diff options
author | Jim Dixon <telesistant@hotmail.com> | 2003-11-28 04:38:07 +0000 |
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committer | Jim Dixon <telesistant@hotmail.com> | 2003-11-28 04:38:07 +0000 |
commit | e1f471f89c8c9f7138a2a33ddec0886faa9df858 (patch) | |
tree | d09c226cb52f77b6021b94025cbd98d7a6cb4fc2 /apps | |
parent | 9ccfcb3d242821a2e89f58269771063b858746b9 (diff) |
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
Diffstat (limited to 'apps')
-rwxr-xr-x | apps/app_dial.c | 37 |
1 files changed, 1 insertions, 36 deletions
diff --git a/apps/app_dial.c b/apps/app_dial.c index b59e0aa37..33293fe73 100755 --- 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); |