[asterisk-dev] [svn-commits] jrose: branch 1.8 r369750 - /branches/1.8/channels/chan_sip.c

[Kevin P. Fleming]

On 07/09/2012 09:31 AM, Jonathan Rose wrote:

> I agree handing receipt of an event but not having a way to send it is
> iffy, but since I'm not trying to change behavior in any meaningful way
> that just falls beyond the scope of what I was working on. It will
> continue to be sent through audio once res is set to -1 I think.

There is no 'audio' equivalent of a flash-hook; 'flash' is not an audio 
event, it's a line event (and one of the reasons it was dropped in the 
move from RFC 2833 to RFC 4733, I think).

If we wanted chan_sip and the RTP stack to be able to send them, that 
would mean parsing and remembering the RFC2833 telephony-event codes 
that we received in SDP from our peer; if they only offered to accept 
0-15, we can't send them 'flash' (which is 16). Asterisk may be one of 
the few SIP endpoints that actually offers to accept incoming 'flash' 
events over RTP :-)

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
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