chan_dahdi/sig_pri: Prevent unnecessary PROGRESS events when overlap dialing is enabled.

When overlap dialing is enabled, the lack of inband audio available
information in the SETUP_ACKNOWLEDGE events causes an interoperability
problem with SIP.  sig_pri doesn't know if there is dialtone present when
a SETUP_ACKNOWLEDGE is received so it assumes it is there and posts an
AST_CONTROL_PROGRESS frame.  The SIP channel driver then sends out a 183
Session Progress and blocks the desired 180 Ringing message when the
ALERTING message comes in.

* Made the configure script detect if the installed version of libpri
supports the SETUP_ACKNOWLEDGE enhancements.

* Using the new API, made generate an AST_CONTROL_PROGRESS frame on an
incoming SETUP_ACKNOWLEDGE message when the message indicates inband audio
is present instead of assuming that dialtone is present.

* Using the new API, made SETUP_ACKNOWLEDGE send out an inband audio
available indication only if dialtone is expected.  The change also makes
the fallback behaviour of sending the PROGRESS message better by sending
it only if dialtone is expected.

* Changed receiving a PROCEEDING message to not generate an
AST_CONTROL_PROGRESS frame if the progress indication ie indicates
non-end-to-end-ISDN.  This helps interoperability with SIP.

* Changed sending a PROCEEDING message in response to an
AST_CONTROL_PROCEEDING frame to not indicate inband audio available.  It
was silly to do so anyway because the channel driver doesn't know if
inband audio is even available.  This helps interoperability with SIP.

This patch and a corresponding change in libpri work together to allow
Asterisk to control the inband audio available progress indication ie on
the SETUP_ACKNOWLEDGE message when dialtone is present.

AST-1338 #close
Reported by: Tyler Stewart

Review: https://reviewboard.asterisk.org/r/3521/
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Merged revisions 413714 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 413765 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 413771 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413772 65c4cc65-6c06-0410-ace0-fbb531ad65f3

1 files changed, 65 insertions, 22 deletions

diff --git a/channels/sig_pri.c b/channels/sig_pri.c
index 025117c8d..d153da1db 100644
--- a/channels/sig_pri.c
+++ b/channels/sig_pri.c
@@ -2137,10 +2137,10 @@ static void *pri_ss_thread(void *data)
 		/* Start the real PBX */
 		ast_channel_exten_set(chan, exten);
 		sig_pri_dsp_reset_and_flush_digits(p);
-#if defined(ISSUE_16789)
+#if defined(JIRA_ASTERISK_15594)
 		/*
-		 * Conditionaled out this code to effectively revert the Mantis
-		 * issue 16789 change.  It breaks overlap dialing through
+		 * Conditionaled out this code to effectively revert the JIRA
+		 * ASTERISK-15594 change.  It breaks overlap dialing through
 		 * Asterisk.  There is not enough information available at this
 		 * point to know if dialing is complete.  The
 		 * ast_exists_extension(), ast_matchmore_extension(), and
@@ -2163,7 +2163,7 @@ static void *pri_ss_thread(void *data)
 			}
 			sig_pri_unlock_private(p);
 		}
-#endif	/* defined(ISSUE_16789) */
+#endif	/* defined(JIRA_ASTERISK_15594) */
 
 		sig_pri_set_echocanceller(p, 1);
 		ast_channel_lock(chan);
@@ -6435,6 +6435,9 @@ static void *pri_dchannel(void *vpri)
 				/* Make sure extension exists (or in overlap dial mode, can exist) */
 				if (((pri->overlapdial & DAHDI_OVERLAPDIAL_INCOMING) && ast_canmatch_extension(NULL, pri->pvts[chanpos]->context, pri->pvts[chanpos]->exten, 1, pri->pvts[chanpos]->cid_num)) ||
 					ast_exists_extension(NULL, pri->pvts[chanpos]->context, pri->pvts[chanpos]->exten, 1, pri->pvts[chanpos]->cid_num)) {
+					int could_match_more;
+					int need_dialtone;
+
 					/* Select audio companding mode. */
 					switch (e->ring.layer1) {
 					case PRI_LAYER_1_ALAW:
@@ -6449,6 +6452,23 @@ static void *pri_dchannel(void *vpri)
 						break;
 					}
 
