Process audio stuff in audio mode and then switch back to clear
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@893 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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@ -1455,17 +1455,16 @@ static int zt_hangup(struct ast_channel *ast)
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index = zt_get_index(ast, p, 1);
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if (p->sig == SIG_PRI) {
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x = 0;
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x = 1;
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ast_channel_setoption(ast,AST_OPTION_AUDIO_MODE,&x,sizeof(char),0);
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}
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restore_gains(p);
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if (p->dsp)
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ast_dsp_digitmode(p->dsp,DSP_DIGITMODE_DTMF | p->dtmfrelax);
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x = 0;
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zt_confmute(p, 0);
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restore_gains(p);
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ast_dsp_digitmode(p->dsp,DSP_DIGITMODE_DTMF | p->dtmfrelax);
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ast_log(LOG_DEBUG, "Hangup: channel: %d index = %d, normal = %d, callwait = %d, thirdcall = %d\n",
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p->channel, index, p->subs[SUB_REAL].zfd, p->subs[SUB_CALLWAIT].zfd, p->subs[SUB_THREEWAY].zfd);
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@ -1654,6 +1653,13 @@ static int zt_hangup(struct ast_channel *ast)
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p->dialing = 0;
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strcpy(p->rdnis, "");
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update_conf(p);
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/* Restore data mode */
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if (p->sig == SIG_PRI) {
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x = 0;
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ast_channel_setoption(ast,AST_OPTION_AUDIO_MODE,&x,sizeof(char),0);
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}
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if (p->dsp)
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restart_monitor();
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}
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@ -3528,11 +3534,6 @@ static struct ast_channel *zt_new(struct zt_pvt *i, int state, int startpbx, int
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#ifdef ZAPATA_PRI
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/* Assume calls are not idle calls unless we're told differently */
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i->isidlecall = 0;
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if (i->sig == SIG_PRI) {
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/* Set to audio mode at this poitn mode */
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x = 1;
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ast_channel_setoption(tmp,AST_OPTION_AUDIO_MODE,&x,sizeof(char),0);
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}
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#endif
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/* Assure there is no confmute on this channel */
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zt_confmute(i, 0);
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@ -5529,6 +5530,9 @@ static void *pri_dchannel(void *vpri)
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if (strlen(pri->pvt[chan]->exten) && ast_exists_extension(NULL, pri->pvt[chan]->context, pri->pvt[chan]->exten, 1, pri->pvt[chan]->callerid)) {
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/* Setup law */
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int law;
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/* Set to audio mode at this poitn mode */
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law = 1;
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ast_channel_setoption(tmp,AST_OPTION_AUDIO_MODE,&x,sizeof(char),0);
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if (e->ring.layer1 == PRI_LAYER_1_ALAW)
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law = ZT_LAW_ALAW;
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else
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