[asterisk-users] SIP DTMF not acted on for features in 1.4.0b3

[Russell Brown]

Asterisk seems to be ignoring DTMF for features in Asterisk 1.4.0b3

My SNOM sends the dtmf-relay and Asterisk gets it because I can
see it in the sip debug.

However, once seen, Asterisk doesn't actually do anything about it. I've
checked features and that seems fine. Is this a bug or something that
I've screwed up?

For the record, here's the features setting:

	asterisk*CLI> show features
	Builtin Feature           Default Current
	---------------           ------- -------
	Pickup                    *8      *8     
	Blind Transfer            #       #      
	Attended Transfer                 *2     
	One Touch Monitor                 *1     
	Disconnect Call           *       *      
	Park Call                         #72    

	Dynamic Feature           Default Current
	---------------           ------- -------
	testfeature               no def  #9     

	Call parking
	------------
	Parking extension   :   700
	Parking context     :   parkedcalls
	Parked call extensions: 701-720

	asterisk*CLI> 

and here's a SIP trace of me pressing '*' during a call (which according
to my features should Disconnect the Call.

	asterisk*CLI> 
	<--- SIP read from 192.168.1.12:5060 --->
	INFO sip:114 at 192.168.1.13 SIP/2.0
	Via: SIP/2.0/UDP 192.168.1.12:5060;branch=z9hG4bK-llnm8m4u2wef;rport
	From: "Russell <112>" <sip:112 at asterisk.lls.com>;tag=tmyszljbna
	To: <sip:114 at asterisk.lls.com>;tag=as0b7389e4
	Call-ID: 3c267d2aecd1-pr1an79znzog at snom360-000413231B20
	CSeq: 14 INFO
	Max-Forwards: 70
	Contact: <sip:112 at 192.168.1.12:5060;line=gv8x1x75>;flow-id=1
	User-Agent: snom360/6.5.1
	Content-Type: application/dtmf-relay
	Content-Length: 22

	Signal=*
	Duration=160
	<------------->
	--- (11 headers 2 lines) ---
	Receiving INFO!
	* DTMF-relay event received: *
	asterisk*CLI> 
	<--- Transmitting (no NAT) to 192.168.1.12:5060 --->
	SIP/2.0 200 OK
	Via: SIP/2.0/UDP
	192.168.1.12:5060;branch=z9hG4bK-llnm8m4u2wef;received=192.168.1.12;rport=5060
	From: "Russell <112>" <sip:112 at asterisk.lls.com>;tag=tmyszljbna
	To: <sip:114 at asterisk.lls.com>;tag=as0b7389e4
	Call-ID: 3c267d2aecd1-pr1an79znzog at snom360-000413231B20
	CSeq: 14 INFO
	User-Agent: Asterisk PBX
	Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
	Supported: replaces
	Contact: <sip:114 at 192.168.1.13>
	Content-Length: 0


	<------------>
	<< [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/112-0070a2c0]
	asterisk*CLI>

Can anyone suggest what's wrong here?

Thanks.

-- 
 Regards,
     Russell
 --------------------------------------------------------------------
| Russell Brown          | MAIL: russell at lls.com PHONE: 01780 471800 |
| Lady Lodge Systems     | WWW Work: http://www.lls.com              |
| Peterborough, England  | WWW Play: http://www.ruffle.me.uk         |
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