diff --git a/src/channels/chan_voicemngr.c b/src/channels/chan_voicemngr.c
index 9f968184e740e3fbb199738b15a2cb92aa509886..766289e4883b38cacd8d794d4f315f7ccfa27ca9 100644
--- a/src/channels/chan_voicemngr.c
+++ b/src/channels/chan_voicemngr.c
@@ -2602,12 +2602,11 @@ void handle_dtmf_calling(struct chan_voicemngr_subchannel *sub)
char dtmf_last_char = p->dtmfbuf[(dtmfbuf_len - 1)];
char dtmf_one_before_last_char = p->dtmfbuf[(dtmfbuf_len > 1 ? dtmfbuf_len - 2 : 0)];
- sub->direct_extension = 0;
- if (ast_exists_extension(NULL, p->context_direct, p->dtmfbuf, 1, p->cid_num) && !ast_matchmore_extension(NULL, p->context_direct, p->dtmfbuf, 1, p->cid_num))
+ if (ast_exists_extension(NULL, p->context_direct, p->dtmfbuf, 1, p->cid_num) &&
+ !ast_matchmore_extension(NULL, p->context_direct, p->dtmfbuf, 1, p->cid_num))
{
//We have a full match in the "direct" context, so have asterisk place a call immediately
ast_debug(9, "Direct extension matching %s found\n", p->dtmfbuf);
- sub->direct_extension = 1;
chan_voicemngr_start_calling(sub, p->context_direct);
}
@@ -3141,28 +3140,14 @@ static void handle_dtmf(enum LINE_EVENT event,
if (p->dtmf_first < 0) { // DTMF pressed
p->dtmf_first = dtmf_button;
ast_debug(9,"Pressed DTMF %s\n", dtmfMap->name);
- if (owner) {
- struct ast_channel *bridged_chan = ast_channel_bridge_peer(owner);
- if (bridged_chan) {
- const struct ast_channel_tech *chan_tech = ast_channel_tech(bridged_chan);
- if (chan_tech && chan_tech->type && *chan_tech->type) {
- ast_debug(1, "debug_tmp: the type of the bridged channel is %s, sub->direct_extension = %d\n",
- chan_tech->type, sub->direct_extension);
- } else {
- ast_debug(1, "debug_tmp: the type of the bridged channel is unknown, sub->direct_extension = %d\n",
- sub->direct_extension);
- }
- ast_channel_unref(bridged_chan);
- } else {
- ast_debug(1, "debug_tmp: there is no bridged channel, sub->direct_extension = %d. is there a bridge? %s\n",
- sub->direct_extension, ast_channel_internal_bridge(owner) ? "yes" : "no");
- }
- }
- if (owner && chan_voicemngr_should_relay_dtmf(sub) && (sub->dtmf_mode == AST_SIP_DTMF_INFO || sub->conference_initiator == 1 || sub->direct_extension == 1)) {
- // INCALL
+ /* Frame AST_FRAME_DTMF_BEGIN must be queued to the channel if the call is terminated by Asterisk core or applications,
+ * e.g. VoiceMail, call forwarding. Otherwise DTMF digits won't be collected if DTMF relay method is not SIP INFO.
+ * There is no bridge created for those calls. A bridge is always created for normal 2-way or 3-way calls no matter they are
+ * internal or external. */
+ if (owner && chan_voicemngr_should_relay_dtmf(sub) &&
+ (sub->dtmf_mode == AST_SIP_DTMF_INFO || sub->conference_initiator == 1 || !ast_channel_internal_bridge(owner))) {
send_outgoing_dtmf(owner, dtmf_button, AST_FRAME_DTMF_BEGIN);
}
-
/* Do not send AST_FRAME_DTMF_BEGIN to allow DSP-generated tone to pass through */
} else if (p->dtmf_first == dtmf_button) { // DTMF released
ast_log(LOG_NOTICE,"Released DTMF %s\n", dtmfMap->name);
@@ -3196,25 +3181,11 @@ static void handle_dtmf(enum LINE_EVENT event,
sub->channel_state = DIALING;
//ast_debug(5,"Set to state %s.\n", state2str(sub->channel_state));
} else if (sub->channel_state == INCALL) {
- if (owner) {
- struct ast_channel *bridged_chan = ast_channel_bridge_peer(owner);
- if (bridged_chan) {
- const struct ast_channel_tech *chan_tech = ast_channel_tech(bridged_chan);
- if (chan_tech && chan_tech->type && *chan_tech->type) {
- ast_debug(1, "debug_tmp: the type of the bridged channel is %s, sub->direct_extension = %d\n",
- chan_tech->type, sub->direct_extension);
- } else {
- ast_debug(1, "debug_tmp: the type of the bridged channel is unknown, sub->direct_extension = %d\n",
- sub->direct_extension);
- }
- ast_channel_unref(bridged_chan);
- } else {
- ast_debug(1, "debug_tmp: there is no bridged channel, sub->direct_extension = %d. is there a bridge? %s\n",
- sub->direct_extension, ast_channel_internal_bridge(owner) ? "yes" : "no");
- }
- }
- if (owner && (sub->dtmf_mode == AST_SIP_DTMF_INFO || sub->conference_initiator == 1 || sub->direct_extension)) {
- // INCALL
+ /* Frame AST_FRAME_DTMF_END must be queued to the channel if the call is terminated by Asterisk core or applications,
+ * e.g. VoiceMail, call forwarding. Otherwise DTMF digits won't be collected if DTMF relay method is not SIP INFO.
+ * There is no bridge created for those calls. A bridge is always created for normal 2-way or 3-way calls no matter they are
+ * internal or external. */
+ if (owner && (sub->dtmf_mode == AST_SIP_DTMF_INFO || sub->conference_initiator == 1 || !ast_channel_internal_bridge(owner))) {
send_outgoing_dtmf(owner, dtmf_button, AST_FRAME_DTMF_END);
}
} else {
@@ -3928,7 +3899,6 @@ static struct chan_voicemngr_pvt *chan_voicemngr_allocate_pvt(void)
sub->sip_client_id = -1;
sub->period = default_ptime; // 20 ms
sub->dtmf_mode = AST_SIP_DTMF_RFC_4733;
- sub->direct_extension = 0;
sub->dtmf_pt = RTP_PT_DTMF;
sub->conference_initiator = 0;
tmp->sub[i] = sub;
@@ -5811,7 +5781,6 @@ static int chan_voicemngr_close_connection(struct chan_voicemngr_subchannel *sub
sub->codec = -1;
sub->call_id = CALLID_INVALID;
sub->updated_codec = 0;
- sub->direct_extension = 0;
ast_debug(1, "Virtual Asterisk connection %d/%d destroyed\n", p->line_id, sub->connection_id);
}
diff --git a/src/channels/chan_voicemngr.h b/src/channels/chan_voicemngr.h
index 4ad5e061cb809b6fa9ef8d2b35324d228c2f141d..304761d1e76b0c526b03945c8c41adeb2e9add7a 100644
--- a/src/channels/chan_voicemngr.h
+++ b/src/channels/chan_voicemngr.h
@@ -190,7 +190,6 @@ struct chan_voicemngr_subchannel {
int updated_codec;
int sip_client_id; /* The SIP client used for the current call: -1 = not set, 0 = internal call, 1..MAX_SIP_CLIENTS = sip client id*/
int congestion_timer_id;
- int direct_extension;
};
struct chan_voicemngr_channel_tech {