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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'
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@409705 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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@@ -4196,6 +4196,7 @@ static int unistim_indicate(struct ast_channel *ast, int ind, const void *data,
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s->device->missed_call = 0;
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break;
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case AST_CONTROL_PROCEEDING:
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case AST_CONTROL_UPDATE_RTP_PEER:
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break;
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default:
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ast_log(LOG_WARNING, "Don't know how to indicate condition %d\n", ind);
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@@ -5114,14 +5115,14 @@ static struct unistim_device *build_device(const char *cat, const struct ast_var
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return NULL;
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}
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ast_copy_string(d->name, cat, sizeof(d->name));
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}
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ast_copy_string(context, DEFAULTCONTEXT, sizeof(context));
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d->contrast = -1;
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d->output = OUTPUT_HANDSET;
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d->previous_output = OUTPUT_HANDSET;
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d->volume = VOLUME_LOW;
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d->mute = MUTE_OFF;
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d->height = DEFAULTHEIGHT;
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}
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ast_copy_string(context, DEFAULTCONTEXT, sizeof(context));
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linelabel[0] = '\0';
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dateformat = 1;
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timeformat = 1;
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@@ -5614,15 +5615,51 @@ static enum ast_rtp_glue_result unistim_get_rtp_peer(struct ast_channel *chan, s
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{
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struct unistim_subchannel *sub = chan->tech_pvt;
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if (!sub) {
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return AST_RTP_GLUE_RESULT_FORBID;
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}
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if (!sub->rtp) {
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return AST_RTP_GLUE_RESULT_FORBID;
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}
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ao2_ref(sub->rtp, +1);
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*instance = sub->rtp;
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return AST_RTP_GLUE_RESULT_LOCAL;
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}
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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)
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{
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struct unistim_subchannel *sub;
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struct sockaddr_in them = { 0, };
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struct sockaddr_in us = { 0, };
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if (!rtp) {
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return 0;
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}
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sub = chan->tech_pvt;
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if (!sub) {
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ast_log(LOG_ERROR, "No Private Structure, this is bad\n");
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return -1;
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}
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{
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struct ast_sockaddr tmp;
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ast_rtp_instance_get_remote_address(rtp, &tmp);
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ast_sockaddr_to_sin(&tmp, &them);
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ast_rtp_instance_get_local_address(rtp, &tmp);
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ast_sockaddr_to_sin(&tmp, &us);
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}
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/* TODO: Set rtp on phone in case of direct rtp (not implemented) */
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return 0;
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}
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static struct ast_rtp_glue unistim_rtp_glue = {
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.type = channel_type,
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.get_rtp_info = unistim_get_rtp_peer,
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.update_peer = unistim_set_rtp_peer,
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};
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/*--- load_module: PBX load module - initialization ---*/
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