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https://github.com/asterisk/asterisk.git
synced 2025-09-06 04:30:28 +00:00
Remove access to free'd memory fro dude's code
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@691 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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@@ -83,7 +83,6 @@ struct localuser {
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int ringbackonly;
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int ringbackonly;
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int musiconhold;
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int musiconhold;
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int dataquality;
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int dataquality;
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int clearchannel;
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int allowdisconnect;
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int allowdisconnect;
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struct localuser *next;
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struct localuser *next;
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};
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};
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@@ -299,6 +298,7 @@ static int dial_exec(struct ast_channel *chan, void *data)
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int allowdisconnect=0;
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int allowdisconnect=0;
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int privacy=0;
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int privacy=0;
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int resetcdr=0;
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int resetcdr=0;
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int clearchannel=0;
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char numsubst[AST_MAX_EXTENSION];
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char numsubst[AST_MAX_EXTENSION];
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char restofit[AST_MAX_EXTENSION];
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char restofit[AST_MAX_EXTENSION];
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char *transfer = NULL;
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char *transfer = NULL;
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@@ -430,8 +430,9 @@ static int dial_exec(struct ast_channel *chan, void *data)
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tmp->allowdisconnect = 1;
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tmp->allowdisconnect = 1;
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else tmp->allowdisconnect = 0;
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else tmp->allowdisconnect = 0;
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if (strchr(transfer, 'c'))
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if (strchr(transfer, 'c'))
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tmp->clearchannel = 1;
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clearchannel = 1;
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else tmp->clearchannel = 0;
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else
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clearchannel = 0;
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}
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}
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strncpy(numsubst, number, sizeof(numsubst)-1);
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strncpy(numsubst, number, sizeof(numsubst)-1);
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/* If we're dialing by extension, look at the extension to know what to dial */
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/* If we're dialing by extension, look at the extension to know what to dial */
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@@ -548,13 +549,13 @@ static int dial_exec(struct ast_channel *chan, void *data)
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if (!strcmp(chan->type,"Zap"))
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if (!strcmp(chan->type,"Zap"))
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{
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{
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int x = 2;
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int x = 2;
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if (tmp->dataquality || tmp->clearchannel) x = 0;
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if (tmp->dataquality || clearchannel) x = 0;
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ast_channel_setoption(chan,AST_OPTION_TONE_VERIFY,&x,sizeof(char),0);
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ast_channel_setoption(chan,AST_OPTION_TONE_VERIFY,&x,sizeof(char),0);
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}
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}
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if (!strcmp(peer->type,"Zap"))
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if (!strcmp(peer->type,"Zap"))
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{
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{
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int x = 2;
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int x = 2;
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if (tmp->dataquality || tmp->clearchannel) x = 0;
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if (tmp->dataquality || clearchannel) x = 0;
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ast_channel_setoption(peer,AST_OPTION_TONE_VERIFY,&x,sizeof(char),0);
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ast_channel_setoption(peer,AST_OPTION_TONE_VERIFY,&x,sizeof(char),0);
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}
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}
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hanguptree(outgoing, peer);
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hanguptree(outgoing, peer);
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@@ -578,14 +579,14 @@ static int dial_exec(struct ast_channel *chan, void *data)
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ast_log(LOG_DEBUG, "app_dial: sendurl=%s.\n", url);
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ast_log(LOG_DEBUG, "app_dial: sendurl=%s.\n", url);
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ast_channel_sendurl( peer, url );
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ast_channel_sendurl( peer, url );
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} /* /JDG */
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} /* /JDG */
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if (tmp->clearchannel)
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if (clearchannel)
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{
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{
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int x = 0;
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int x = 0;
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ast_channel_setoption(chan,AST_OPTION_AUDIO_MODE,&x,sizeof(char),0);
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ast_channel_setoption(chan,AST_OPTION_AUDIO_MODE,&x,sizeof(char),0);
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ast_channel_setoption(peer,AST_OPTION_AUDIO_MODE,&x,sizeof(char),0);
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ast_channel_setoption(peer,AST_OPTION_AUDIO_MODE,&x,sizeof(char),0);
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}
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}
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res = ast_bridge_call(chan, peer, allowredir, allowdisconnect | tmp->clearchannel);
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res = ast_bridge_call(chan, peer, allowredir, allowdisconnect | tmp->clearchannel);
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if (tmp->clearchannel)
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if (clearchannel)
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{
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{
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int x = 1;
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int x = 1;
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ast_channel_setoption(chan,AST_OPTION_AUDIO_MODE,&x,sizeof(char),0);
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ast_channel_setoption(chan,AST_OPTION_AUDIO_MODE,&x,sizeof(char),0);
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