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Fixed so that dial from a Zap channel to a Zap channel in 'dataquality' mode actually puts channels into CLEAR mode (so that 56k ISDN calls will work thru it) 64K calls STILL DONT.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@677 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Jim Dixon
2003-03-23 17:14:29 +00:00
parent 445bf6c4f1
commit 63d49a667e
3 changed files with 25 additions and 2 deletions
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6
apps/app_dial.c
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@@ -545,12 +545,16 @@ static int dial_exec(struct ast_channel *chan, void *data)
int x = 2;
if (tmp->dataquality) 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;
@@ -573,7 +577,7 @@ 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);
res = ast_bridge_call(chan, peer, allowredir, allowdisconnect | tmp->dataquality);
ast_hangup(peer);
}
out: