[Asterisk-Users] Re: Sipura SPA-841 auto-answer support [patch]
Geoff Speicher
geoff at speicher.org
Sun Jan 30 10:37:26 MST 2005
> The best place for patches to Asterisk is the Asterisk bug tracker,
> which can be found at http://bugs.digium.com .
>
> *Please* don't forget to read the guidelines:
> http://www.digium.com/bugguidelines.html and file a disclaimer.
>
> Thanks for the patch! I'll look for it on the bug tracker :)
Actually, I didn't really expect that the patch would be committed to
Asterisk CVS, for a couple of reasons:
1. Asterisk devel already has sipAddHeader (or something like that),
which is a more generally useful way to accomplish the same thing.
Until that function makes its way to a stable release, I just need a
quick, stable way to support Call-Info, and I assumed that I wasn't
the only one.
2. It ain't the prettiest hack. Granted it's simple, but it's making
an existing problem worse. Take a look at transmit_reinvite(). I'm
reminded of a favorite fortune(6) of mine:
If you have a procedure with 10 parameters, you probably missed some.
char* call_info is parameter number nine. :)
A better patch would create a more flexible mechanism for Dial to pass
information from dialplan variables into transmit_reinvite().
That being said, if enough people find this patch useful, then I'd
certainly file whatever disclaimers necessary to get it committed.
Geoff
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