[asterisk-commits] mnicholson: branch 1.6.2 r322668 - in /branches/1.6.2: ./ channels/chan_sip.c

[SVN commits to the Asterisk project]

Author: mnicholson
Date: Thu Jun  9 10:22:50 2011
New Revision: 322668

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=322668
Log:
Merged revisions 322646 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r322646 | mnicholson | 2011-06-09 10:10:30 -0500 (Thu, 09 Jun 2011) | 5 lines
  
  don't drop any voice frames when checking for T.38 during early media
  
  (closes ASTERISK-17705)
  Review: https://reviewboard.asterisk.org/r/1186/
........

Modified:
    branches/1.6.2/   (props changed)
    branches/1.6.2/channels/chan_sip.c

Propchange: branches/1.6.2/
------------------------------------------------------------------------------
Binary property 'branch-1.4-merged' - no diff available.

Modified: branches/1.6.2/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/branches/1.6.2/channels/chan_sip.c?view=diff&rev=322668&r1=322667&r2=322668
==============================================================================
--- branches/1.6.2/channels/chan_sip.c (original)
+++ branches/1.6.2/channels/chan_sip.c Thu Jun  9 10:22:50 2011
@@ -6509,7 +6509,10 @@
 		}
 		if (p) {
 			sip_pvt_lock(p);
-			if (p->rtp) {
+			if (p->t38.state == T38_ENABLED && !p->t38.direct) {
+				/* drop frame, can't sent VOICE frames while in T.38 mode */
+				break;
+			} else if (p->rtp) {
 				/* If channel is not up, activate early media session */
 				if ((ast->_state != AST_STATE_UP) &&
 				    !ast_test_flag(&p->flags[0], SIP_PROGRESS_SENT) &&
@@ -6520,12 +6523,9 @@
 						transmit_provisional_response(p, "183 Session Progress", &p->initreq, TRUE);
 						ast_set_flag(&p->flags[0], SIP_PROGRESS_SENT);
 					}
-				} else if (p->t38.state == T38_ENABLED) {
-					/* drop frame, can't sent VOICE frames while in T.38 mode */
-				} else {
-					p->lastrtptx = time(NULL);
-					res = ast_rtp_write(p->rtp, frame);
 				}
+				p->lastrtptx = time(NULL);
+				res = ast_rtp_write(p->rtp, frame);
 			}
 			sip_pvt_unlock(p);
 		}