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/trunk@409747 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Igor Goncharovskiy
parent
33eaf4a4b8
commit
fe73de1302
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@ -3234,6 +3234,11 @@ static void handle_call_incoming(struct unistimsession *s)
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ast_verb(0, "Handle Call Incoming for %s@%s\n", sub->parent->name,
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s->device->name);
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}
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start_rtp(sub);
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if (!sub->rtp) {
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ast_log(LOG_WARNING, "Unable to create channel for %s@%s\n", sub->parent->name, s->device->name);
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return;
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}
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if (sub->owner) {
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ast_queue_control(sub->owner, AST_CONTROL_ANSWER);
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}
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@ -4477,7 +4482,7 @@ static void process_request(int size, unsigned char *buf, struct unistimsession
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}
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if (memcmp(buf + SIZE_HEADER, packet_recv_pick_up, sizeof(packet_recv_pick_up)) == 0) {
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if (unistimdebug) {
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ast_verb(0, "Handset off hook\n");
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ast_verb(0, "Handset off hook, current state: %s\n", ptestate_tostr(pte->state));
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}
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if (!pte->device) { /* We are not yet registered (asking for a TN in AUTOPROVISIONING_TN) */
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return;
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@ -4503,7 +4508,7 @@ static void process_request(int size, unsigned char *buf, struct unistimsession
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}
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if (memcmp(buf + SIZE_HEADER, packet_recv_hangup, sizeof(packet_recv_hangup)) == 0) {
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if (unistimdebug) {
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ast_verb(0, "Handset on hook\n");
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ast_verb(0, "Handset on hook, current state: %s\n", ptestate_tostr(pte->state));
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}
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if (!pte->device) {
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return;
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@ -4878,7 +4883,7 @@ static int unistim_hangup(struct ast_channel *ast)
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unistim_unalloc_sub(d, sub);
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return 0;
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}
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if (sub_real && (sub_real->owner) && (sub->subtype == SUB_THREEWAY)) { /* 3way call cancelled by softkey pressed */
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if (sub_real && (sub_real->owner) && (sub->subtype == SUB_THREEWAY) && (s->state == STATE_CALL)) { /* 3way call cancelled by softkey pressed */
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if (unistimdebug) {
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ast_verb(0, "Real call disconnected, stay in call\n");
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}
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@ -4966,9 +4971,6 @@ static int unistim_answer(struct ast_channel *ast)
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l = sub->parent;
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d = l->parent;
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if ((!sub->rtp) && (!get_sub(d, SUB_THREEWAY))) {
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start_rtp(sub);
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}
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if (unistimdebug) {
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ast_verb(0, "unistim_answer(%s) on %s@%s-%d\n", ast_channel_name(ast), l->name,
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l->parent->name, sub->softkey);
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@ -5223,8 +5225,7 @@ static void in_band_indication(struct ast_channel *ast, const struct ast_tone_zo
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}
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}
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static int unistim_indicate(struct ast_channel *ast, int ind, const void *data,
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size_t datalen)
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static int unistim_indicate(struct ast_channel *ast, int ind, const void *data, size_t datalen)
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{
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struct unistim_subchannel *sub;
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struct unistim_line *l;
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@ -5282,6 +5283,7 @@ static int unistim_indicate(struct ast_channel *ast, int ind, const void *data,
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case AST_CONTROL_PROGRESS:
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case AST_CONTROL_SRCUPDATE:
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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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case -1:
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ast_playtones_stop(ast);
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@ -5296,7 +5298,9 @@ static int unistim_indicate(struct ast_channel *ast, int ind, const void *data,
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}
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break;
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case AST_CONTROL_SRCCHANGE:
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ast_rtp_instance_change_source(sub->rtp);
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if (sub->rtp) {
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ast_rtp_instance_change_source(sub->rtp);
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}
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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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@ -5908,14 +5912,7 @@ static struct ast_channel *unistim_request(const char *type, struct ast_format_c
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if (unistimdebug) {
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ast_verb(0, "unistim_request owner = %p\n", sub->owner);
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}
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start_rtp(sub);
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if (!sub->rtp) {
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ast_log(LOG_WARNING, "Unable to create channel for %s@%s\n", sub->parent->name, d->name);
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return NULL;
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}
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restart_monitor();
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/* and finish */
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return tmpc;
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}
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@ -6345,6 +6342,16 @@ static struct unistim_device *build_device(const char *cat, const struct ast_var
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}
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ast_mutex_init(&d->lock);
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ast_copy_string(d->name, cat, sizeof(d->name));
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ast_copy_string(d->context, DEFAULTCONTEXT, sizeof(d->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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d->selected = -1;
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d->interdigit_timer = DEFAULT_INTERDIGIT_TIMER;
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} else {
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/* Delete existing line information */
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AST_LIST_LOCK(&d->lines);
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@ -6364,16 +6371,6 @@ static struct unistim_device *build_device(const char *cat, const struct ast_var
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memset(d->sline, 0, sizeof(d->sline));
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memset(d->sp, 0, sizeof(d->sp));
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}
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ast_copy_string(d->context, DEFAULTCONTEXT, sizeof(d->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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d->selected = -1;
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d->interdigit_timer = DEFAULT_INTERDIGIT_TIMER;
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linelabel[0] = '\0';
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dateformat = 1;
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timeformat = 1;
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@ -6905,15 +6902,51 @@ 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 = ast_channel_tech_pvt(chan);
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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, const struct ast_format_cap *codecs, int nat_active)
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{
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struct unistim_subchannel *sub = ast_channel_tech_pvt(chan);
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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 = (struct unistim_subchannel *) ast_channel_tech_pvt(chan);
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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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