Summary: | ASTERISK-10697: Allow direct RTP in "Dial" with "tT" options if phones are "canreinvite=yes" and "dtmfmode=info" | ||
Reporter: | Iñaki Baz Castillo (ibc) | Labels: | |
Date Opened: | 2007-11-06 13:04:15.000-0600 | Date Closed: | 2007-11-20 08:18:05.000-0600 |
Priority: | Minor | Regression? | No |
Status: | Closed/Complete | Components: | Applications/app_dial |
Versions: | Frequency of Occurrence | ||
Related Issues: | |||
Environment: | Attachments: | ||
Description: | In case two phones have "canreinvite=yes" and "dtmfmode=info" there is no reason Asterisk to be in the media path when "Dial" with "t" or "T" option. "dtmfmode=info" means DTMF in SIP INFO messages, so Asterisk doesn't need to be in the media path to get phones DTMF for native transfer. But unfortunatelly Asterisk remains in the media path in the case above. Could it be possible to consider this in "channels.c"? | ||
Comments: | By: Digium Subversion (svnbot) 2007-11-06 14:54:04.000-0600 Repository: asterisk Revision: 89057 U trunk/main/channel.c ------------------------------------------------------------------------ r89057 | file | 2007-11-06 14:54:04 -0600 (Tue, 06 Nov 2007) | 4 lines Remove native bridging check for DTMF based transfers. Thanks to the last batch of RTP changes it is no longer required for the media stream to go through Asterisk if DTMF is going over signalling. It will simply reinvite back as needed. (closes issue ASTERISK-10697) Reported by: ibc ------------------------------------------------------------------------ By: Digium Subversion (svnbot) 2007-11-06 15:31:34.000-0600 Repository: asterisk Revision: 89066 _U team/murf/fast-ast2/ U team/murf/fast-ast2/apps/app_amd.c U team/murf/fast-ast2/apps/app_chanisavail.c U team/murf/fast-ast2/apps/app_chanspy.c U team/murf/fast-ast2/apps/app_directed_pickup.c U team/murf/fast-ast2/apps/app_exec.c U team/murf/fast-ast2/apps/app_festival.c U team/murf/fast-ast2/apps/app_followme.c U team/murf/fast-ast2/apps/app_forkcdr.c U team/murf/fast-ast2/apps/app_getcpeid.c U team/murf/fast-ast2/apps/app_macro.c U team/murf/fast-ast2/apps/app_minivm.c U team/murf/fast-ast2/apps/app_mixmonitor.c U team/murf/fast-ast2/apps/app_morsecode.c U team/murf/fast-ast2/apps/app_mp3.c U team/murf/fast-ast2/apps/app_nbscat.c U team/murf/fast-ast2/apps/app_playback.c U team/murf/fast-ast2/apps/app_readfile.c U team/murf/fast-ast2/apps/app_sayunixtime.c U team/murf/fast-ast2/apps/app_sms.c U team/murf/fast-ast2/apps/app_softhangup.c U team/murf/fast-ast2/apps/app_speech_utils.c U team/murf/fast-ast2/apps/app_stack.c U team/murf/fast-ast2/apps/app_test.c U team/murf/fast-ast2/apps/app_waitforring.c U team/murf/fast-ast2/apps/app_waitforsilence.c U team/murf/fast-ast2/apps/app_while.c U team/murf/fast-ast2/channels/chan_agent.c U team/murf/fast-ast2/channels/chan_gtalk.c U team/murf/fast-ast2/channels/chan_jingle.c U team/murf/fast-ast2/channels/chan_sip.c U team/murf/fast-ast2/codecs/codec_zap.c U team/murf/fast-ast2/include/asterisk/jabber.h U team/murf/fast-ast2/include/asterisk/lock.h U team/murf/fast-ast2/include/asterisk/tdd.h U team/murf/fast-ast2/main/ast_expr2.fl U team/murf/fast-ast2/main/ast_expr2f.c U team/murf/fast-ast2/main/astmm.c U team/murf/fast-ast2/main/channel.c U team/murf/fast-ast2/main/config.c U team/murf/fast-ast2/main/fskmodem.c U team/murf/fast-ast2/main/loader.c U team/murf/fast-ast2/main/pbx.c U team/murf/fast-ast2/main/tdd.c U team/murf/fast-ast2/res/res_features.c U team/murf/fast-ast2/res/res_indications.c U team/murf/fast-ast2/res/res_jabber.c U team/murf/fast-ast2/res/res_monitor.c U team/murf/fast-ast2/res/res_musiconhold.c ------------------------------------------------------------------------ r89066 | murf | 2007-11-06 15:31:33 -0600 (Tue, 06 Nov 2007) | 188 lines Merged revisions 89031,89034,89038,89041,89043-89044,89047-89052,89054-89055,89057,89062 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r89031 | rizzo | 2007-11-06 10:05:13 -0700 (Tue, 06 Nov 2007) | 17 lines Fix embedding of modules on FreeBSD: the constructor for the list of modules was run after the constructors for the embedded modules (which appended entries to the list). As a result, the list appeared empty when it was time to use it. On linux the order of execution of constructor was evidently different (it may depend on the ordering of modules in the ELF file). This is only a workaround - there may be other situations where the execution of constructors causes problems, so if we manage to find a more general solution this workaround can go away. ................ r89034 | file | 2007-11-06 10:10:03 -0700 (Tue, 06 Nov 2007) | 12 lines Merged revisions 89032 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89032 | file | 2007-11-06 13:08:05 -0400 (Tue, 06 Nov 2007) | 4 lines Make it so that if a peer is determined to be unreachable using qualify their devicestate will report back unavailable. (closes issue ASTERISK-10551) Reported by: pj ........ ................ r89038 | russell | 2007-11-06 11:23:36 -0700 (Tue, 06 Nov 2007) | 19 lines Merged revisions 89037 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89037 | russell | 2007-11-06 12:20:07 -0600 (Tue, 06 Nov 2007) | 11 lines If someone were to delete the files used by an existing MOH class, and then issue a reload, further use of that class could result in a crash due to dividing by zero. This set of changes fixes up some places to prevent this from happening. (closes issue ASTERISK-10500) Reported by: jcomellas Patches: res_musiconhold_division_by_zero.patch uploaded by jcomellas (license 282) Additional changes added by me. ........ ................ r89041 | qwell | 2007-11-06 11:44:19 -0700 (Tue, 06 Nov 2007) | 4 lines Allow gtalk and jingle to use TLS connections again. Closes issue ASTERISK-9675 ................ r89043 | oej | 2007-11-06 12:04:29 -0700 (Tue, 06 Nov 2007) | 12 lines Merged revisions 89042 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89042 | oej | 2007-11-06 19:53:37 +0100 (Tis, 06 Nov 2007) | 2 lines Bug fixes to tdd support in zaptel. ........ (Small changes for trunk) ................ r89044 | mmichelson | 2007-11-06 12:04:45 -0700 (Tue, 06 Nov 2007) | 7 lines "show application <foo>" changes for clarity. (closes issue ASTERISK-10696, reported and patched by blitzrage) Many thanks! ................ r89047 | qwell | 2007-11-06 12:10:18 -0700 (Tue, 06 Nov 2007) | 12 lines Merged revisions 89046 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89046 | qwell | 2007-11-06 13:09:30 -0600 (Tue, 06 Nov 2007) | 4 lines Correctly set the total number of channels from a zaptel transcoder board. SPD-49, patch by Matthew Nicholson. ........ ................ r89048 | oej | 2007-11-06 12:10:26 -0700 (Tue, 06 Nov 2007) | 2 lines Additional TDD changes (preparing for SIP changes - adding TDD support to SIP) ................ r89049 | tilghman | 2007-11-06 12:16:02 -0700 (Tue, 06 Nov 2007) | 10 lines Merged revisions 89045 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89045 | tilghman | 2007-11-06 13:09:06 -0600 (Tue, 06 Nov 2007) | 2 lines We went to the trouble of creating a method of tracking failed trylocks, then never turned it on (oops). ........ ................ r89050 | oej | 2007-11-06 12:23:10 -0700 (Tue, 06 Nov 2007) | 2 lines Formatting. Illegaly using some spare spaces from Russell's space-bucket. ................ r89051 | murf | 2007-11-06 12:40:33 -0700 (Tue, 06 Nov 2007) | 1 line Hoping to avoid a crash in OSX for a problem blitzrage found ................ r89052 | russell | 2007-11-06 12:51:37 -0700 (Tue, 06 Nov 2007) | 4 lines Fix the memory show allocations CLI command so that it doesn't spew out all of the current memory allocations when you start Asterisk, when the command's handler gets called for initialization. ................ r89054 | russell | 2007-11-06 13:22:50 -0700 (Tue, 06 Nov 2007) | 11 lines Merged revisions 89053 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89053 | russell | 2007-11-06 14:18:49 -0600 (Tue, 06 Nov 2007) | 3 lines Fix init_classes() so that classes that actually do have files loaded aren't treated as empty, and immediately destroyed ... ........ ................ r89055 | mmichelson | 2007-11-06 13:32:49 -0700 (Tue, 06 Nov 2007) | 9 lines Instead of trying to callback a local channel on a failed attended transfer, call the device that made the transfer instead. This makes for much smoother calling back when queues are involved. (closes issue ASTERISK-10680, reported by IPetrov) Tremendous thanks to Russell for pulling me out of my block I was having on this one ................ r89057 | file | 2007-11-06 13:55:58 -0700 (Tue, 06 Nov 2007) | 4 lines Remove native bridging check for DTMF based transfers. Thanks to the last batch of RTP changes it is no longer required for the media stream to go through Asterisk if DTMF is going over signalling. It will simply reinvite back as needed. (closes issue ASTERISK-10697) Reported by: ibc ................ r89062 | murf | 2007-11-06 14:08:38 -0700 (Tue, 06 Nov 2007) | 9 lines Merged revisions 89036 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89036 | murf | 2007-11-06 10:52:50 -0700 (Tue, 06 Nov 2007) | 1 line closes issue ASTERISK-8547 - where the [catname](!) and [catname](othercat1,othercat2,...) notation gets dropped across a ConfigUpdate (or any other thing that would cause a config file to be written). While I was at it, I also cleaned up some of the destroy routines to free up comments, which was not being done. Made sure the new struct I introduced is also cleaned up properly at destruction time. My code handles multiple template inclusions. Many thanks to ssokol for his patch, which, while not literally used in the final merge, served as a foundation for the fix. ........ ................ ------------------------------------------------------------------------ By: Iñaki Baz Castillo (ibc) 2007-11-20 08:07:24.000-0600 Good job, it works in the test I've done. ;) By: Iñaki Baz Castillo (ibc) 2007-11-20 08:12:38.000-0600 Im' sorry, I selected "reopen" because I went to report a bug about this, but it was a failure of mine in phone configuration. It works in fact. Please, close again the bug. Sorry. |