[asterisk-commits] tilghman: branch 1.4 r171187 - /branches/1.4/channels/chan_oss.c

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Sun Jan 25 17:44:02 CST 2009


Author: tilghman
Date: Sun Jan 25 17:44:01 2009
New Revision: 171187

URL: http://svn.digium.com/svn-view/asterisk?view=rev&rev=171187
Log:
Correctly track the hookstate
(closes issue #13686)
 Reported by: itiliti
 Patches: 
       20081013__bug13686.diff.txt uploaded by Corydon76 (license 14)

Modified:
    branches/1.4/channels/chan_oss.c

Modified: branches/1.4/channels/chan_oss.c
URL: http://svn.digium.com/svn-view/asterisk/branches/1.4/channels/chan_oss.c?view=diff&rev=171187&r1=171186&r2=171187
==============================================================================
--- branches/1.4/channels/chan_oss.c (original)
+++ branches/1.4/channels/chan_oss.c Sun Jan 25 17:44:01 2009
@@ -336,9 +336,9 @@
 	int total_blocks;			/* total blocks in the output device */
 	int sounddev;
 	enum { M_UNSET, M_FULL, M_READ, M_WRITE } duplex;
-	int autoanswer;
-	int autohangup;
-	int hookstate;
+	int autoanswer;             /*!< Boolean: whether to answer the immediately upon calling */
+	int autohangup;             /*!< Boolean: whether to hangup the call when the remote end hangs up */
+	int hookstate;              /*!< Boolean: 1 if offhook; 0 if onhook */
 	char *mixer_cmd;			/* initial command to issue to the mixer */
 	unsigned int queuesize;		/* max fragments in queue */
 	unsigned int frags;			/* parameter for SETFRAGMENT */
@@ -367,8 +367,8 @@
 	char ext[AST_MAX_EXTENSION];
 	char ctx[AST_MAX_CONTEXT];
 	char language[MAX_LANGUAGE];
-	char cid_name[256];			/*XXX */
-	char cid_num[256];			/*XXX */
+	char cid_name[256];         /*!< Initial CallerID name */
+	char cid_num[256];          /*!< Initial CallerID number  */
 	char mohinterpret[MAX_MUSICCLASS];
 
 	/* buffers used in oss_write */
@@ -401,8 +401,7 @@
 
 static int setformat(struct chan_oss_pvt *o, int mode);
 
-static struct ast_channel *oss_request(const char *type, int format, void *data
-, int *cause);
+static struct ast_channel *oss_request(const char *type, int format, void *data, int *cause);
 static int oss_digit_begin(struct ast_channel *c, char digit);
 static int oss_digit_end(struct ast_channel *c, char digit, unsigned int duration);
 static int oss_text(struct ast_channel *c, const char *text);
@@ -808,6 +807,7 @@
 		f.frametype = AST_FRAME_CONTROL;
 		f.subclass = AST_CONTROL_ANSWER;
 		ast_queue_frame(c, &f);
+		o->hookstate = 1;
 	} else {
 		ast_verbose("<< Type 'answer' to answer, or use 'autoanswer' for future calls >> \n");
 		f.frametype = AST_FRAME_CONTROL;
@@ -833,6 +833,7 @@
 	ast_setstate(c, AST_STATE_UP);
 	o->cursound = -1;
 	o->nosound = 0;
+	o->hookstate = 1;
 	return 0;
 }
 
@@ -1028,8 +1029,6 @@
 			ast_log(LOG_WARNING, "Unable to start PBX on %s\n", c->name);
 			ast_hangup(c);
 			o->owner = c = NULL;
-			/* XXX what about the channel itself ? */
-			/* XXX what about usecnt ? */
 		}
 	}
 
@@ -1296,7 +1295,7 @@
 		return RESULT_FAILURE;
 	}
 	o->hookstate = 0;
-	if (o->owner) /* XXX must be true, right ? */
+	if (o->owner)
 		ast_queue_frame(o->owner, &f);
 	return RESULT_SUCCESS;
 }
@@ -1315,7 +1314,7 @@
 		return RESULT_FAILURE;
 	}
 	o->hookstate = 0;
-	if (o->owner)				/* XXX must be true, right ? */
+	if (o->owner)
 		ast_queue_frame(o->owner, &f);
 	return RESULT_SUCCESS;
 }
@@ -1888,7 +1887,7 @@
 		}
 		if (o->owner)
 			ast_softhangup(o->owner, AST_SOFTHANGUP_APPUNLOAD);
-		if (o->owner)			/* XXX how ??? */
+		if (o->owner)
 			return -1;
 		/* XXX what about the thread ? */
 		/* XXX what about the memory allocated ? */




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