+					could_match_more = !e->ring.complete
+						&& (pri->overlapdial & DAHDI_OVERLAPDIAL_INCOMING)
+						&& ast_matchmore_extension(NULL, pri->pvts[chanpos]->context,
+							pri->pvts[chanpos]->exten, 1, pri->pvts[chanpos]->cid_num);
+
+					need_dialtone = could_match_more
+						/*
+						 * Must explicitly check the digital capability this
+						 * way instead of checking the pvt->digital flag
+						 * because the flag hasn't been set yet.
+						 */
+						&& !(e->ring.ctype & AST_TRANS_CAP_DIGITAL)
+						&& !pri->pvts[chanpos]->no_b_channel
+						&& (!strlen(pri->pvts[chanpos]->exten)
+							|| ast_ignore_pattern(pri->pvts[chanpos]->context,
+								pri->pvts[chanpos]->exten));
+
 					if (e->ring.complete || !(pri->overlapdial & DAHDI_OVERLAPDIAL_INCOMING)) {
 						/* Just announce proceeding */
 						pri->pvts[chanpos]->call_level = SIG_PRI_CALL_LEVEL_PROCEEDING;
@@ -6458,13 +6478,17 @@ static void *pri_dchannel(void *vpri)
 						pri_answer(pri->pri, e->ring.call, PVT_TO_CHANNEL(pri->pvts[chanpos]), 1);
 					} else {
 						pri->pvts[chanpos]->call_level = SIG_PRI_CALL_LEVEL_OVERLAP;
-						pri_need_more_info(pri->pri, e->ring.call, PVT_TO_CHANNEL(pri->pvts[chanpos]), 1);
+#if defined(HAVE_PRI_SETUP_ACK_INBAND)
+						pri_setup_ack(pri->pri, e->ring.call,
+							PVT_TO_CHANNEL(pri->pvts[chanpos]), 1, need_dialtone);
+#else	/* !defined(HAVE_PRI_SETUP_ACK_INBAND) */
+						pri_need_more_info(pri->pri, e->ring.call,
+							PVT_TO_CHANNEL(pri->pvts[chanpos]), 1);
+#endif	/* !defined(HAVE_PRI_SETUP_ACK_INBAND) */
 					}
 
 					/* Start PBX */
-					if (!e->ring.complete
-						&& (pri->overlapdial & DAHDI_OVERLAPDIAL_INCOMING)
-						&& ast_matchmore_extension(NULL, pri->pvts[chanpos]->context, pri->pvts[chanpos]->exten, 1, pri->pvts[chanpos]->cid_num)) {
+					if (could_match_more) {
 						/*
 						 * Release the PRI lock while we create the channel so other
 						 * threads can send D channel messages.  We must also release
@@ -6549,12 +6573,9 @@ static void *pri_dchannel(void *vpri)
 							sig_pri_handle_subcmds(pri, chanpos, e->e, e->ring.subcmds,
 								e->ring.call);
 
