[asterisk-commits] tilghman: trunk r171188 - in /trunk: ./ channels/chan_oss.c
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Sun Jan 25 17:58:01 CST 2009
Author: tilghman
Date: Sun Jan 25 17:58:00 2009
New Revision: 171188
URL: http://svn.digium.com/svn-view/asterisk?view=rev&rev=171188
Log:
Merged revisions 171187 via svnmerge from
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)
........
Modified:
trunk/ (props changed)
trunk/channels/chan_oss.c
Propchange: trunk/
------------------------------------------------------------------------------
Binary property 'branch-1.4-merged' - no diff available.
Modified: trunk/channels/chan_oss.c
URL: http://svn.digium.com/svn-view/asterisk/trunk/channels/chan_oss.c?view=diff&rev=171188&r1=171187&r2=171188
==============================================================================
--- trunk/channels/chan_oss.c (original)
+++ trunk/channels/chan_oss.c Sun Jan 25 17:58:00 2009
@@ -255,9 +255,9 @@
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 */
@@ -289,8 +289,8 @@
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 */
@@ -332,8 +332,7 @@
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);
@@ -622,6 +621,7 @@
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;
@@ -637,8 +637,10 @@
*/
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 @@
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 @@
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 @@
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);
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