[asterisk-commits] kmoore: branch kmoore/stasis-channel_events-take2 r384883 - in /team/kmoore/s...
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Mon Apr 8 08:39:37 CDT 2013
Author: kmoore
Date: Mon Apr 8 08:39:34 2013
New Revision: 384883
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=384883
Log:
Multiple revisions 384828,384857,384879
........
r384828 | elguero | 2013-04-05 15:41:27 -0500 (Fri, 05 Apr 2013) | 29 lines
Fix For Not Overriding The Default Settings In chan_sip
The initial report was that the "nat" setting in the [general] section was not
having any effect in overriding the default setting. Upon confirming that this
was happening and looking into what was causing this, it was discovered that
other default settings would not be overriden as well.
This patch works similar to what occurs in build_peer(). We create a temporary
ast_flags structure and using a mask, we override the default settings with
whatever is set in the [general] section.
In the bug report, the reporter who helped to test this patch noted that the
directmedia settings were being overriden properly as well as the nat settings.
This issue is also present in Asterisk 1.8 and a separate patch will be applied
to it.
(issue ASTERISK-21225)
Reported by: Alexandre Vezina
Tested by: Alexandre Vezina, Michael L. Young
Patches:
asterisk-21225-handle-options-default-prob_v4.diff
Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/2385/
........
Merged revisions 384827 from http://svn.asterisk.org/svn/asterisk/branches/11
........
r384857 | file | 2013-04-06 11:00:20 -0500 (Sat, 06 Apr 2013) | 4 lines
Add a res_sorcery_astdb module which uses the astdb to persist objects.
Review: https://reviewboard.asterisk.org/r/2420/
........
r384879 | dlee | 2013-04-08 08:27:45 -0500 (Mon, 08 Apr 2013) | 26 lines
Stasis application WebSocket support
This is the API that binds the Stasis dialplan application to external
Stasis applications. It also adds the beginnings of WebSocket
application support.
This module registers a dialplan function named Stasis, which is used
to put a channel into the named Stasis app. As a channel enters and
leaves the Stasis diaplan applcation, the Stasis app receives a
'stasis-start' and 'stasis-end' events.
Stasis apps register themselves using the stasis_app_register and
stasis_app_unregister functions. Messages are sent to an appliction
using stasis_app_send.
Finally, Stasis apps control channels through the use of the
stasis_app_control object, and the family of stasis_app_control_*
functions.
Other changes along for the ride are:
* An ast_frame_dtor function that's RAII_VAR safe
* Some common JSON encoders for name/number, timeval, and
context/extension/priority
Review: https://reviewboard.asterisk.org/r/2361/
........
Merged revisions 384828,384857,384879 from http://svn.asterisk.org/svn/asterisk/trunk
Added:
team/kmoore/stasis-channel_events-take2/res/res_sorcery_astdb.c
- copied unchanged from r384879, trunk/res/res_sorcery_astdb.c
team/kmoore/stasis-channel_events-take2/tests/test_sorcery_astdb.c
- copied unchanged from r384879, trunk/tests/test_sorcery_astdb.c
Modified:
team/kmoore/stasis-channel_events-take2/ (props changed)
team/kmoore/stasis-channel_events-take2/UPGRADE-11.txt
team/kmoore/stasis-channel_events-take2/channels/chan_sip.c
Propchange: team/kmoore/stasis-channel_events-take2/
------------------------------------------------------------------------------
Binary property 'branch-11-merged' - no diff available.
Propchange: team/kmoore/stasis-channel_events-take2/
------------------------------------------------------------------------------
--- svnmerge-integrated (original)
+++ svnmerge-integrated Mon Apr 8 08:39:34 2013
@@ -1,1 +1,1 @@
-/trunk:1-384825
+/trunk:1-384880
Modified: team/kmoore/stasis-channel_events-take2/UPGRADE-11.txt
URL: http://svnview.digium.com/svn/asterisk/team/kmoore/stasis-channel_events-take2/UPGRADE-11.txt?view=diff&rev=384883&r1=384882&r2=384883
==============================================================================
--- team/kmoore/stasis-channel_events-take2/UPGRADE-11.txt (original)
+++ team/kmoore/stasis-channel_events-take2/UPGRADE-11.txt Mon Apr 8 08:39:34 2013
@@ -19,6 +19,30 @@
=== UPGRADE-10.txt -- Upgrade info for 1.8 to 10
===
===========================================================
+
+From 11.4 to 11.5:
+* The default settings for chan_sip are now overriden properly by the general
+ settings in sip.conf. Please look over your settings upon upgrading.
