[svn-commits] igorg: trunk r409747 - /trunk/channels/chan_unistim.c
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Wed Mar 5 00:17:05 CST 2014
Author: igorg
Date: Wed Mar 5 00:17:03 2014
New Revision: 409747
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=409747
Log:
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
Modified:
trunk/channels/chan_unistim.c
Modified: trunk/channels/chan_unistim.c
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/chan_unistim.c?view=diff&rev=409747&r1=409746&r2=409747
==============================================================================
--- trunk/channels/chan_unistim.c (original)
+++ trunk/channels/chan_unistim.c Wed Mar 5 00:17:03 2014
@@ -3234,6 +3234,11 @@
ast_verb(0, "Handle Call Incoming for %s@%s\n", sub->parent->name,
s->device->name);
}
+ start_rtp(sub);
+ if (!sub->rtp) {
+ ast_log(LOG_WARNING, "Unable to create channel for %s@%s\n", sub->parent->name, s->device->name);
+ return;
+ }
if (sub->owner) {
ast_queue_control(sub->owner, AST_CONTROL_ANSWER);
}
@@ -4477,7 +4482,7 @@
}
if (memcmp(buf + SIZE_HEADER, packet_recv_pick_up, sizeof(packet_recv_pick_up)) == 0) {
if (unistimdebug) {
- ast_verb(0, "Handset off hook\n");
+ ast_verb(0, "Handset off hook, current state: %s\n", ptestate_tostr(pte->state));
}
if (!pte->device) { /* We are not yet registered (asking for a TN in AUTOPROVISIONING_TN) */
return;
@@ -4503,7 +4508,7 @@
}
if (memcmp(buf + SIZE_HEADER, packet_recv_hangup, sizeof(packet_recv_hangup)) == 0) {
if (unistimdebug) {
- ast_verb(0, "Handset on hook\n");
+ ast_verb(0, "Handset on hook, current state: %s\n", ptestate_tostr(pte->state));
}
if (!pte->device) {
return;
@@ -4878,7 +4883,7 @@
unistim_unalloc_sub(d, sub);
return 0;
}
- if (sub_real && (sub_real->owner) && (sub->subtype == SUB_THREEWAY)) { /* 3way call cancelled by softkey pressed */
+ if (sub_real && (sub_real->owner) && (sub->subtype == SUB_THREEWAY) && (s->state == STATE_CALL)) { /* 3way call cancelled by softkey pressed */
if (unistimdebug) {
ast_verb(0, "Real call disconnected, stay in call\n");
}
@@ -4966,9 +4971,6 @@
l = sub->parent;
d = l->parent;
- if ((!sub->rtp) && (!get_sub(d, SUB_THREEWAY))) {
- start_rtp(sub);
- }
if (unistimdebug) {
ast_verb(0, "unistim_answer(%s) on %s@%s-%d\n", ast_channel_name(ast), l->name,
l->parent->name, sub->softkey);
@@ -5223,8 +5225,7 @@
}
}
-static int unistim_indicate(struct ast_channel *ast, int ind, const void *data,
- size_t datalen)
+static int unistim_indicate(struct ast_channel *ast, int ind, const void *data, size_t datalen)
{
struct unistim_subchannel *sub;
struct unistim_line *l;
@@ -5282,6 +5283,7 @@
case AST_CONTROL_PROGRESS:
case AST_CONTROL_SRCUPDATE:
case AST_CONTROL_PROCEEDING:
+ case AST_CONTROL_UPDATE_RTP_PEER:
break;
case -1:
ast_playtones_stop(ast);
@@ -5296,7 +5298,9 @@
}
break;
case AST_CONTROL_SRCCHANGE:
- ast_rtp_instance_change_source(sub->rtp);
+ if (sub->rtp) {
+ ast_rtp_instance_change_source(sub->rtp);
+ }
break;
default:
ast_log(LOG_WARNING, "Don't know how to indicate condition %d\n", ind);
@@ -5908,14 +5912,7 @@
if (unistimdebug) {
ast_verb(0, "unistim_request owner = %p\n", sub->owner);
}
- start_rtp(sub);
- if (!sub->rtp) {
- ast_log(LOG_WARNING, "Unable to create channel for %s@%s\n", sub->parent->name, d->name);
- return NULL;
- }
-
restart_monitor();
-
/* and finish */
return tmpc;
}
@@ -6345,6 +6342,16 @@
}
ast_mutex_init(&d->lock);
ast_copy_string(d->name, cat, sizeof(d->name));
+
+ 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;
} else {
/* Delete existing line information */
AST_LIST_LOCK(&d->lines);
@@ -6364,16 +6371,6 @@
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';
dateformat = 1;
timeformat = 1;
@@ -6905,15 +6902,51 @@
{
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 = ast_channel_tech_pvt(chan);
+ struct sockaddr_in them = { 0, };
+ struct sockaddr_in us = { 0, };
+
+ if (!rtp) {
+ return 0;
+ }
+
+ sub = (struct unistim_subchannel *) 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 ---*/
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