[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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