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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'

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


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@409745 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Igor Goncharovskiy
2014-03-05 05:54:11 +00:00
parent 7f1450e32c
commit 62a4018771
1 changed files with 44 additions and 7 deletions
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51
channels/chan_unistim.c
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@@ -5240,6 +5240,7 @@ 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);
@@ -6291,6 +6292,13 @@ 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);
@@ -6310,14 +6318,7 @@ 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';
@@ -6849,15 +6850,51 @@ 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 ---*/