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Fix Dial m and r options and forked calls generating warnings for voice frames.

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When connected line support was added, the wait_for_answer() variable
single changed its meaning slightly.  Unfortunately, the places where
single was used did not necessarily get updated to reflect that change.
Also audio/video frames were sent to all forked calls when the endpoints
were never made compatible.

* Don't pass audio/video media frames when the channels have not been made
compatible.

* Added handling of AST_CONTROL_SRCCHANGE to app_dial.c.

* Fixed app_dial.c passing on AST_CONTROL_HOLD because that frame can also
pass a requested MOH class.

(closes issue ASTERISK-16901)
Reported by: Chris Gentle

(closes issue ASTERISK-17541)
Reported by: clint

Review: https://reviewboard.asterisk.org/r/1805/
........

Merged revisions 359344 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 359355 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359357 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Richard Mudgett
2012-03-14 17:39:45 +00:00
parent 6df0ae5c1f
commit 2019a7e6b9
2 changed files with 103 additions and 46 deletions
Download Patch File Download Diff File Expand all files Collapse all files

133
apps/app_dial.c
View File

@@ -823,13 +823,21 @@ static void senddialendevent(struct ast_channel *src, const char *dialstatus)
/*!
* helper function for wait_for_answer()
*
* \param o Outgoing call channel list.
* \param num Incoming call channel cause accumulation
* \param peerflags Dial option flags
* \param single_caller_bored From wait_for_answer: single && !caller_entertained
* \param to Remaining call timeout time.
* \param forced_clid OPT_FORCECLID caller id to send
* \param stored_clid Caller id representing the called party if needed
*
* XXX this code is highly suspicious, as it essentially overwrites
* the outgoing channel without properly deleting it.
*
* \todo eventually this function should be intergrated into and replaced by ast_call_forward()
* \todo eventually this function should be intergrated into and replaced by ast_call_forward()
*/
static void do_forward(struct chanlist *o,
struct cause_args *num, struct ast_flags64 *peerflags, int single, int *to,
struct cause_args *num, struct ast_flags64 *peerflags, int single_caller_bored, int *to,
struct ast_party_id *forced_clid, struct ast_party_id *stored_clid)
{
char tmpchan[256];
@@ -871,8 +879,9 @@ static void do_forward(struct chanlist *o,
/* Setup parameters */
c = o->chan = ast_request(tech, ast_channel_nativeformats(in), in, stuff, &cause);
if (c) {
if (single)
if (single_caller_bored) {
ast_channel_make_compatible(o->chan, in);
}
ast_channel_inherit_variables(in, o->chan);
ast_channel_datastore_inherit(in, o->chan);
/* When a call is forwarded, we don't want to track new interfaces
@@ -893,7 +902,7 @@ static void do_forward(struct chanlist *o,
} else {
struct ast_party_redirecting redirecting;
if (single && CAN_EARLY_BRIDGE(peerflags, c, in)) {
if (single_caller_bored && CAN_EARLY_BRIDGE(peerflags, c, in)) {
ast_rtp_instance_early_bridge_make_compatible(c, in);
}
@@ -987,7 +996,7 @@ static void do_forward(struct chanlist *o,
/* Hangup the original channel now, in case we needed it */
ast_hangup(original);
}
if (single) {
if (single_caller_bored) {
ast_indicate(in, -1);
}
}
@@ -1016,6 +1025,8 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in,
struct ast_channel *peer = NULL;
/* single is set if only one destination is enabled */
int single = outgoing && !outgoing->next;
int caller_entertained = outgoing
&& ast_test_flag64(outgoing, OPT_MUSICBACK | OPT_RINGBACK);
#ifdef HAVE_EPOLL
struct chanlist *epollo;
#endif
@@ -1029,7 +1040,7 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in,
ast_party_connected_line_init(&connected_caller);
if (single) {
/* Turn off hold music, etc */
if (!ast_test_flag64(outgoing, OPT_MUSICBACK | OPT_RINGBACK)) {
if (!caller_entertained) {
ast_deactivate_generator(in);
/* If we are calling a single channel, and not providing ringback or music, */
/* then, make them compatible for in-band tone purpose */
@@ -1154,7 +1165,8 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in,
}
ast_frfree(f);
}
do_forward(o, &num, peerflags, single, to, forced_clid, stored_clid);
do_forward(o, &num, peerflags, single && !caller_entertained, to,
forced_clid, stored_clid);
continue;
}
f = ast_read(winner);
@@ -1169,7 +1181,8 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in,
handle_cause(ast_channel_hangupcause(in), &num);
continue;
}
if (f->frametype == AST_FRAME_CONTROL) {
switch (f->frametype) {
case AST_FRAME_CONTROL:
switch (f->subclass.integer) {
case AST_CONTROL_ANSWER:
/* This is our guy if someone answered. */
@@ -1266,8 +1279,10 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in,
if (ignore_cc || cc_frame_received || num_ringing == numlines) {
ast_verb(3, "%s is ringing\n", ast_channel_name(c));
/* Setup early media if appropriate */
if (single && CAN_EARLY_BRIDGE(peerflags, in, c))
if (single && !caller_entertained
&& CAN_EARLY_BRIDGE(peerflags, in, c)) {
ast_channel_early_bridge(in, c);
}
if (!(pa->sentringing) && !ast_test_flag64(outgoing, OPT_MUSICBACK) && ast_strlen_zero(opt_args[OPT_ARG_RINGBACK])) {
ast_indicate(in, AST_CONTROL_RINGING);
pa->sentringing++;
@@ -1277,8 +1292,10 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in,
case AST_CONTROL_PROGRESS:
ast_verb(3, "%s is making progress passing it to %s\n", ast_channel_name(c), ast_channel_name(in));
