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Got rid of un-necessary 'c' and 'd' options in app_dial.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@1804 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Jim Dixon
2003-11-28 04:38:07 +00:00
parent 9ccfcb3d24
commit e1f471f89c
1 changed files with 1 additions and 36 deletions
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37
apps/app_dial.c
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@@ -62,8 +62,6 @@ static char *descrip =
" 'T' -- to allow the calling user to transfer the call.\n" " 'T' -- to allow the calling user to transfer the call.\n"
" 'r' -- indicate ringing to the calling party, pass no audio until answered.\n" " 'r' -- indicate ringing to the calling party, pass no audio until answered.\n"
" 'm' -- provide hold music to the calling party 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" " 'H' -- allow caller to hang up by hitting *.\n"
" 'C' -- reset call detail record for this call.\n" " 'C' -- reset call detail record for this call.\n"
" 'P[(x)]' -- privacy mode, using 'x' as database if provided.\n" " 'P[(x)]' -- privacy mode, using 'x' as database if provided.\n"
@@ -85,7 +83,6 @@ struct localuser {
int allowredirect_out; int allowredirect_out;
int ringbackonly; int ringbackonly;
int musiconhold; int musiconhold;
int dataquality;
int allowdisconnect; int allowdisconnect;
struct localuser *next; struct localuser *next;
}; };
@@ -350,7 +347,6 @@ static int dial_exec(struct ast_channel *chan, void *data)
int privacy=0; int privacy=0;
int announce=0; int announce=0;
int resetcdr=0; int resetcdr=0;
int clearchannel=0;
int cnt=0; int cnt=0;
char numsubst[AST_MAX_EXTENSION]; char numsubst[AST_MAX_EXTENSION];
char restofit[AST_MAX_EXTENSION]; char restofit[AST_MAX_EXTENSION];
@@ -490,16 +486,9 @@ static int dial_exec(struct ast_channel *chan, void *data)
if (strchr(transfer, 'm')) if (strchr(transfer, 'm'))
tmp->musiconhold = 1; tmp->musiconhold = 1;
else tmp->musiconhold = 0; else tmp->musiconhold = 0;
if (strchr(transfer, 'd'))
tmp->dataquality = 1;
else tmp->dataquality = 0;
if (strchr(transfer, 'H')) if (strchr(transfer, 'H'))
allowdisconnect = tmp->allowdisconnect = 1; allowdisconnect = tmp->allowdisconnect = 1;
else allowdisconnect = tmp->allowdisconnect = 0; else allowdisconnect = tmp->allowdisconnect = 0;
if (strchr(transfer, 'c'))
clearchannel = 1;
else
clearchannel = 0;
if(strchr(transfer, 'g')) if(strchr(transfer, 'g'))
go_on=1; go_on=1;
} }
@@ -647,18 +636,6 @@ static int dial_exec(struct ast_channel *chan, void *data)
/* Ah ha! Someone answered within the desired timeframe. Of course after this /* Ah ha! Someone answered within the desired timeframe. Of course after this
we will always return with -1 so that it is hung up properly after the we will always return with -1 so that it is hung up properly after the
conversation. */ conversation. */
if (!strcmp(chan->type,"Zap"))
{
int x = 2;
if (tmp->dataquality || clearchannel) x = 0;
ast_channel_setoption(chan,AST_OPTION_TONE_VERIFY,&x,sizeof(char),0);
}
if (!strcmp(peer->type,"Zap"))
{
int x = 2;
if (tmp->dataquality || clearchannel) x = 0;
ast_channel_setoption(peer,AST_OPTION_TONE_VERIFY,&x,sizeof(char),0);
}
hanguptree(outgoing, peer); hanguptree(outgoing, peer);
outgoing = NULL; outgoing = NULL;
/* If appropriate, log that we have a destination channel */ /* If appropriate, log that we have a destination channel */
@@ -680,12 +657,6 @@ static int dial_exec(struct ast_channel *chan, void *data)
ast_log(LOG_DEBUG, "app_dial: sendurl=%s.\n", url); ast_log(LOG_DEBUG, "app_dial: sendurl=%s.\n", url);
ast_channel_sendurl( peer, url ); ast_channel_sendurl( peer, url );
} /* /JDG */ } /* /JDG */
if (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);
}
if (announce && announcemsg) if (announce && announcemsg)
{ {
int res2; int res2;
@@ -699,13 +670,7 @@ static int dial_exec(struct ast_channel *chan, void *data)
// Ok, done. stop autoservice // Ok, done. stop autoservice
res2 = ast_autoservice_stop(chan); res2 = ast_autoservice_stop(chan);
} }
res = ast_bridge_call(chan, peer, allowredir_in, allowredir_out, allowdisconnect | clearchannel); res = ast_bridge_call(chan, peer, allowredir_in, allowredir_out, allowdisconnect);
if (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);
}
if (res != AST_PBX_NO_HANGUP_PEER) if (res != AST_PBX_NO_HANGUP_PEER)
ast_hangup(peer); ast_hangup(peer);