[asterisk-commits] kpfleming: branch 1.4 r223692 - /branches/1.4/channels/chan_sip.c

[SVN commits to the Asterisk project]

Author: kpfleming
Date: Mon Oct 12 10:30:40 2009
New Revision: 223692

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=223692
Log:
Remove automatic switching from T.38 to voice mode in chan_sip.
  
chan_sip has some code to automatically switch from T.38 mode to voice mode when
a voice frame is written to the channel while it is in T.38 mode; this was
intended to handle the situation when a FAX transmission has ended and the channel
is not yet hung up, but is causing problems at the beginning of FAX sessions as
well when there are still voice frames 'in flight' at the time the T.38 negotiation
completes. This patch removes the automatic switchover.
  
(issue #16025)
Reported by: jamicque


Modified:
    branches/1.4/channels/chan_sip.c

Modified: branches/1.4/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/branches/1.4/channels/chan_sip.c?view=diff&rev=223692&r1=223691&r2=223692
==============================================================================
--- branches/1.4/channels/chan_sip.c (original)
+++ branches/1.4/channels/chan_sip.c Mon Oct 12 10:30:40 2009
@@ -3910,8 +3910,7 @@
 						ast_set_flag(&p->flags[0], SIP_PROGRESS_SENT);
 					}
 				} else if (p->t38.state == T38_ENABLED && !p->t38.direct) {
-					p->t38.state = T38_DISABLED;
-					transmit_reinvite_with_sdp(p);
+					/* drop frame, can't sent VOICE frames while in T.38 mode */
 				} else {
 					p->lastrtptx = time(NULL);
 					res = ast_rtp_write(p->rtp, frame);