[asterisk-commits] file: branch 1.4 r69794 - /branches/1.4/channels/chan_sip.c

[SVN commits to the Asterisk project]

Author: file
Date: Mon Jun 18 14:00:50 2007
New Revision: 69794

URL: http://svn.digium.com/view/asterisk?view=rev&rev=69794
Log:
Don't count RTP timeout when involved in a T38 fax session. (issue #9222 reported by ivoc)

Modified:
    branches/1.4/channels/chan_sip.c

Modified: branches/1.4/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/branches/1.4/channels/chan_sip.c?view=diff&rev=69794&r1=69793&r2=69794
==============================================================================
--- branches/1.4/channels/chan_sip.c (original)
+++ branches/1.4/channels/chan_sip.c Mon Jun 18 14:00:50 2007
@@ -15176,7 +15176,8 @@
 			/* Check RTP timeouts and kill calls if we have a timeout set and do not get RTP */
 			if (sip->rtp && sip->owner &&
 			    (sip->owner->_state == AST_STATE_UP) &&
-			    !sip->redirip.sin_addr.s_addr) {
+			    !sip->redirip.sin_addr.s_addr &&
+			    sip->t38.state != T38_ENABLED) {
 				if (sip->lastrtptx &&
 				    ast_rtp_get_rtpkeepalive(sip->rtp) &&
 				    (t > sip->lastrtptx + ast_rtp_get_rtpkeepalive(sip->rtp))) {