asterisk users - Sep 2008 - Sip Info events

Hi
I'm trying to get a simens IP pjone working so I can transfer calls 
using the recall key when I run sip debug I get  the below text on 
screen, but I don't get dialtone  returned, any advice would be greatly 
appriciated

* DTMF-relay event received: FLASH
mythtv*CLI>
<--- Transmitting (no NAT) to 10.1.1.99:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 
10.1.1.99:5060;branch=z9hG4bK6c2f829ddf26e0a35104f6a7345df827;received=10.1.1.99;rport=5060
From: <sip:gigaset at 10.1.1.250>;tag=782243386
To: <sip:6001 at 10.1.1.250>;tag=as016f3248
Call-ID: 2672455879 at 10_1_1_99
CSeq: 667817631 INFO
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:6001 at 10.1.1.250>
Content-Length: 0



Regards

Robb

Hi Robb,

Have a look in your features.conf file and see what keys you have
enabled for transfers.

Regards,

Greyman.

On Sat, 2008-09-13 at 01:13 +0100, robb wrote:
> I'm trying to get a simens IP pjone working so I can transfer calls 
> using the recall key when I run sip debug I get  the below text on 
> screen, but I don't get dialtone  returned, any advice would be greatly
> appriciated
I don't claim to know much about the SIP protocol, but it's my (possibly
flawed) understanding that Asterisk doesn't interpret an incoming FLASH
message as part of a SIP INFO packet as a request to put the call on
hold or to transfer the call.  And just to be perfectly clear, I
personally don't think that FLASH is the proper way to initiate these
sorts of events in the SIP protocol.  To do a call transfer in SIP, for
example, you should be using the REFER primative.

(And just for the sake of general trivia, I haven't yet been able to
figure out how to get Asterisk to generate an outbound FLASH message
over SIP INFO either.)

-- 
Jared Smith
Training Manager
Digium, Inc.