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Merged revisions 171187 via svnmerge from

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https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r171187 | tilghman | 2009-01-25 17:44:01 -0600 (Sun, 25 Jan 2009) | 6 lines
  
  Correctly track the hookstate
  (closes issue #13686)
   Reported by: itiliti
   Patches: 
         20081013__bug13686.diff.txt uploaded by Corydon76 (license 14)
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@171188 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Tilghman Lesher
2009-01-25 23:58:00 +00:00
parent aada4991c1
commit b0a29390ec
1 changed files with 11 additions and 10 deletions
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21
channels/chan_oss.c
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@@ -255,9 +255,9 @@ struct chan_oss_pvt {
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 */
@@ -289,8 +289,8 @@ struct chan_oss_pvt {
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 */
@@ -332,8 +332,7 @@ static struct chan_oss_pvt oss_default = {
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);
@@ -622,6 +621,7 @@ static int oss_call(struct ast_channel *c, char *dest, int timeout)
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;
@@ -637,8 +637,10 @@ static int oss_call(struct ast_channel *c, char *dest, int timeout)
*/
static int oss_answer(struct ast_channel *c)
{
struct chan_oss_pvt *o = c->tech_pvt;
ast_verbose(" << Console call has been answered >> \n");
ast_setstate(c, AST_STATE_UP);
o->hookstate = 1;
return 0;
}
@@ -821,7 +823,6 @@ static struct ast_channel *oss_new(struct chan_oss_pvt *o, char *ext, char *ctx,
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 ? */
}
}
console_video_start(get_video_desc(c), c); /* XXX cleanup */
@@ -1067,7 +1068,7 @@ static char *console_flash(struct ast_cli_entry *e, int cmd, struct ast_cli_args
return CLI_FAILURE;
}
o->hookstate = 0;
if (o->owner) /* XXX must be true, right ? */
if (o->owner)
ast_queue_frame(o->owner, &f);
return CLI_SUCCESS;
}
@@ -1483,7 +1484,7 @@ static int unload_module(void)
close(o->sounddev);
if (o->owner)
ast_softhangup(o->owner, AST_SOFTHANGUP_APPUNLOAD);
if (o->owner) /* XXX how ??? */
if (o->owner)
return -1;
next = o->next;
ast_free(o->name);