[asterisk-commits] dbrooks: branch dbrooks/jinglegtalkwork r179459 - /team/dbrooks/jinglegtalkwo...

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Mon Mar 2 16:13:04 CST 2009


Author: dbrooks
Date: Mon Mar  2 16:12:24 2009
New Revision: 179459

URL: http://svn.digium.com/svn-view/asterisk?view=rev&rev=179459
Log:
Just getting rid of extraneous whitespace


Modified:
    team/dbrooks/jinglegtalkwork/channels/chan_gtalk.c

Modified: team/dbrooks/jinglegtalkwork/channels/chan_gtalk.c
URL: http://svn.digium.com/svn-view/asterisk/team/dbrooks/jinglegtalkwork/channels/chan_gtalk.c?view=diff&rev=179459&r1=179458&r2=179459
==============================================================================
--- team/dbrooks/jinglegtalkwork/channels/chan_gtalk.c (original)
+++ team/dbrooks/jinglegtalkwork/channels/chan_gtalk.c Mon Mar  2 16:12:24 2009
@@ -238,12 +238,12 @@
 	get_codec: gtalk_get_codec,
 };
 
-static char show_channels_usage[] = 
-"Usage: gtalk show channels\n" 
+static char show_channels_usage[] =
+"Usage: gtalk show channels\n"
 "       Shows current state of the Gtalk channels.\n";
 
-static char reload_usage[] = 
-"Usage: gtalk reload\n" 
+static char reload_usage[] =
+"Usage: gtalk reload\n"
 "       Reload gtalk channel driver.\n";
 
 
@@ -304,7 +304,7 @@
 		iks *payload_eg711u, *payload_pcmu;
 		payload_pcmu = iks_new("payload-type");
 		payload_eg711u = iks_new("payload-type");
-	
+
 		if(!payload_eg711u || !payload_pcmu) {
 			if(payload_pcmu)
 				iks_delete(payload_pcmu);
@@ -447,7 +447,7 @@
 		codecs_num = add_codec_to_answer(p, pref_codec, dcodecs);
 		alreadysent |= pref_codec;
 	}
-	
+
 	if (codecs_num) {
 		/* only propose DTMF within an audio session */
 		iks_insert_attrib(payload_telephone, "id", "106");
@@ -455,7 +455,7 @@
 		iks_insert_attrib(payload_telephone, "clockrate", "8000");
 	}
 	iks_insert_attrib(transport,"xmlns","http://www.google.com/transport/p2p");
-	
+
 	iks_insert_attrib(iq, "type", "set");
 	iks_insert_attrib(iq, "to", to);
 	iks_insert_attrib(iq, "from", from);
@@ -548,7 +548,7 @@
 {
 	struct gtalk_pvt *p = ast->tech_pvt;
 	int res = 0;
-	
+
 	if (option_debug)
 		ast_log(LOG_DEBUG, "Answer!\n");
 	ast_mutex_lock(&p->lock);
@@ -655,10 +655,10 @@
 		ast_rtp_set_rtpmap_type(tmp->rtp, atoi(iks_find_attrib(codec, "id")), "audio", iks_find_attrib(codec, "name"), 0);
 		codec = iks_next_tag(codec);
 	}
-	
+
 	/* Now gather all of the codecs that we are asked for */
 	ast_rtp_get_current_formats(tmp->rtp, &tmp->peercapability, &peernoncodeccapability);
-	
+
 	/* at this point, we received an awser from the remote Gtalk client,
 	   which allows us to compare capabilities */
 	tmp->jointcapability = tmp->capability & tmp->peercapability;
@@ -672,7 +672,7 @@
 		return -1;
 
 	}	
-	
+
 	from = iks_find_attrib(pak->x, "to");
 	if(!from)
 		from = client->connection->jid->full;
@@ -1184,7 +1184,7 @@
 	from = iks_find_attrib(pak->x,"to");
 	if(!from)
 		from = client->connection->jid->full;
-	
+
 	while (tmp) {
 		if (iks_find_with_attrib(pak->x, "session", "id", tmp->sid)) {
 			ast_log(LOG_NOTICE, "Ignoring duplicate call setup on SID %s\n", tmp->sid);
@@ -1194,15 +1194,15 @@
 		tmp = tmp->next;
 	}
 
- 	if (!strcasecmp(client->name, "guest")){
- 		/* the guest account is not tied to any configured XMPP client,
- 		   let's set it now */
- 		client->connection = ast_aji_get_client(from);
- 		if (!client->connection) {
- 			ast_log(LOG_ERROR, "No XMPP client to talk to, us (partial JID) : %s\n", from);
- 			return -1;
- 		}
- 	}
+	if (!strcasecmp(client->name, "guest")){
+		/* the guest account is not tied to any configured XMPP client,
+		   let's set it now */
+		client->connection = ast_aji_get_client(from);
+		if (!client->connection) {
+			ast_log(LOG_ERROR, "No XMPP client to talk to, us (partial JID) : %s\n", from);
+			return -1;
+		}
+	}
 
 	p = gtalk_alloc(client, from, pak->from->full, iks_find_attrib(pak->query, "id"));
 	if (!p) {
@@ -1223,20 +1223,20 @@
 				sizeof(p->sid));
 	}
 
