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asterisk
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Remove access to free'd memory fro dude's code

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@691 65c4cc65-6c06-0410-ace0-fbb531ad65f3
1.0
Mark Spencer 23 years ago
parent d391e86036
commit e1503e9af7
1 changed files with 8 additions and 7 deletions
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15
apps/app_dial.c
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@ -83,7 +83,6 @@ struct localuser {
int ringbackonly; int ringbackonly;
int musiconhold; int musiconhold;
int dataquality; int dataquality;
int clearchannel;
int allowdisconnect; int allowdisconnect;
struct localuser *next; struct localuser *next;
}; };
@ -299,6 +298,7 @@ static int dial_exec(struct ast_channel *chan, void *data)
int allowdisconnect=0; int allowdisconnect=0;
int privacy=0; int privacy=0;
int resetcdr=0; int resetcdr=0;
int clearchannel=0;
char numsubst[AST_MAX_EXTENSION]; char numsubst[AST_MAX_EXTENSION];
char restofit[AST_MAX_EXTENSION]; char restofit[AST_MAX_EXTENSION];
char *transfer = NULL; char *transfer = NULL;
@ -430,8 +430,9 @@ static int dial_exec(struct ast_channel *chan, void *data)
tmp->allowdisconnect = 1; tmp->allowdisconnect = 1;
else tmp->allowdisconnect = 0; else tmp->allowdisconnect = 0;
if (strchr(transfer, 'c')) if (strchr(transfer, 'c'))
tmp->clearchannel = 1; clearchannel = 1;
else tmp->clearchannel = 0; else
clearchannel = 0;
} }
strncpy(numsubst, number, sizeof(numsubst)-1); strncpy(numsubst, number, sizeof(numsubst)-1);
/* If we're dialing by extension, look at the extension to know what to dial */ /* If we're dialing by extension, look at the extension to know what to dial */
@ -548,13 +549,13 @@ static int dial_exec(struct ast_channel *chan, void *data)
if (!strcmp(chan->type,"Zap")) if (!strcmp(chan->type,"Zap"))
{ {
int x = 2; int x = 2;
if (tmp->dataquality || tmp->clearchannel) x = 0; if (tmp->dataquality || clearchannel) x = 0;
ast_channel_setoption(chan,AST_OPTION_TONE_VERIFY,&x,sizeof(char),0); ast_channel_setoption(chan,AST_OPTION_TONE_VERIFY,&x,sizeof(char),0);
} }
if (!strcmp(peer->type,"Zap")) if (!strcmp(peer->type,"Zap"))
{ {
int x = 2; int x = 2;
if (tmp->dataquality || tmp->clearchannel) x = 0; if (tmp->dataquality || clearchannel) x = 0;
ast_channel_setoption(peer,AST_OPTION_TONE_VERIFY,&x,sizeof(char),0); ast_channel_setoption(peer,AST_OPTION_TONE_VERIFY,&x,sizeof(char),0);
} }
hanguptree(outgoing, peer); hanguptree(outgoing, peer);
@ -578,14 +579,14 @@ static int dial_exec(struct ast_channel *chan, void *data)
ast_log(LOG_DEBUG, "app_dial: sendurl=%s.\n", url); ast_log(LOG_DEBUG, "app_dial: sendurl=%s.\n", url);
ast_channel_sendurl( peer, url ); ast_channel_sendurl( peer, url );
} /* /JDG */ } /* /JDG */
if (tmp->clearchannel) if (clearchannel)
{ {
int x = 0; int x = 0;
ast_channel_setoption(chan,AST_OPTION_AUDIO_MODE,&x,sizeof(char),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); ast_channel_setoption(peer,AST_OPTION_AUDIO_MODE,&x,sizeof(char),0);
} }
res = ast_bridge_call(chan, peer, allowredir, allowdisconnect | tmp->clearchannel); res = ast_bridge_call(chan, peer, allowredir, allowdisconnect | tmp->clearchannel);
if (tmp->clearchannel) if (clearchannel)
{ {
int x = 1; int x = 1;
ast_channel_setoption(chan,AST_OPTION_AUDIO_MODE,&x,sizeof(char),0); ast_channel_setoption(chan,AST_OPTION_AUDIO_MODE,&x,sizeof(char),0);

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