-							if (!pri->pvts[chanpos]->digital
-								&& !pri->pvts[chanpos]->no_b_channel) {
-								/*
-								 * Call has a channel.
-								 * Indicate that we are providing dialtone.
-								 */
+#if !defined(HAVE_PRI_SETUP_ACK_INBAND)
+							if (need_dialtone) {
+								/* Indicate that we are providing dialtone. */
 								pri->pvts[chanpos]->progress = 1;/* No need to send plain PROGRESS again. */
 #ifdef HAVE_PRI_PROG_W_CAUSE
 								pri_progress_with_cause(pri->pri, e->ring.call,
@@ -6564,6 +6585,7 @@ static void *pri_dchannel(void *vpri)
 									PVT_TO_CHANNEL(pri->pvts[chanpos]), 1);
 #endif
 							}
+#endif	/* !defined(HAVE_PRI_SETUP_ACK_INBAND) */
 							ast_channel_stage_snapshot_done(c);
 						}
 						if (c && !ast_pthread_create_detached(&threadid, NULL, pri_ss_thread, pri->pvts[chanpos])) {
@@ -6866,8 +6888,15 @@ static void *pri_dchannel(void *vpri)
 				if (!pri->pvts[chanpos]->progress
 					&& !pri->pvts[chanpos]->no_b_channel
 #ifdef PRI_PROGRESS_MASK
-					&& (e->proceeding.progressmask
-						& (PRI_PROG_CALL_NOT_E2E_ISDN | PRI_PROG_INBAND_AVAILABLE))
+					/*
+					 * We only care about PRI_PROG_INBAND_AVAILABLE to open the
+					 * voice path.
+					 *
+					 * We explicitly DO NOT want to check PRI_PROG_CALL_NOT_E2E_ISDN
+					 * because it will mess up ISDN to SIP interoperability for
+					 * the ALERTING message.
+					 */
+					&& (e->proceeding.progressmask & PRI_PROG_INBAND_AVAILABLE)
 #else
 					&& e->proceeding.progress == 8
 #endif
@@ -6878,6 +6907,12 @@ static void *pri_dchannel(void *vpri)
 					sig_pri_set_dialing(pri->pvts[chanpos], 0);
 					sig_pri_open_media(pri->pvts[chanpos]);
 				} else if (pri->inband_on_proceeding) {
+					/*
+					 * XXX This is to accomodate a broken switch that sends a
+					 * PROCEEDING without any progress indication ie for
+					 * inband audio.  This should be part of the conditional
+					 * test above to bring the voice path up.
+					 */
 					sig_pri_set_dialing(pri->pvts[chanpos], 0);
 				}
 				sig_pri_unlock_private(pri->pvts[chanpos]);
@@ -7496,7 +7531,19 @@ static void *pri_dchannel(void *vpri)
 				if (!pri->pvts[chanpos]->progress
 					&& (pri->overlapdial & DAHDI_OVERLAPDIAL_OUTGOING)
 					&& !pri->pvts[chanpos]->digital
-					&& !pri->pvts[chanpos]->no_b_channel) {
+					&& !pri->pvts[chanpos]->no_b_channel
+#if defined(HAVE_PRI_SETUP_ACK_INBAND)
+					/*
+					 * We only care about PRI_PROG_INBAND_AVAILABLE to open the
+					 * voice path.
+					 *
+					 * We explicitly DO NOT want to check PRI_PROG_CALL_NOT_E2E_ISDN
+					 * because it will mess up ISDN to SIP interoperability for
+					 * the ALERTING message.
+					 */
+					&& (e->setup_ack.progressmask & PRI_PROG_INBAND_AVAILABLE)
+#endif	/* defined(HAVE_PRI_SETUP_ACK_INBAND) */
+					) {
 					/*
 					 * Call has a channel.
 					 * Indicate for overlap dialing that dialtone may be present.
@@ -8374,11 +8421,7 @@ int sig_pri_indicate(struct sig_pri_chan *p, struct ast_channel *chan, int condi
 			p->call_level = SIG_PRI_CALL_LEVEL_PROCEEDING;
 			if (p->pri && p->pri->pri) {
 				pri_grab(p, p->pri);
-				pri_proceeding(p->pri->pri,p->call, PVT_TO_CHANNEL(p),
-					p->no_b_channel || p->digital ? 0 : 1);
-				if (!p->no_b_channel && !p->digital) {
-					sig_pri_set_dialing(p, 0);
-				}
+				pri_proceeding(p->pri->pri,p->call, PVT_TO_CHANNEL(p), 0);
 				pri_rel(p->pri);
 			}
 		}