-	/* codec points to the first <payload-type/> tag */	
+	/* codec points to the first <payload-type/> tag */
 	codec = iks_first_tag(iks_first_tag(iks_first_tag(pak->x)));
-	
+
 	while (codec) {
 		ast_rtp_set_m_type(p->rtp, atoi(iks_find_attrib(codec, "id")));
 		ast_rtp_set_rtpmap_type(p->rtp, atoi(iks_find_attrib(codec, "id")), "audio", iks_find_attrib(codec, "name"), 0);
 		codec = iks_next_tag(codec);
 	}
-	
+
 	/* Now gather all of the codecs that we are asked for */
 	ast_rtp_get_current_formats(p->rtp, &p->peercapability, &peernoncodeccapability);
 	p->jointcapability = p->capability & p->peercapability;
 	ast_mutex_unlock(&p->lock);
-		
+
 	ast_setstate(chan, AST_STATE_RING);
 	if (!p->jointcapability) {
 		ast_log(LOG_WARNING, "Capabilities don't match : us - %s, peer - %s, combined - %s \n", ast_getformatname_multiple(s1, BUFSIZ, p->capability),
@@ -1248,10 +1248,10 @@
 		gtalk_hangup(chan);
 		ast_channel_free(chan);
 		return -1;
-	}	
+	}
 
 	res = ast_pbx_start(chan);
-	
+
 	switch (res) {
 	case AST_PBX_FAILED:
 		ast_log(LOG_WARNING, "Failed to start PBX :(\n");
@@ -1295,19 +1295,19 @@
 		sin.sin_port = htons(tmp->port);
 		snprintf(username, sizeof(username), "%s%s", tmp->username,
 			 p->ourcandidates->username);
-		
+
 		/* Find out the result of the STUN */
 		ast_rtp_get_peer(p->rtp, &aux);
 
 		/* If the STUN result is different from the IP of the hostname,
 			lock on the stun IP of the hostname advertised by the
 			remote client */
-		if (aux.sin_addr.s_addr && 
+		if (aux.sin_addr.s_addr &&
 		    aux.sin_addr.s_addr != sin.sin_addr.s_addr)
 			ast_rtp_stun_request(p->rtp, &aux, username);
-		else 
+		else
 			ast_rtp_stun_request(p->rtp, &sin, username);
-		
+
 		if (aux.sin_addr.s_addr && option_debug > 3) {
 			ast_log(LOG_DEBUG, "Receiving RTP traffic from IP %s, matches with remote candidate's IP %s\n", ast_inet_ntoa(aux.sin_addr), tmp->ip);
 			ast_log(LOG_DEBUG, "Sending STUN request to %s\n", tmp->ip);
@@ -1368,7 +1368,7 @@
 				newcandidate->protocol = AJI_PROTOCOL_UDP;
 			if (!strcasecmp(iks_find_attrib(traversenodes, "protocol"), "ssltcp"))
 				newcandidate->protocol = AJI_PROTOCOL_SSLTCP;
-		
+
 			if (!strcasecmp(iks_find_attrib(traversenodes, "type"), "stun"))
 				newcandidate->type = AJI_CONNECT_STUN;
 			if (!strcasecmp(iks_find_attrib(traversenodes, "type"), "local"))
@@ -1379,7 +1379,7 @@
 							sizeof(newcandidate->network));
 			newcandidate->generation = atoi(iks_find_attrib(traversenodes, "generation"));
 			newcandidate->next = NULL;
-		
+
 			newcandidate->next = p->theircandidates;
 			p->theircandidates = newcandidate;
 			p->laststun = 0;
@@ -1388,7 +1388,7 @@
 		}
 		traversenodes = iks_next_tag(traversenodes);
 	}
-	
+
 	receipt = iks_new("iq");
 	iks_insert_attrib(receipt, "type", "result");
 	iks_insert_attrib(receipt, "from", from);
@@ -1610,7 +1610,7 @@
 */
 
 /*! \brief Initiate new call, part of PBX interface 
- * 	dest is the dial string */
+ *dest is the dial string */
 static int gtalk_call(struct ast_channel *ast, char *dest, int timeout)
 {
 	struct gtalk_pvt *p = ast->tech_pvt;
@@ -1731,14 +1731,14 @@
 				resource ++;
 			}
 			if (chan)
-				ast_cli(fd, FORMAT, 
+				ast_cli(fd, FORMAT,
 					chan->name,
 					jid,
 					resource,
 					ast_getformatname(chan->readformat),
-					ast_getformatname(chan->writeformat)					
+					ast_getformatname(chan->writeformat)
 					);
-			else 
+			else
 				ast_log(LOG_WARNING, "No available channel\n");
 			numchans ++;
 			p = p->next;
@@ -1874,8 +1874,8 @@
 		else if (!strcasecmp(var->name, "connection")) {
 			if ((client = ast_aji_get_client(var->value))) {
 				member->connection = client;
-				iks_filter_add_rule(client->f, gtalk_parser, member, 
-						    IKS_RULE_TYPE, IKS_PAK_IQ, 
+				iks_filter_add_rule(client->f, gtalk_parser, member,
+						    IKS_RULE_TYPE, IKS_PAK_IQ,
 						    IKS_RULE_FROM_PARTIAL, member->user,
 						    IKS_RULE_NS, "http://www.google.com/session",
 						    IKS_RULE_DONE);
@@ -2041,11 +2041,11 @@
 	}
 
 	sched = sched_context_create();
-	if (!sched) 
+	if (!sched)
 		ast_log(LOG_WARNING, "Unable to create schedule context\n");
 
 	io = io_context_create();
-	if (!io) 
+	if (!io)
 		ast_log(LOG_WARNING, "Unable to create I/O context\n");
 
 	if (ast_find_ourip(&__ourip, bindaddr)) {




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