[asterisk-dev] [svn-commits] jrose: branch 1.8 r369750 - /branches/1.8/channels/chan_sip.c
Jonathan Rose
jrose at digium.com
Mon Jul 9 09:31:34 CDT 2012
Olle E. Johansson wrote:
> So we can receive but not send. Does that make sense in a product, do
> you think?
> Either remove the receive part or add the send part is my suggestion.
>
> Now, removing functionality that people may depend on is something
> the community normally doesn't like.
Well, again, I didn't really have any intention of changing functionality
right now, and I'm certainly not going to be removing any functionality
from chan_sip that has been around for multiple versions. What I'm going
to do is just set the res to -1 like it would have before when it fell
through to the default case, but without displaying the warning saying
chan_sip doesn't know how to handle it, because it isn't something we
should really be warning for. It's a common thing that can happen when you
have analog phones connected with SIP phones.
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.
--
Jonathan R. Rose
Digium, Inc. | Software Engineer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct +1 256 428 6139
Check us out at: http://digium.com & http://asterisk.org
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