[asterisk-commits] kmoore: branch 2.0 r328936 - in /branches/2.0: ./ channels/chan_sip.c

[SVN commits to the Asterisk project]

Author: kmoore
Date: Wed Jul 20 14:01:37 2011
New Revision: 328936

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=328936
Log:
Merged revisions 328935 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r328935 | kmoore | 2011-07-20 14:00:23 -0500 (Wed, 20 Jul 2011) | 8 lines
  
  Inband DTMF regression
  
  The functionality of inband DTMF in chan_sip relied upon
  ast_rtp_instance_dtmf_mode_get/set not working properly to avoid calling
  ast_rtp_instance_dtmf_begin/end on RTP streams with inband DTMF. According to
  documentation, ast_rtp_instance_dtmf_begin/end is meant only for RFC2833 DTMF,
  never inband.  This fixes the regression introduced in revision 328823.
........

Modified:
    branches/2.0/   (props changed)
    branches/2.0/channels/chan_sip.c

Propchange: branches/2.0/
------------------------------------------------------------------------------
Binary property 'branch-1.8-merged' - no diff available.

Modified: branches/2.0/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/branches/2.0/channels/chan_sip.c?view=diff&rev=328936&r1=328935&r2=328936
==============================================================================
--- branches/2.0/channels/chan_sip.c (original)
+++ branches/2.0/channels/chan_sip.c Wed Jul 20 14:01:37 2011
@@ -6536,11 +6536,7 @@
 	sip_pvt_lock(p);
 	switch (ast_test_flag(&p->flags[0], SIP_DTMF)) {
 	case SIP_DTMF_INBAND:
-		if (p->rtp && ast_rtp_instance_dtmf_mode_get(p->rtp) == AST_RTP_DTMF_MODE_INBAND) {
-			ast_rtp_instance_dtmf_begin(p->rtp, digit);
-		} else {
-			res = -1; /* Tell Asterisk to generate inband indications */
-		}
+		res = -1; /* Tell Asterisk to generate inband indications */
 		break;
 	case SIP_DTMF_RFC2833:
 		if (p->rtp)
@@ -6572,11 +6568,7 @@
 			ast_rtp_instance_dtmf_end_with_duration(p->rtp, digit, duration);
 		break;
 	case SIP_DTMF_INBAND:
-		if (p->rtp && ast_rtp_instance_dtmf_mode_get(p->rtp) == AST_RTP_DTMF_MODE_INBAND) {
-			ast_rtp_instance_dtmf_end(p->rtp, digit);
-		} else {
-			res = -1; /* Tell Asterisk to stop inband indications */
-		}
+		res = -1; /* Tell Asterisk to stop inband indications */
 		break;
 	}
 	sip_pvt_unlock(p);