/* Setup early media if appropriate */
if (single && CAN_EARLY_BRIDGE(peerflags, in, c))
if (single && !caller_entertained
&& CAN_EARLY_BRIDGE(peerflags, in, c)) {
ast_channel_early_bridge(in, c);
}
if (!ast_test_flag64(outgoing, OPT_RINGBACK)) {
if (single || (!single && !pa->sentringing)) {
ast_indicate(in, AST_CONTROL_PROGRESS);
@@ -1292,12 +1309,14 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in,
}
break;
case AST_CONTROL_VIDUPDATE:
ast_verb(3, "%s requested a video update, passing it to %s\n", ast_channel_name(c), ast_channel_name(in));
ast_indicate(in, AST_CONTROL_VIDUPDATE);
break;
case AST_CONTROL_SRCUPDATE:
ast_verb(3, "%s requested a source update, passing it to %s\n", ast_channel_name(c), ast_channel_name(in));
ast_indicate(in, AST_CONTROL_SRCUPDATE);
case AST_CONTROL_SRCCHANGE:
if (!single || caller_entertained) {
break;
}
ast_verb(3, "%s requested media update control %d, passing it to %s\n",
ast_channel_name(c), f->subclass.integer, ast_channel_name(in));
ast_indicate(in, f->subclass.integer);
break;
case AST_CONTROL_CONNECTED_LINE:
if (ast_test_flag64(peerflags, OPT_IGNORE_CONNECTEDLINE)) {
@@ -1342,16 +1361,20 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in,
break;
case AST_CONTROL_PROCEEDING:
ast_verb(3, "%s is proceeding passing it to %s\n", ast_channel_name(c), ast_channel_name(in));
if (single && CAN_EARLY_BRIDGE(peerflags, in, c))
if (single && !caller_entertained
&& CAN_EARLY_BRIDGE(peerflags, in, c)) {
ast_channel_early_bridge(in, c);
}
if (!ast_test_flag64(outgoing, OPT_RINGBACK))
ast_indicate(in, AST_CONTROL_PROCEEDING);
break;
case AST_CONTROL_HOLD:
/* XXX this should be saved like AST_CONTROL_CONNECTED_LINE for !single || caller_entertained */
ast_verb(3, "Call on %s placed on hold\n", ast_channel_name(c));
ast_indicate(in, AST_CONTROL_HOLD);
ast_indicate_data(in, AST_CONTROL_HOLD, f->data.ptr, f->datalen);
break;
case AST_CONTROL_UNHOLD:
/* XXX this should be saved like AST_CONTROL_CONNECTED_LINE for !single || caller_entertained */
ast_verb(3, "Call on %s left from hold\n", ast_channel_name(c));
ast_indicate(in, AST_CONTROL_UNHOLD);
break;
@@ -1366,7 +1389,7 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in,
}
break;
case -1:
if (!ast_test_flag64(outgoing, OPT_RINGBACK | OPT_MUSICBACK)) {
if (single && !caller_entertained) {
ast_verb(3, "%s stopped sounds\n", ast_channel_name(c));
ast_indicate(in, -1);
pa->sentringing = 0;
@@ -1374,26 +1397,29 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in,
break;
default:
ast_debug(1, "Dunno what to do with control type %d\n", f->subclass.integer);
}
} else if (single) {
switch (f->frametype) {
case AST_FRAME_VOICE:
case AST_FRAME_IMAGE:
case AST_FRAME_TEXT:
if (!ast_test_flag64(outgoing, OPT_RINGBACK | OPT_MUSICBACK) && ast_write(in, f)) {
ast_log(LOG_WARNING, "Unable to write frametype: %d\n",
f->frametype);
}
break;
case AST_FRAME_HTML:
if (!ast_test_flag64(outgoing, DIAL_NOFORWARDHTML)
&& ast_channel_sendhtml(in, f->subclass.integer, f->data.ptr, f->datalen) == -1) {
ast_log(LOG_WARNING, "Unable to send URL\n");
}
break;
default:
break;
}
break;
case AST_FRAME_VOICE:
case AST_FRAME_IMAGE:
if (caller_entertained) {
break;
}
/* Fall through */
case AST_FRAME_TEXT:
if (single && ast_write(in, f)) {
ast_log(LOG_WARNING, "Unable to write frametype: %d\n",
f->frametype);
}
break;
case AST_FRAME_HTML:
if (single && !ast_test_flag64(outgoing, DIAL_NOFORWARDHTML)
&& ast_channel_sendhtml(in, f->subclass.integer, f->data.ptr, f->datalen) == -1) {
ast_log(LOG_WARNING, "Unable to send URL\n");
}
break;
default:
break;
}
ast_frfree(f);
} /* end for */
@@ -1474,6 +1500,15 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in,
break;
case AST_FRAME_VOICE:
case AST_FRAME_IMAGE:
if (!single || caller_entertained) {
/*
* We are calling multiple parties or caller is being
* entertained and has thus not been made compatible.
* No need to check any other called parties.
*/
goto skip_frame;
}
/* Fall through */
case AST_FRAME_TEXT:
case AST_FRAME_DTMF_BEGIN:
case AST_FRAME_DTMF_END:
@@ -1485,12 +1520,27 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in,
case AST_FRAME_CONTROL:
switch (f->subclass.integer) {
case AST_CONTROL_HOLD:
ast_verb(3, "Call on %s placed on hold\n", ast_channel_name(o->chan));
ast_indicate_data(o->chan, AST_CONTROL_HOLD, f->data.ptr, f->datalen);
break;
case AST_CONTROL_UNHOLD:
ast_verb(3, "Call on %s left from hold\n", ast_channel_name(o->chan));
ast_indicate(o->chan, AST_CONTROL_UNHOLD);
break;
case AST_CONTROL_VIDUPDATE:
case AST_CONTROL_SRCUPDATE:
ast_verb(3, "%s requested special control %d, passing it to %s\n",
case AST_CONTROL_SRCCHANGE:
if (!single || caller_entertained) {
/*
* We are calling multiple parties or caller is being
* entertained and has thus not been made compatible.
* No need to check any other called parties.
*/
goto skip_frame;
}
ast_verb(3, "%s requested media update control %d, passing it to %s\n",
ast_channel_name(in), f->subclass.integer, ast_channel_name(o->chan));
ast_indicate_data(o->chan, f->subclass.integer, f->data.ptr, f->datalen);
ast_indicate(o->chan, f->subclass.integer);
break;
case AST_CONTROL_CONNECTED_LINE:
if (ast_channel_connected_line_sub(in, o->chan, f, 1) &&
@@ -1505,13 +1555,16 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in,
}
break;
default:
break;
/* We are not going to do anything with this frame. */
goto skip_frame;
}
break;
default:
break;
/* We are not going to do anything with this frame. */
goto skip_frame;
}
}
skip_frame:;
ast_frfree(f);
}
if (!*to)

