diff --git a/src/channels/chan_voicemngr.c b/src/channels/chan_voicemngr.c
index 024ba01c462795e4a1f217b1cb654ed72a0c8a85..725462786d9f60b297bd725810dd247e17e14f4f 100644
--- a/src/channels/chan_voicemngr.c
+++ b/src/channels/chan_voicemngr.c
@@ -736,6 +736,10 @@ static void endpt_connection_data(int line, int id, const char *action, const st
if (data->mask & UBUS_DATA_PTIME_BIT){
blobmsg_add_u32(&bb, "ptime", data->ptime);
}
+ if (data->mask & UBUS_DATA_DTMF_BIT){
+ blobmsg_add_u32(&bb, "dtmf_mode", data->dtmf_mode);
+ ast_log(LOG_NOTICE, "connection parm_update, dtmf_mode: %d", data->dtmf_mode);
+ }
}
req = ast_calloc(1, sizeof(struct ubus_request));
@@ -815,6 +819,7 @@ static void chan_voicemngr_modify_codec(struct chan_voicemngr_subchannel *sub) {
ast_channel_lock(sub->owner);
ast_channel_codec_set(sub->owner, !ast_strlen_zero(ast_channel_codec_get(bridged_chan)) ?
ast_channel_codec_get(bridged_chan) : ast_format_get_name(ast_channel_writeformat(bridged_chan)));
+ ast_channel_dtmf_mode_set(sub->owner, ast_channel_dtmf_mode_get(bridged_chan));
ast_channel_unlock(sub->owner);
}
ast_channel_unref(bridged_chan);
@@ -851,9 +856,18 @@ static void chan_voicemngr_modify_codec(struct chan_voicemngr_subchannel *sub) {
}
data.ptime = sub->period;
data.mask = data.mask|UBUS_DATA_PTIME_BIT;
+ /* DTMF mode */
+ if (ast_channel_dtmf_mode_get(sub->owner) != AST_SIP_DTMF_NONE){
+ sub->dtmf_mode = ast_channel_dtmf_mode_get(sub->owner);
+ }
+ if (sub->dtmf_mode != AST_SIP_DTMF_RFC_4733){
+ data.dtmf_mode = sub->dtmf_mode;
+ data.mask = data.mask|UBUS_DATA_DTMF_BIT;
+ }
+
/* push parm_update to voicemngr */
endpt_connection_data(sub->parent->line_id, sub->call_id, "parm_update", &data);
- ast_log(LOG_NOTICE, "connection parm_update, codec: %d, ptime: %d",data.codec, data.ptime);
+ ast_log(LOG_NOTICE, "connection parm_update, codec: %d, ptime: %d\n",data.codec, data.ptime);
if (data.mask & UBUS_DATA_CODEC_BIT) {
sub->updated_codec = 1;
@@ -1718,6 +1732,7 @@ static int chan_voicemngr_classify_rtp_packet(int payload_type) {
case RTCP_SR: return CHAN_VOICEMNGR_RTCP_SR;
case RTCP_RR: return CHAN_VOICEMNGR_RTCP_RR;
case RTP_PT_CN: return CHAN_VOICEMNGR_AUDIO;
+ case RTP_PT_DTMF: return CHAN_VOICEMNGR_AUDIO;
}
for (int i=0; i<VOICEMNGR_CODECS_NUM; i++)
@@ -3104,7 +3119,7 @@ static void handle_dtmf(enum LINE_EVENT event,
if (p->dtmf_first < 0) {
p->dtmf_first = dtmf_button;
ast_debug(9,"Pressed DTMF %s\n", dtmfMap->name);
- if (owner && chan_voicemngr_should_relay_dtmf(sub)) {
+ if (owner && chan_voicemngr_should_relay_dtmf(sub) && (sub->dtmf_mode == AST_SIP_DTMF_INFO || sub->conference_initiator == 1)) {
// INCALL
send_outgoing_dtmf(owner, dtmf_button, AST_FRAME_DTMF_BEGIN);
}
@@ -3156,7 +3171,7 @@ static void handle_dtmf(enum LINE_EVENT event,
ast_queue_frame(owner, &f);
