[asterisk-commits] rmudgett: branch 1.8 r293807 - in /branches/1.8: ./ channels/
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Wed Nov 3 13:35:24 CDT 2010
Author: rmudgett
Date: Wed Nov 3 13:35:19 2010
New Revision: 293807
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=293807
Log:
Merged revisions 293806 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................
r293806 | rmudgett | 2010-11-03 13:31:57 -0500 (Wed, 03 Nov 2010) | 27 lines
Merged revisions 293805 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r293805 | rmudgett | 2010-11-03 13:23:04 -0500 (Wed, 03 Nov 2010) | 20 lines
Party A in an analog 3-way call would continue to hear ringback after party C answers.
All parties are analog FXS ports.
1) A calls B.
2) A flash hooks to call C.
3) A flash hooks to bring C into 3-way call before C answers. (A and B hear ringback)
4) C answers
5) A continues to hear ringback during the 3-way call. (All parties can hear each other.)
* Fixed use of wrong variable in dahdi_bridge() that stopped ringback on
the wrong subchannel.
* Made several debug messages have more information.
A similar issue happens if B and C are SIP channels. B continues to hear
ringback. For some reason this only affects v1.8 and trunk.
* Don't start ringback on the real and 3-way subchannels when creating the
3-way conference. Removing this code is benign on v1.6.2 and earlier.
........
................
Modified:
branches/1.8/ (props changed)
branches/1.8/channels/chan_dahdi.c
branches/1.8/channels/sig_analog.c
Propchange: branches/1.8/
------------------------------------------------------------------------------
Binary property 'branch-1.6.2-merged' - no diff available.
Modified: branches/1.8/channels/chan_dahdi.c
URL: http://svnview.digium.com/svn/asterisk/branches/1.8/channels/chan_dahdi.c?view=diff&rev=293807&r1=293806&r2=293807
==============================================================================
--- branches/1.8/channels/chan_dahdi.c (original)
+++ branches/1.8/channels/chan_dahdi.c Wed Nov 3 13:35:19 2010
@@ -7082,11 +7082,14 @@
p1->subs[SUB_REAL].owner &&
p1->subs[SUB_REAL].inthreeway &&
(p1->subs[SUB_REAL].owner->_state == AST_STATE_RINGING)) {
- ast_debug(1, "Playing ringback on %s since %s is in a ringing three-way\n", c0->name, c1->name);
+ ast_debug(1,
+ "Playing ringback on %d/%d(%s) since %d/%d(%s) is in a ringing three-way\n",
+ p0->channel, oi0, c0->name, p1->channel, oi1, c1->name);
tone_zone_play_tone(p0->subs[oi0].dfd, DAHDI_TONE_RINGTONE);
os1 = p1->subs[SUB_REAL].owner->_state;
} else {
- ast_debug(1, "Stopping tones on %d/%d talking to %d/%d\n", p0->channel, oi0, p1->channel, oi1);
+ ast_debug(1, "Stopping tones on %d/%d(%s) talking to %d/%d(%s)\n",
+ p0->channel, oi0, c0->name, p1->channel, oi1, c1->name);
tone_zone_play_tone(p0->subs[oi0].dfd, -1);
}
if ((oi0 == SUB_THREEWAY) &&
@@ -7094,12 +7097,15 @@
p0->subs[SUB_REAL].owner &&
p0->subs[SUB_REAL].inthreeway &&
(p0->subs[SUB_REAL].owner->_state == AST_STATE_RINGING)) {
- ast_debug(1, "Playing ringback on %s since %s is in a ringing three-way\n", c1->name, c0->name);
+ ast_debug(1,
+ "Playing ringback on %d/%d(%s) since %d/%d(%s) is in a ringing three-way\n",