+
+From 11.3 to 11.4:
+* Added the 'n' option to MeetMe to prevent application of the DENOISE function
+ to a channel joining a conference. Some channel drivers that vary the number
+ of audio samples in a voice frame will experience significant quality problems
+ if a denoiser is attached to the channel; this option gives them the ability
+ to remove the denoiser without having to unload func_speex.
+
+* The Registry AMI event for SIP registrations will now always include the
+ Username field. A previous bug fix missed an instance where it was not
+ included; that has been corrected in this release.
+
+From 11.2.0 to 11.2.1:
+* Asterisk would previously not output certain error messages when a remote
+ console attempted to connect to Asterisk and no instance of Asterisk was
+ running. This error message is displayed on stderr; as a result, some
+ initialization scripts that used remote consoles to test for the presence
+ of a running Asterisk instance started to display erroneous error messages.
+ The init.d scripts and the safe_asterisk have been updated in the contrib
+ folder to account for this.
From 11.2 to 11.3:
Modified: team/kmoore/stasis-channel_events-take2/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/team/kmoore/stasis-channel_events-take2/channels/chan_sip.c?view=diff&rev=384883&r1=384882&r2=384883
==============================================================================
--- team/kmoore/stasis-channel_events-take2/channels/chan_sip.c (original)
+++ team/kmoore/stasis-channel_events-take2/channels/chan_sip.c Mon Apr 8 08:39:34 2013
@@ -31402,7 +31402,8 @@
struct sip_peer *peer;
char *cat, *stringp, *context, *oldregcontext;
char newcontexts[AST_MAX_CONTEXT], oldcontexts[AST_MAX_CONTEXT];
- struct ast_flags dummy[3];
+ struct ast_flags mask[3] = {{0}};
+ struct ast_flags setflags[3] = {{0}};
struct ast_flags config_flags = { (reason == CHANNEL_MODULE_LOAD || reason == CHANNEL_ACL_RELOAD) ? 0 : ast_test_flag(&global_flags[1], SIP_PAGE2_RTCACHEFRIENDS) ? 0 : CONFIG_FLAG_FILEUNCHANGED };
int auto_sip_domains = FALSE;
struct ast_sockaddr old_bindaddr = bindaddr;
@@ -31647,13 +31648,12 @@
ast_clear_flag(&global_flags[1], SIP_PAGE2_TEXTSUPPORT);
ast_clear_flag(&global_flags[1], SIP_PAGE2_IGNORESDPVERSION);
-
/* Read the [general] config section of sip.conf (or from realtime config) */
for (v = ast_variable_browse(cfg, "general"); v; v = v->next) {
- if (handle_common_options(&global_flags[0], &dummy[0], v)) {
+ if (handle_common_options(&setflags[0], &mask[0], v)) {
continue;
}
- if (handle_t38_options(&global_flags[0], &dummy[0], v, &global_t38_maxdatagram)) {
+ if (handle_t38_options(&setflags[0], &mask[0], v, &global_t38_maxdatagram)) {
continue;
}
/* handle jb conf */
@@ -32198,6 +32198,11 @@
global_refer_addheaders = ast_true(v->value);
}
}
+
+ /* Override global defaults if setting found in general section */
+ ast_copy_flags(&global_flags[0], &setflags[0], mask[0].flags);
+ ast_copy_flags(&global_flags[1], &setflags[1], mask[1].flags);
+ ast_copy_flags(&global_flags[2], &setflags[2], mask[2].flags);
/* For backwards compatibility the corresponding registration timer value is used if subscription timer value isn't set by configuration */
if (!min_subexpiry_set) {
More information about the asterisk-commits
mailing list