From e1503e9af7afffb74ccfda2fdad8a1e4ca49ae53 Mon Sep 17 00:00:00 2001 From: Mark Spencer <markster@digium.com> Date: Wed, 26 Mar 2003 00:30:54 +0000 Subject: [PATCH] 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 --- apps/app_dial.c | 15 ++++++++------- 1 file changed, 8 insertions(+), 7 deletions(-) diff --git a/apps/app_dial.c b/apps/app_dial.c index eff1214b38..e64a81cdaa 100755 --- a/apps/app_dial.c +++ b/apps/app_dial.c @@ -83,7 +83,6 @@ struct localuser { int ringbackonly; int musiconhold; int dataquality; - int clearchannel; int allowdisconnect; struct localuser *next; }; @@ -299,6 +298,7 @@ static int dial_exec(struct ast_channel *chan, void *data) int allowdisconnect=0; int privacy=0; int resetcdr=0; + int clearchannel=0; char numsubst[AST_MAX_EXTENSION]; char restofit[AST_MAX_EXTENSION]; char *transfer = NULL; @@ -430,8 +430,9 @@ static int dial_exec(struct ast_channel *chan, void *data) tmp->allowdisconnect = 1; else tmp->allowdisconnect = 0; if (strchr(transfer, 'c')) - tmp->clearchannel = 1; - else tmp->clearchannel = 0; + clearchannel = 1; + else + clearchannel = 0; } strncpy(numsubst, number, sizeof(numsubst)-1); /* 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")) { 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); } if (!strcmp(peer->type,"Zap")) { 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); } 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_channel_sendurl( peer, url ); } /* /JDG */ - if (tmp->clearchannel) + 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); } res = ast_bridge_call(chan, peer, allowredir, allowdisconnect | tmp->clearchannel); - if (tmp->clearchannel) + if (clearchannel) { int x = 1; ast_channel_setoption(chan,AST_OPTION_AUDIO_MODE,&x,sizeof(char),0); -- GitLab