tilghman Lesher [Code Review] add silence gen to wait apps 3 David Vossel 123 Tue Jan 12, 2010 11:53 pm Tilghman Lesher Out-of-order events from AMI interface - to be expected, or a bug? 1 Alex Villací­s Lasso 92 Tue Jan 12, 2010 11:44 pm Tilghman Lesher Asterisk Mod Request 1 Glenn Elliott 75 Tue Jan 12, 2010 3:47 pm David Ruggles [Code Review] CDR: add Dialed Number Identifier field (DNID) field in record 6 Alec Davis 176 Tue Jan 12, 2010 3:41 pm Tilghman Lesher Application that make sip call 1 Salvatore Frandina 77 Tue Jan 12, 2010 12:51 pm Stephen Coles Asterisk application development 0 Salvatore Frandina 75 Tue Jan 12, 2010 8:52 am Salvatore Frandina StopMonitor() reverts itself after SIP transfer: bug or feature? 0 Kirill 'Big K' Katsnelson 56 Tue Jan 12, 2010 8:19 am Kirill 'Big K' Katsnelson astmm 2 Bhrugu Mehta 74 Tue Jan 12, 2010 6:59 am Bhrugu Mehta Asterisk dialplan application 3 Salvatore Frandina 103 Tue Jan 12, 2010 6:22 am Bhrugu Mehta Asterisk 1.4.29-rc1, 1.6.0.21-rc1, 1.6.1.13-rc1, and 1.6.2.1-rc1 Now Released 0 Asterisk Development Team 78 Mon Jan 11, 2010 7:29 pm Asterisk Development Team [Code Review] Fix ao2_callback when both OBJ_MULTIPLE and OBJ_NODATA are passed 2 Sean Bright 84 Mon Jan 11, 2010 4:19 pm Kevin Fleming CSTA support for asterisk... or begining this module/project 5 Guilherme Rezende 196 Mon Jan 11, 2010 3:35 pm Chris Mylonas Display topics from previous: All Topics1 Day7 Days2 Weeks1 Month3 Months6 Months1 Year  Archived Mails Forum Index » Asterisk Developers Page 1 of 6 Goto page 1, 2, 3, 4, 5, 6  Next Jump to:  Select a forum VOIP----------------Asterisk UsersAsterisk BusinessAsterisk DevelopersCISCO VOIPOpenSIPs UserspjsipsipXecs Users Server virtualization----------------XEN Radius----------------FreeRadius VPN----------------OpenVPN Users OS----------------CentOS Tips----------------Tips | Tools | Articles