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Make sure we turn on the DSP when we answer the call 

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@87231 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Matthew Fredrickson 2007-10-27 00:43:59 +00:00
parent a55b6954e8
commit 6c452a6408
1 changed files with 9 additions and 1 deletions
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10
channels/chan_zap.c
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@ -5604,7 +5604,7 @@ static struct ast_channel *zt_new(struct zt_pvt *i, int state, int startpbx, int
i->dsp = NULL;
if (i->dsp) {
i->dsp_features = features & ~DSP_PROGRESS_TALK;
#ifdef HAVE_PRI
#if defined(HAVE_PRI) || defined(HAVE_SS7)
/* We cannot do progress detection until receives PROGRESS message */
if (i->outgoing && ((i->sig == SIG_PRI) || (i->sig == SIG_SS7))) {
/* Remember requested DSP features, don't treat
@ -8699,6 +8699,10 @@ static void *ss7_linkset(void *data)
ast_debug(1, "Queuing frame PROGRESS on CIC %d\n", p->cic);
zap_queue_frame(p, &f, linkset);
p->progress = 1;
if (p->dsp && p->dsp_features) {
ast_dsp_set_features(p->dsp, p->dsp_features);
p->dsp_features = 0;
}
}
break;
default:
@ -8978,6 +8982,10 @@ static void *ss7_linkset(void *data)
p = linkset->pvts[chanpos];
ast_mutex_lock(&p->lock);
p->subs[SUB_REAL].needanswer = 1;
if (p->dsp && p->dsp_features) {
ast_dsp_set_features(p->dsp, p->dsp_features);
p->dsp_features = 0;
}
zt_enable_ec(p);
ast_mutex_unlock(&p->lock);
}