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

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@1804 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/apps/app_dial.c b/apps/app_dial.c
index b59e0aa37b377212531e2bed512d83c405d1db9b..33293fe73cb62fc16f4c7587dbf75753a6d124af 100755 (executable)
--- a/apps/app_dial.c
+++ b/apps/app_dial.c
@@ -62,8 +62,6 @@ static char *descrip =
 "      'T' -- to allow the calling user to transfer the call.\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"
-"      '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"
@@ -85,7 +83,6 @@ struct localuser {
        int allowredirect_out;
        int ringbackonly;
        int musiconhold;
-       int dataquality;
        int allowdisconnect;
        struct localuser *next;
 };
@@ -350,7 +347,6 @@ static int dial_exec(struct ast_channel *chan, void *data)
        int privacy=0;
        int announce=0;
        int resetcdr=0;
-       int clearchannel=0;
        int cnt=0;
        char numsubst[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'))
                                tmp->musiconhold = 1;
                         else    tmp->musiconhold = 0;
-                       if (strchr(transfer, 'd'))
-                               tmp->dataquality = 1;
-                        else    tmp->dataquality = 0;
                        if (strchr(transfer, 'H'))
                                allowdisconnect = tmp->allowdisconnect = 1;
                         else    allowdisconnect = tmp->allowdisconnect = 0;
-                       if (strchr(transfer, 'c'))
-                               clearchannel = 1;
-            else    
-                               clearchannel = 0;
                        if(strchr(transfer, 'g'))
                                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
                   we will always return with -1 so that it is hung up properly after the 
                   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);
                outgoing = NULL;
                /* 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_channel_sendurl( peer, url );
                } /* /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)
                {
                        int res2;
@@ -699,13 +670,7 @@ static int dial_exec(struct ast_channel *chan, void *data)
                        // Ok, done. stop autoservice
                        res2 = ast_autoservice_stop(chan);
                }
-               res = ast_bridge_call(chan, peer, allowredir_in, allowredir_out, allowdisconnect | clearchannel);
-               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);
-               }
+               res = ast_bridge_call(chan, peer, allowredir_in, allowredir_out, allowdisconnect);
 
                if (res != AST_PBX_NO_HANGUP_PEER)
                        ast_hangup(peer);