I have a TA 916e running ROS A4.11.00.E for a SIP to FXS application with a metaswitch as sip server . I am trying to use the flash hook option to transfer a call to an external phone number but the IAD is not generating any sip message after the I operate the flash hook, the only message I see is the "14:27:01.891 FXS.0/1 Onhook Detected" in the FXS interface. Please find attached the output of the debug command.
Jayh:
I opened a ticket with Adtran support and they gave me the solution, here is by point:
Thanks for your assistance I appreciate your time.
Jean Louis
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.
Jayh:
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 cos
no first-name
no last-name
password "1234"
no description
no location
no aa-dnd
call-waiting
no group-ring-call-waiting
no block-caller-id
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 did
no dnd
no forward
fwd-courtesy
special-ring-cadences
no hotel
no hotline
message-waiting both
no station-lock admin
no station-lock admin inbound
no station-lock user
no station-lock user inbound
no num-rings
no coverage
forward-disconnect battery remove
forward-disconnect delay 750
sip-authentication password "1234"
no vad
no modem-passthrough
no t38
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
no plc
no alc
nls
no anlp
echo-cancellation
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
no codec-group
no sip-keep-alive
no dnis-digits
Jean Louis
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.
Jayh:
I opened a ticket with Adtran support and they gave me the solution, here is by point:
Thanks for your assistance I appreciate your time.
Jean Louis
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
Is the 908 configured in gateway mode? Are you forwarding any FXO's or PRI's to the SIP server?