From: Igor Goncharovskiy <igor.goncharovsky@gmail.com>
Date: Wed, 5 Mar 2014 06:11:21 +0000 (+0000)
Subject: Add update_peer function to unistim_rtp_glue, improve other unistim_rtp_glue function... 
X-Git-Tag: 12.2.0-rc1~99
X-Git-Url: http://git.ipfire.org/cgi-bin/gitweb.cgi?a=commitdiff_plain;h=9d963d4a54974d8ee84cd3f4f4adb1b969587f5b;p=thirdparty%2Fasterisk.git

Add update_peer function to unistim_rtp_glue, improve other unistim_rtp_glue functions conforming to other channel drivers. Do not forget auto-detected and user-selected phone settings on 'unistim reload'
........

Merged revisions 409705 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 409745 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@409746 65c4cc65-6c06-0410-ace0-fbb531ad65f3
---

diff --git a/channels/chan_unistim.c b/channels/chan_unistim.c
index 2f4b9c6a79..fa4f2c5c13 100644
--- a/channels/chan_unistim.c
+++ b/channels/chan_unistim.c
@@ -5221,6 +5221,7 @@ static int unistim_indicate(struct ast_channel *ast, int ind, const void *data,
 		if (sub->subtype == SUB_REAL) {
 			send_callerid_screen(s, sub);
 		}
+	case AST_CONTROL_UPDATE_RTP_PEER:
 		break;
 	case AST_CONTROL_SRCCHANGE:
 		ast_rtp_instance_change_source(sub->rtp);
@@ -6272,6 +6273,13 @@ static struct unistim_device *build_device(const char *cat, const struct ast_var
 		}
 		ast_mutex_init(&d->lock);
 		ast_copy_string(d->name, cat, sizeof(d->name));
+		d->contrast = -1;
+		d->output = OUTPUT_HANDSET;
+		d->previous_output = OUTPUT_HANDSET;
+		d->volume = VOLUME_LOW;
+		d->mute = MUTE_OFF;
+		d->height = DEFAULTHEIGHT;
+		d->selected = -1;
 	} else {
 		/* Delete existing line information */
 		AST_LIST_LOCK(&d->lines);
@@ -6291,14 +6299,7 @@ static struct unistim_device *build_device(const char *cat, const struct ast_var
 		memset(d->sline, 0, sizeof(d->sline));
 		memset(d->sp, 0, sizeof(d->sp));
 	}
-
 	ast_copy_string(d->context, DEFAULTCONTEXT, sizeof(d->context));
-	d->contrast = -1;
-	d->output = OUTPUT_HANDSET;
-	d->previous_output = OUTPUT_HANDSET;
-	d->volume = VOLUME_LOW;
-	d->mute = MUTE_OFF;
-	d->height = DEFAULTHEIGHT;
 	d->selected = -1;
 	d->interdigit_timer = DEFAULT_INTERDIGIT_TIMER;
 	linelabel[0] = '\0';
@@ -6830,15 +6831,51 @@ static enum ast_rtp_glue_result unistim_get_rtp_peer(struct ast_channel *chan, s
 {
 	struct unistim_subchannel *sub = ast_channel_tech_pvt(chan);
 
+	if (!sub) {
+		return AST_RTP_GLUE_RESULT_FORBID;
+	}
+	if (!sub->rtp) {
+		return AST_RTP_GLUE_RESULT_FORBID;
+	}
+
 	ao2_ref(sub->rtp, +1);
 	*instance = sub->rtp;
 
 	return AST_RTP_GLUE_RESULT_LOCAL;
 }
 
+static int unistim_set_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance *rtp, struct ast_rtp_instance *vrtp, struct ast_rtp_instance *trtp, const struct ast_format_cap *codecs, int nat_active)
+{
+	struct unistim_subchannel *sub;
+	struct sockaddr_in them = { 0, };
+	struct sockaddr_in us = { 0, };
+
+	if (!rtp) {
+		return 0;
+	}
+	
+	sub = ast_channel_tech_pvt(chan);
+	if (!sub) {
+		ast_log(LOG_ERROR, "No Private Structure, this is bad\n");
+		return -1;
+	}
+	{
+		struct ast_sockaddr tmp;
+		ast_rtp_instance_get_remote_address(rtp, &tmp);
+		ast_sockaddr_to_sin(&tmp, &them);
+		ast_rtp_instance_get_local_address(rtp, &tmp);
+		ast_sockaddr_to_sin(&tmp, &us);
+	}
+	
+	/* TODO: Set rtp on phone in case of direct rtp (not implemented) */
+	
+	return 0;
+}
+
 static struct ast_rtp_glue unistim_rtp_glue = {
 	.type = channel_type,
 	.get_rtp_info = unistim_get_rtp_peer,
+	.update_peer = unistim_set_rtp_peer,
 };
 
 /*--- load_module: PBX load module - initialization ---*/