TA 916e voice flashhook mode transparent not working

[rivera]

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]

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.

[rivera]

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

[jayh]

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. 

[rivera · Accepted solution]

Jayh:

I opened a ticket with Adtran support and they gave me the solution, here is by point:

1. Configure the voice flashhook threshold to the right values: 200 1000 to allow the system to capture the flash hook.
2. 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.

Jean Louis

[jayh]

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

[quirty]

Is the 908 configured in gateway mode? Are you forwarding any FXO's or PRI's to the SIP server?