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From: Maxim Sobolev <sobomax@FreeBSD.org>
Date: Tue, 3 May 2005 13:58:26 +0000 (UTC)
To: ports-committers@FreeBSD.org, cvs-ports@FreeBSD.org,
	cvs-all@FreeBSD.org
X-FreeBSD-CVS-Branch: HEAD
Subject: cvs commit: ports/net/asterisk Makefile ports/net/asterisk/files
 patch-channels::chan_sip.c patch-channels::chan_zap.c
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sobomax     2005-05-03 13:58:26 UTC

  FreeBSD ports repository

  Modified files:
    net/asterisk         Makefile 
    net/asterisk/files   patch-channels::chan_sip.c 
                         patch-channels::chan_zap.c 
  Log:
  o chan_sip.c:
  
    - Improve codec negotiation logic.
  
    - make sure to parse SDP on re-INVITE.
  
  o chan_zap.c:
  
    - If unanswered Zap channnel is hanged up wait until the calling party
      in turn hangs up (by detecting ring timeout). Otherwise next ring will
      trigger a "new" incoming call. This appears to be design limitation of
      the analogue telephone system - there is no way to reject the call without
      answering it first.
  
  Sponsored by:   Porta Software Ltd, PBXpress Inc
  
  Revision  Changes    Path
  1.31      +1 -1      ports/net/asterisk/Makefile
  1.3       +32 -8     ports/net/asterisk/files/patch-channels::chan_sip.c
  1.3       +88 -3     ports/net/asterisk/files/patch-channels::chan_zap.c