[asterisk-commits] kmoore: trunk r328937 - in /trunk: ./ channels/chan_sip.c

[SVN commits to the Asterisk project]

Author: kmoore
Date: Wed Jul 20 14:03:17 2011
New Revision: 328937

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=328937
Log:
Merged revisions 328936 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/2.0

................
  r328936 | kmoore | 2011-07-20 14:01:37 -0500 (Wed, 20 Jul 2011) | 15 lines
  
  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:
    trunk/   (props changed)
    trunk/channels/chan_sip.c

Propchange: trunk/
------------------------------------------------------------------------------
--- branch-2.0-merged (original)
+++ branch-2.0-merged Wed Jul 20 14:03:17 2011
@@ -1,1 +1,1 @@
-/branches/2.0:1-328075,328120,328162,328207,328247,328317,328329,328428-328429,328448,328451,328541,328609,328611,328664,328717,328771,328824,328879
+/branches/2.0:1-328075,328120,328162,328207,328247,328317,328329,328428-328429,328448,328451,328541,328609,328611,328664,328717,328771,328824,328879,328936

Modified: trunk/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/chan_sip.c?view=diff&rev=328937&r1=328936&r2=328937
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Wed Jul 20 14:03:17 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);