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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@409705 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Igor Goncharovskiy
2014-03-05 05:10:50 +00:00
parent 0e3e1b678c
commit 28b726e79b
1 changed files with 43 additions and 6 deletions
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49
channels/chan_unistim.c
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@@ -4196,6 +4196,7 @@ static int unistim_indicate(struct ast_channel *ast, int ind, const void *data,
s->device->missed_call = 0; s->device->missed_call = 0;
break; break;
case AST_CONTROL_PROCEEDING: case AST_CONTROL_PROCEEDING:
case AST_CONTROL_UPDATE_RTP_PEER:
break; break;
default: default:
ast_log(LOG_WARNING, "Don't know how to indicate condition %d\n", ind); ast_log(LOG_WARNING, "Don't know how to indicate condition %d\n", ind);
@@ -5114,14 +5115,14 @@ static struct unistim_device *build_device(const char *cat, const struct ast_var
return NULL; return NULL;
} }
ast_copy_string(d->name, cat, sizeof(d->name)); 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;
} }
ast_copy_string(context, DEFAULTCONTEXT, sizeof(context)); ast_copy_string(context, DEFAULTCONTEXT, sizeof(context));
d->contrast = -1;
d->output = OUTPUT_HANDSET;
d->previous_output = OUTPUT_HANDSET;
d->volume = VOLUME_LOW;
d->mute = MUTE_OFF;
d->height = DEFAULTHEIGHT;
linelabel[0] = '\0'; linelabel[0] = '\0';
dateformat = 1; dateformat = 1;
timeformat = 1; timeformat = 1;
@@ -5614,15 +5615,51 @@ static enum ast_rtp_glue_result unistim_get_rtp_peer(struct ast_channel *chan, s
{ {
struct unistim_subchannel *sub = chan->tech_pvt; struct unistim_subchannel *sub = chan->tech_pvt;
if (!sub) {
return AST_RTP_GLUE_RESULT_FORBID;
}
if (!sub->rtp) {
return AST_RTP_GLUE_RESULT_FORBID;
}
ao2_ref(sub->rtp, +1); ao2_ref(sub->rtp, +1);
*instance = sub->rtp; *instance = sub->rtp;
return AST_RTP_GLUE_RESULT_LOCAL; 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, format_t codecs, int nat_active)
{
struct unistim_subchannel *sub;
struct sockaddr_in them = { 0, };
struct sockaddr_in us = { 0, };
if (!rtp) {
return 0;
}
sub = chan->tech_pvt;
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 = { static struct ast_rtp_glue unistim_rtp_glue = {
.type = channel_type, .type = channel_type,
.get_rtp_info = unistim_get_rtp_peer, .get_rtp_info = unistim_get_rtp_peer,
.update_peer = unistim_set_rtp_peer,
}; };
/*--- load_module: PBX load module - initialization ---*/ /*--- load_module: PBX load module - initialization ---*/