Flashhook events are sent in much the same fashion as DTMF relay. If you have the voice user configured for DTMF inband it isn't likely to work. NTE 101 is the default and this should work. SIP INFO may as well.
Thanks for your respond, as you said the DTMF relay is by default NTE 101 and that is how it was configured when I did the test, see information below. Still not working.
LAB_IAD_Summit_Broadband#sh ru verbose | beg voice user
voice user 2392191001
connect fxs 0/1
no caller-id-override emergency-number
no caller-id-override external-name
no caller-id-override external-number
no caller-id-override internal-name
no caller-id-override internal-number
no station-lock admin
no station-lock admin inbound
no station-lock user
no station-lock user inbound
forward-disconnect battery remove
forward-disconnect delay 750
sip-authentication password "1234"
t38 redundancy high-speed 0
t38 redundancy low-speed 0
t38 max-buffer 200
t38 max-datagram 72
t38 max-rate 14400
t38 ced length 3000
no t38 ced auto-generate
no t38 generate-cng
t38 v21-preamble-timeout 1700
t38 error-correction fec
t38 fallback-mode g711
rtp frame-packetization 20
rtp frame-packetization mode negotiated
rtp delay-mode adaptive
rtp packet-delay nominal 50
rtp packet-delay fax 50
rtp packet-delay maximum 100
rtp dtmf-relay nte 101
no rtp qos dscp
It may be that the flashhook is indeed being sent but the other side is ignoring it. You won't necessarily see it in the debug, unless the debug level also shows you DTMF sent after the call is answered.
Do a wireshark capture and try sending DTMF 123 - flash - 456 and look at the RTP and RTCP for events, see if you can find the DTMF and the flash.
I opened a ticket with Adtran support and they gave me the solution, here is by point:
- Configure the voice flashhook threshold to the right values: 200 1000 to allow the system to capture the flash hook.
- Configure the voice flashhook mode to be interpreted because the sip server needed the re-invite instead of the info message.
Thanks for your assistance I appreciate your time.
Is the 908 configured in gateway mode? Are you forwarding any FXO's or PRI's to the SIP server?
The SDP doesn't look like you are sending DTMF as NTE101. It looks to be inband.
Here is your invite:
14:26:53.717 SIP.STACK MSG v=0 14:26:53.717 SIP.STACK MSG o=Sansay-VSXi 188 1 IN IP4 172.16.17.201 14:26:53.717 SIP.STACK MSG s=Session Controller 14:26:53.717 SIP.STACK MSG c=IN IP4 172.16.16.40 14:26:53.718 SIP.STACK MSG t=0 0 14:26:53.718 SIP.STACK MSG m=audio 15666 RTP/AVP 0 14:26:53.718 SIP.STACK MSG a=rtpmap:0 PCMU/8000 14:26:53.718 SIP.STACK MSG a=ptime:20
Try the following:
In global config:
voice feature-mode network
voice forward-mode network
In the SIP trunk config facing the provider:
rtp dtmf-relay offer nte 101