16
main/channel.c
View File

@@ -4730,15 +4730,19 @@ int ast_write(struct ast_channel *chan, struct ast_frame *fr)
if (ast_format_cmp(&fr->subclass.format, ast_channel_rawwriteformat(chan)) != AST_FORMAT_CMP_NOT_EQUAL) {
f = fr;
} else {
/* XXX Something is not right we are not compatible with this frame bad things can happen
* problems range from no/one-way audio to unexplained line hangups as a last resort try adjust the format
* ideally we do not want to do this and this indicates a deeper problem for now we log these events to
* eliminate user impact and help identify the problem areas
* JIRA issues related to this :-
* ASTERISK-14384, ASTERISK-17502, ASTERISK-17541, ASTERISK-18063, ASTERISK-18325, ASTERISK-18422*/
if ((!ast_format_cap_iscompatible(ast_channel_nativeformats(chan), &fr->subclass.format)) &&
(ast_format_cmp(ast_channel_writeformat(chan), &fr->subclass.format) != AST_FORMAT_CMP_EQUAL)) {
char nf[512];
/*
* XXX Something is not right. We are not compatible with this
* frame. Bad things can happen. Problems range from no audio,
* one-way audio, to unexplained line hangups. As a last resort
* try to adjust the format. Ideally, we do not want to do this
* because it indicates a deeper problem. For now, we log these
* events to reduce user impact and help identify the problem
* areas.
*/
ast_log(LOG_WARNING, "Codec mismatch on channel %s setting write format to %s from %s native formats %s\n",
ast_channel_name(chan), ast_getformatname(&fr->subclass.format), ast_getformatname(ast_channel_writeformat(chan)),
ast_getformatname_multiple(nf, sizeof(nf), ast_channel_nativeformats(chan)));