[asterisk-commits] qwell: branch 1.4 r113118 - in /branches/1.4: channels/ configs/

[SVN commits to the Asterisk project]

Author: qwell
Date: Mon Apr  7 13:00:09 2008
New Revision: 113118

URL: http://svn.digium.com/view/asterisk?view=rev&rev=113118
Log:
Allow playback with noanswer (and add earlyrtp option).

(closes issue #9077)
Reported by: pj
Patches:
      earlyrtp.diff uploaded by wedhorn (license 30)
Tested by: pj, qwell, DEA, wedhorn

Modified:
    branches/1.4/channels/chan_skinny.c
    branches/1.4/configs/skinny.conf.sample

Modified: branches/1.4/channels/chan_skinny.c
URL: http://svn.digium.com/view/asterisk/branches/1.4/channels/chan_skinny.c?view=diff&rev=113118&r1=113117&r2=113118
==============================================================================
--- branches/1.4/channels/chan_skinny.c (original)
+++ branches/1.4/channels/chan_skinny.c Mon Apr  7 13:00:09 2008
@@ -1028,6 +1028,7 @@
 	int lastlineinstance;
 	int lastcallreference;
 	int capability;
+	int earlyrtp;
 	char exten[AST_MAX_EXTENSION];
 	struct sockaddr_in addr;
 	struct in_addr ourip;
@@ -2100,6 +2101,7 @@
 		d->lastlineinstance = 1;
 		d->capability = default_capability;
 		d->prefs = default_prefs;
+		d->earlyrtp = 1;
 		while(v) {
 			if (!strcasecmp(v->name, "host")) {
 				if (ast_get_ip(&d->addr, v->value)) {
@@ -2120,6 +2122,8 @@
 				ast_parse_allow_disallow(&d->prefs, &d->capability, v->value, 0);
 			} else if (!strcasecmp(v->name, "version")) {
 				ast_copy_string(d->version_id, v->value, sizeof(d->version_id));
+			} else if (!strcasecmp(v->name, "earlyrtp")) {
+				d->earlyrtp = ast_true(v->value);
 			} else if (!strcasecmp(v->name, "nat")) {
 				nat = ast_true(v->value);
 			} else if (!strcasecmp(v->name, "callerid")) {
@@ -2322,6 +2326,9 @@
 		l->hidecallerid ? "" : l->cid_name,
 		c->cid.cid_ani ? NULL : l->cid_num);
 	ast_setstate(c, AST_STATE_RING);
+	if (!sub->rtp) {
+		start_rtp(sub);
+	}
 	res = ast_pbx_run(c);
 	if (res) {
 		ast_log(LOG_WARNING, "PBX exited non-zero\n");
@@ -2746,45 +2753,61 @@
 	case AST_CONTROL_RINGING:
 		if (ast->_state != AST_STATE_UP) {
 			if (!sub->progress) {
-				transmit_tone(s, SKINNY_ALERT, l->instance, sub->callid);
+				if (!d->earlyrtp) {
+					transmit_tone(s, SKINNY_ALERT, l->instance, sub->callid);
+				}
 				transmit_callstate(s, l->instance, SKINNY_RINGOUT, sub->callid);
 				transmit_dialednumber(s, exten, l->instance, sub->callid);
 				transmit_displaypromptstatus(s, "Ring Out", 0, l->instance, sub->callid);
 				transmit_callinfo(s, ast->cid.cid_name, ast->cid.cid_num, exten, exten, l->instance, sub->callid, 2); /* 2 = outgoing from phone */
 				sub->ringing = 1;
-				break;
+				if (!d->earlyrtp) {
+					break;
+				}
 			}
 		}
-		return -1;
+		return -1; /* Tell asterisk to provide inband signalling */
 	case AST_CONTROL_BUSY:
 		if (ast->_state != AST_STATE_UP) {
-			transmit_tone(s, SKINNY_BUSYTONE, l->instance, sub->callid);
+			if (!d->earlyrtp) {
+				transmit_tone(s, SKINNY_BUSYTONE, l->instance, sub->callid);
+			}
 			transmit_callstate(s, l->instance, SKINNY_BUSY, sub->callid);
 			sub->alreadygone = 1;
 			ast_softhangup_nolock(ast, AST_SOFTHANGUP_DEV);
-			break;
-		}
-		return -1;
+			if (!d->earlyrtp) {
+				break;
+			}
+		}
+		return -1; /* Tell asterisk to provide inband signalling */
 	case AST_CONTROL_CONGESTION:
 		if (ast->_state != AST_STATE_UP) {
-			transmit_tone(s, SKINNY_REORDER, l->instance, sub->callid);
+			if (!d->earlyrtp) {
+				transmit_tone(s, SKINNY_REORDER, l->instance, sub->callid);
+			}
 			transmit_callstate(s, l->instance, SKINNY_CONGESTION, sub->callid);
 			sub->alreadygone = 1;
 			ast_softhangup_nolock(ast, AST_SOFTHANGUP_DEV);
-			break;
-		}
-		return -1;
+			if (!d->earlyrtp) {
+				break;
+			}
+		}
+		return -1; /* Tell asterisk to provide inband signalling */
 	case AST_CONTROL_PROGRESS:
 		if ((ast->_state != AST_STATE_UP) && !sub->progress && !sub->outgoing) {
-			transmit_tone(s, SKINNY_ALERT, l->instance, sub->callid);
+			if (!d->earlyrtp) {
+				transmit_tone(s, SKINNY_ALERT, l->instance, sub->callid);
+			}
 			transmit_callstate(s, l->instance, SKINNY_PROGRESS, sub->callid);
 			transmit_displaypromptstatus(s, "Call Progress", 0, l->instance, sub->callid);
 			transmit_callinfo(s, ast->cid.cid_name, ast->cid.cid_num, exten, exten, l->instance, sub->callid, 2); /* 2 = outgoing from phone */
 			sub->progress = 1;
-			break;
-		}
-		return -1;
-	case -1:
+			if (!d->earlyrtp) {
+				break;
+			}
+		}
+		return -1; /* Tell asterisk to provide inband signalling */
+	case -1:  /* STOP_TONE */
 		transmit_tone(s, SKINNY_SILENCE, l->instance, sub->callid);
 		break;
 	case AST_CONTROL_HOLD:
@@ -2800,7 +2823,7 @@
 		break;
 	default:
 		ast_log(LOG_WARNING, "Don't know how to indicate condition %d\n", ind);
-		return -1;
+		return -1; /* Tell asterisk to provide inband signalling */
 	}
 	return 0;
 }

Modified: branches/1.4/configs/skinny.conf.sample
URL: http://svn.digium.com/view/asterisk/branches/1.4/configs/skinny.conf.sample?view=diff&rev=113118&r1=113117&r2=113118
==============================================================================
--- branches/1.4/configs/skinny.conf.sample (original)
+++ branches/1.4/configs/skinny.conf.sample Mon Apr  7 13:00:09 2008
@@ -39,6 +39,11 @@
 ;jblog = no                  ; Enables jitterbuffer frame logging. Defaults to "no".
 ;-----------------------------------------------------------------------------------
 
+;----------------------------------- DEVICE OPTIONS --------------------------------
+;earlyrtp=1                  ; whether audio signalling should be provided by asterisk
+                             ; (earlyrtp=1) or device generated (earlyrtp=0). 
+                             ; defaults to earlyrtp=1
+;-----------------------------------------------------------------------------------
 
 ; Typical config for 12SP+
 ;[florian]