}
} else {
- if (owner) {
+ if (owner && (sub->dtmf_mode == AST_SIP_DTMF_INFO || sub->conference_initiator == 1)) {
// INCALL
send_outgoing_dtmf(owner, dtmf_button, AST_FRAME_DTMF_END);
}
@@ -3207,7 +3222,7 @@ static struct ast_format *map_rtpname_to_format(char* name) {
if (!strcmp(voicemngr_codecs[i].name,name))
return *voicemngr_codecs[i].ast_format;
- ast_verbose("Warning: unknown RTP payload_type %s is received. Treated it as PCMU\n", name);
+ ast_verbose("Warning: unknown RTP payload_type name %s is received. Treated it as PCMU\n", name);
return ast_format_ulaw;
}
@@ -3223,7 +3238,7 @@ static void audio_packet_handler(pe_packet_t *p) {
unsigned int* packet_buf32 = (unsigned int*)ap->rtp;
// Clear the RTP marker bit
- if ((payload_type & RTP_MARKER_BIT) && payload_type < RTCP_SR)
+ if ((payload_type & RTP_MARKER_BIT) && (payload_type < RTCP_SR || payload_type > RTCP_XR))
payload_type &= (~RTP_MARKER_BIT);
packet_type = chan_voicemngr_classify_rtp_packet(payload_type);
@@ -3254,9 +3269,15 @@ static void audio_packet_handler(pe_packet_t *p) {
frame.frametype = AST_FRAME_CNG;
frame.subclass.integer = ap->rtp[12];
} else if (sub->channel_state == OFFHOOK
- || sub->channel_state == DIALING
- || (owner && !ast_strlen_zero(ast_channel_codec_get(owner)) && strcmp(chan_voicemngr_get_codec_string(payload_type),ast_channel_codec_get(owner)))){
+ || sub->channel_state == DIALING){
drop_frame=1;
+ } else if(payload_type == RTP_PT_DTMF) {
+ frame.frametype = AST_FRAME_DTMF_BYPASS;
+ frame.subclass.integer = ap->rtp[1]; //payload_type
+ ast_debug(3, "DTMF rtp_event for digit: %d, with payload_type: %d \n", ap->rtp[12], ap->rtp[1]);
+ if (sub->conference_initiator == 1) {
+ drop_frame=1;
+ }
} else {
struct ast_format *format = map_rtptype_to_format(payload_type);
if (format) {
@@ -3855,6 +3876,7 @@ static struct chan_voicemngr_pvt *chan_voicemngr_allocate_pvt(void)
sub->congestion_timer_id = -1;
sub->sip_client_id = -1;
sub->period = default_ptime; // 20 ms
+ sub->dtmf_mode = AST_SIP_DTMF_RFC_4733;
sub->conference_initiator = 0;
tmp->sub[i] = sub;
sub->ingressRtcpPackets = 0;
diff --git a/src/channels/chan_voicemngr.h b/src/channels/chan_voicemngr.h
index 173747d802e10d3440a7953923577a126797d104..a024544b93251552c178f3923d726e4a2830ee80 100644
--- a/src/channels/chan_voicemngr.h
+++ b/src/channels/chan_voicemngr.h
@@ -164,6 +164,7 @@ struct chan_voicemngr_subchannel {
uint16_t sequence_number; /* Endpoint RTP sequence number state */
unsigned int time_stamp; /* Endpoint RTP time stamp state */
unsigned int period; /* Endpoint RTP period */
+ enum ast_sip_dtmf_mode dtmf_mode;
int codec; /* Used codec */
struct chan_voicemngr_pvt *parent; /* chan_voicemngr_line owning this subchannel */
int cw_timer_id; /* Current call waiting timer id, -1 if no active timer */