[asterisk-commits] dvossel: branch dvossel/gtalk_fixup r290610 - /team/dvossel/gtalk_fixup/chann...

[SVN commits to the Asterisk project]

Author: dvossel
Date: Wed Oct  6 11:10:58 2010
New Revision: 290610

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=290610
Log:
Send telephony events as RTP type 101 not 106.  otherwise we can not send DTMF correctly.

Modified:
    team/dvossel/gtalk_fixup/channels/chan_gtalk.c

Modified: team/dvossel/gtalk_fixup/channels/chan_gtalk.c
URL: http://svnview.digium.com/svn/asterisk/team/dvossel/gtalk_fixup/channels/chan_gtalk.c?view=diff&rev=290610&r1=290609&r2=290610
==============================================================================
--- team/dvossel/gtalk_fixup/channels/chan_gtalk.c (original)
+++ team/dvossel/gtalk_fixup/channels/chan_gtalk.c Wed Oct  6 11:10:58 2010
@@ -173,7 +173,7 @@
 
 /* Forward declarations */
 static struct ast_channel *gtalk_request(const char *type, format_t format, const struct ast_channel *requestor, void *data, int *cause);
-static int gtalk_digit(struct ast_channel *ast, char digit, unsigned int duration);
+/*static int gtalk_digit(struct ast_channel *ast, char digit, unsigned int duration);*/
 static int gtalk_sendtext(struct ast_channel *ast, const char *text);
 static int gtalk_digit_begin(struct ast_channel *ast, char digit);
 static int gtalk_digit_end(struct ast_channel *ast, char digit, unsigned int duration);
@@ -412,7 +412,7 @@
 
 	if (codecs_num) {
 		/* only propose DTMF within an audio session */
-		iks_insert_attrib(payload_telephone, "id", "106");
+		iks_insert_attrib(payload_telephone, "id", "101");
 		iks_insert_attrib(payload_telephone, "name", "telephone-event");
 		iks_insert_attrib(payload_telephone, "clockrate", "8000");
 	}
@@ -997,6 +997,8 @@
 	  return NULL;
 	}
 	ast_rtp_instance_set_prop(tmp->rtp, AST_RTP_PROPERTY_RTCP, 1);
+	ast_rtp_instance_set_prop(tmp->rtp, AST_RTP_PROPERTY_DTMF, 1);
+	ast_rtp_instance_dtmf_mode_set(tmp->rtp, AST_RTP_DTMF_MODE_RFC2833);
 	ast_rtp_codecs_payloads_clear(ast_rtp_instance_get_codecs(tmp->rtp), tmp->rtp);
 
 	/* add user configured codec capabilites */
@@ -1590,14 +1592,25 @@
 
 static int gtalk_digit_begin(struct ast_channel *chan, char digit)
 {
-	return gtalk_digit(chan, digit, 0);
+	struct gtalk_pvt *p = chan->tech_pvt;
+	if (!p) {
+		return -1;
+	}
+
+	return ast_rtp_instance_dtmf_begin(p->rtp, digit);
 }
 
 static int gtalk_digit_end(struct ast_channel *chan, char digit, unsigned int duration)
 {
-	return gtalk_digit(chan, digit, duration);
-}
-
+	struct gtalk_pvt *p = chan->tech_pvt;
+	if (!p) {
+		return -1;
+	}
+	return ast_rtp_instance_dtmf_end(p->rtp, digit);
+}
+
+/* This is not being used at the moment, but I want this code to remain in the code base
+ * as a reference.
 static int gtalk_digit(struct ast_channel *ast, char digit, unsigned int duration)
 {
 	struct gtalk_pvt *p = ast->tech_pvt;
@@ -1623,8 +1636,8 @@
 	ast_aji_increment_mid(client->connection->mid);
 	iks_insert_attrib(gtalk, "xmlns", "http://jabber.org/protocol/gtalk");
 	iks_insert_attrib(gtalk, "action", "session-info");
-	/* put the initiator attribute to lower case if we receive the call
-	 * otherwise GoogleTalk won't establish the session */
+	// put the initiator attribute to lower case if we receive the call
+	// otherwise GoogleTalk won't establish the session
 	if (!p->initiator) {
 	        char c;
 	        char *t = lowerthem = ast_strdupa(p->them);
@@ -1651,7 +1664,7 @@
 	ast_mutex_unlock(&p->lock);
 	return 0;
 }
-
+*/
 static int gtalk_sendhtml(struct ast_channel *ast, int subclass, const char *data, int datalen)
 {
 	ast_log(LOG_NOTICE, "XXX Implement gtalk sendhtml XXX\n");