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Stop caller id transmission when offhook event detected.

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This fixes the problem that would occur if an analog phone was picked up while
the caller id was being sent. The caller id before sent the whole spill even
after pickup and is now corrected.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@217744 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Jeff Peeler
2009-09-10 20:18:30 +00:00
parent 1fc7fb0241
commit 5561ba19aa
3 changed files with 26 additions and 0 deletions
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12
channels/sig_analog.c
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@@ -771,6 +771,15 @@ static int analog_check_confirmanswer(struct analog_pvt *p)
return 0;
}
static void analog_cancel_cidspill(struct analog_pvt *p)
{
if (!p->calls->cancel_cidspill) {
return;
}
p->calls->cancel_cidspill(p->chan_pvt);
}
static int analog_set_linear_mode(struct analog_pvt *p, int index, int linear_mode)
{
if (p->calls->set_linear_mode) {
@@ -2541,6 +2550,7 @@ static struct ast_frame *__analog_handle_event(struct analog_pvt *p, struct ast_
/* Make sure it stops ringing */
analog_off_hook(p);
ast_debug(1, "channel %d answered\n", p->channel);
analog_cancel_cidspill(p);
analog_set_dialing(p, 0);
p->callwaitcas = 0;
if (analog_check_confirmanswer(p)) {
@@ -2662,6 +2672,7 @@ static struct ast_frame *__analog_handle_event(struct analog_pvt *p, struct ast_
}
if (ast->rings > p->cidrings) {
analog_cancel_cidspill(p);
p->callwaitcas = 0;
}
p->subs[index].f.frametype = AST_FRAME_CONTROL;
@@ -3187,6 +3198,7 @@ int analog_handle_init_event(struct analog_pvt *i, int event)
if (res && (errno == EBUSY)) {
break;
}
analog_cancel_cidspill(i);
if (i->immediate) {
analog_set_echocanceller(i, 1);
/* The channel is immediately up. Start right away */