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Fixed more stuff for clearchannel mode in app_dial

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@678 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Jim Dixon
2003-03-23 18:55:52 +00:00
parent 63d49a667e
commit e2c23ff3db
2 changed files with 25 additions and 10 deletions
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25
apps/app_dial.c
View File

@@ -63,6 +63,7 @@ static char *descrip =
" 'r' -- indicate ringing to the calling party, pass no audio until answered.\n"
" 'm' -- provide hold music to the calling party until answered.\n"
" 'd' -- data-quality (modem) call (minimum delay).\n"
" 'c' -- clear-channel data call (PRI-PRI only).\n"
" 'H' -- allow caller to hang up by hitting *.\n"
" 'C' -- reset call detail record for this call.\n"
" 'P[(x)]' -- privacy mode, using 'x' as database if provided.\n"
@@ -82,6 +83,7 @@ struct localuser {
int ringbackonly;
int musiconhold;
int dataquality;
int clearchannel;
int allowdisconnect;
struct localuser *next;
};
@@ -427,6 +429,9 @@ static int dial_exec(struct ast_channel *chan, void *data)
if (strchr(transfer, 'H'))
tmp->allowdisconnect = 1;
else tmp->allowdisconnect = 0;
if (strchr(transfer, 'c'))
tmp->clearchannel = 1;
else tmp->clearchannel = 0;
}
strncpy(numsubst, number, sizeof(numsubst)-1);
/* If we're dialing by extension, look at the extension to know what to dial */
@@ -543,18 +548,14 @@ static int dial_exec(struct ast_channel *chan, void *data)
if (!strcmp(chan->type,"Zap"))
{
int x = 2;
if (tmp->dataquality) x = 0;
if (tmp->dataquality | tmp->clearchannel) x = 0;
ast_channel_setoption(chan,AST_OPTION_TONE_VERIFY,&x,sizeof(char),0);
x = 0;
ast_channel_setoption(chan,AST_OPTION_AUDIO_MODE,&x,sizeof(char),0);
}
if (!strcmp(peer->type,"Zap"))
{
int x = 2;
if (tmp->dataquality) x = 0;
ast_channel_setoption(peer,AST_OPTION_TONE_VERIFY,&x,sizeof(char),0);
x = 0;
ast_channel_setoption(chan,AST_OPTION_AUDIO_MODE,&x,sizeof(char),0);
}
hanguptree(outgoing, peer);
outgoing = NULL;
@@ -577,7 +578,19 @@ static int dial_exec(struct ast_channel *chan, void *data)
ast_log(LOG_DEBUG, "app_dial: sendurl=%s.\n", url);
ast_channel_sendurl( peer, url );
} /* /JDG */
res = ast_bridge_call(chan, peer, allowredir, allowdisconnect | tmp->dataquality);
if (tmp->clearchannel)
{
int x = 0;
ast_channel_setoption(chan,AST_OPTION_AUDIO_MODE,&x,sizeof(char),0);
ast_channel_setoption(peer,AST_OPTION_AUDIO_MODE,&x,sizeof(char),0);
}
res = ast_bridge_call(chan, peer, allowredir, allowdisconnect | tmp->clearchannel);
if (tmp->clearchannel)
{
int x = 1;
ast_channel_setoption(chan,AST_OPTION_AUDIO_MODE,&x,sizeof(char),0);
ast_channel_setoption(peer,AST_OPTION_AUDIO_MODE,&x,sizeof(char),0);
}
ast_hangup(peer);
}
out:

10
channels/chan_zap.c
View File

@@ -1433,6 +1433,9 @@ static int zt_hangup(struct ast_channel *ast)
index = zt_get_index(ast, p, 1);
x = 1;
ast_channel_setoption(ast,AST_OPTION_AUDIO_MODE,&x,sizeof(char),0);
restore_gains(p);
if (p->dsp)
@@ -1617,8 +1620,6 @@ static int zt_hangup(struct ast_channel *ast)
x = 0;
ast_channel_setoption(ast,AST_OPTION_TONE_VERIFY,&x,sizeof(char),0);
ast_channel_setoption(ast,AST_OPTION_TDD,&x,sizeof(char),0);
x = 1;
ast_channel_setoption(ast,AST_OPTION_AUDIO_MODE,&x,sizeof(char),0);
p->didtdd = 0;
p->cidspill = NULL;
p->callwaitcas = 0;
@@ -1864,6 +1865,7 @@ int x;
{
ast_log(LOG_DEBUG, "Set option AUDIO MODE, value: OFF(0) on %s\n",chan->name);
x = 0;
zt_disable_ec(p);
}
else
{
@@ -5461,10 +5463,10 @@ static void *pri_dchannel(void *vpri)
ast_verbose(VERBOSE_PREFIX_2 "Restart on requested on entire span %d\n", pri->span);
for (x=1;x <= pri->channels;x++)
if ((x != pri->dchannel) && pri->pvt[x]) {
ast_pthread_mutex_lock(&pri->pvt[chan]->lock);
ast_pthread_mutex_lock(&pri->pvt[x]->lock);
if (pri->pvt[x]->owner)
pri->pvt[x]->owner->_softhangup |= AST_SOFTHANGUP_DEV;
ast_pthread_mutex_unlock(&pri->pvt[chan]->lock);
ast_pthread_mutex_unlock(&pri->pvt[x]->lock);
}
}
break;