+ p1->channel, oi1, c1->name, p0->channel, oi0, c0->name);
tone_zone_play_tone(p1->subs[oi1].dfd, DAHDI_TONE_RINGTONE);
os0 = p0->subs[SUB_REAL].owner->_state;
} else {
- ast_debug(1, "Stopping tones on %d/%d talking to %d/%d\n", p1->channel, oi1, p0->channel, oi0);
- tone_zone_play_tone(p1->subs[oi0].dfd, -1);
+ ast_debug(1, "Stopping tones on %d/%d(%s) talking to %d/%d(%s)\n",
+ p1->channel, oi1, c1->name, p0->channel, oi0, c0->name);
+ tone_zone_play_tone(p1->subs[oi1].dfd, -1);
}
if ((oi0 == SUB_REAL) && (oi1 == SUB_REAL)) {
if (!p0->echocanbridged || !p1->echocanbridged) {
@@ -8215,7 +8221,9 @@
(p->transfertobusy || (ast->_state != AST_STATE_BUSY))) {
int otherindex = SUB_THREEWAY;
- ast_verb(3, "Building conference on call on %s and %s\n", p->subs[SUB_THREEWAY].owner->name, p->subs[SUB_REAL].owner->name);
+ ast_verb(3, "Building conference call with %s and %s\n",
+ p->subs[SUB_THREEWAY].owner->name,
+ p->subs[SUB_REAL].owner->name);
/* Put them in the threeway, and flip */
p->subs[SUB_THREEWAY].inthreeway = 1;
p->subs[SUB_REAL].inthreeway = 1;
@@ -8227,11 +8235,6 @@
ast_queue_control(p->subs[otherindex].owner, AST_CONTROL_UNHOLD);
p->subs[otherindex].needunhold = 1;
p->owner = p->subs[SUB_REAL].owner;
- if (ast->_state == AST_STATE_RINGING) {
- ast_debug(1, "Enabling ringtone on real and threeway\n");
- res = tone_zone_play_tone(p->subs[SUB_REAL].dfd, DAHDI_TONE_RINGTONE);
- res = tone_zone_play_tone(p->subs[SUB_THREEWAY].dfd, DAHDI_TONE_RINGTONE);
- }
} else {
ast_verb(3, "Dumping incomplete call on on %s\n", p->subs[SUB_THREEWAY].owner->name);
swap_subs(p, SUB_THREEWAY, SUB_REAL);
Modified: branches/1.8/channels/sig_analog.c
URL: http://svnview.digium.com/svn/asterisk/branches/1.8/channels/sig_analog.c?view=diff&rev=293807&r1=293806&r2=293807
==============================================================================
--- branches/1.8/channels/sig_analog.c (original)
+++ branches/1.8/channels/sig_analog.c Wed Nov 3 13:35:19 2010
@@ -3170,7 +3170,9 @@
/* Lets see what we're up to */
if (((ast->pbx) || (ast->_state == AST_STATE_UP)) &&
(p->transfertobusy || (ast->_state != AST_STATE_BUSY))) {
- ast_verb(3, "Building conference on call on %s and %s\n", p->subs[ANALOG_SUB_THREEWAY].owner->name, p->subs[ANALOG_SUB_REAL].owner->name);
+ ast_verb(3, "Building conference call with %s and %s\n",
+ p->subs[ANALOG_SUB_THREEWAY].owner->name,
+ p->subs[ANALOG_SUB_REAL].owner->name);
/* Put them in the threeway, and flip */
analog_set_inthreeway(p, ANALOG_SUB_THREEWAY, 1);
analog_set_inthreeway(p, ANALOG_SUB_REAL, 1);
@@ -3182,11 +3184,6 @@
ast_queue_control(p->subs[orig_3way_sub].owner, AST_CONTROL_UNHOLD);
}
p->owner = p->subs[ANALOG_SUB_REAL].owner;
- if (ast->_state == AST_STATE_RINGING) {
- ast_debug(1, "Enabling ringtone on real and threeway\n");
- analog_play_tone(p, ANALOG_SUB_REAL, ANALOG_TONE_RINGTONE);
- analog_play_tone(p, ANALOG_SUB_THREEWAY, ANALOG_TONE_RINGTONE);
- }
} else {
ast_verb(3, "Dumping incomplete call on %s\n", p->subs[ANALOG_SUB_THREEWAY].owner->name);
analog_swap_subs(p, ANALOG_SUB_THREEWAY, ANALOG_SUB_REAL);
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