diff options
| author | Jim Dixon <telesistant@hotmail.com> | 2003-03-23 18:55:52 +0000 |
|---|---|---|
| committer | Jim Dixon <telesistant@hotmail.com> | 2003-03-23 18:55:52 +0000 |
| commit | e2c23ff3dbc9bc6163a0a4b0188467bb2f807e21 (patch) | |
| tree | e2086cfe150289c198d3de8f215ab10d8b1d6265 /apps | |
| parent | 63d49a667e802238e813f258b64618549d3293b4 (diff) | |
Fixed more stuff for clearchannel mode in app_dial
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@678 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Diffstat (limited to 'apps')
| -rwxr-xr-x | apps/app_dial.c | 25 |
1 files changed, 19 insertions, 6 deletions
diff --git a/apps/app_dial.c b/apps/app_dial.c index e68f5137e..472f14436 100755 --- a/apps/app_dial.c +++ b/apps/app_dial.c @@ -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: |