[asterisk-commits] file: branch 13 r423209 - in /branches/13: ./ res/res_rtp_asterisk.c

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Tue Sep 16 15:35:36 CDT 2014


Author: file
Date: Tue Sep 16 15:35:34 2014
New Revision: 423209

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=423209
Log:
res_rtp_asterisk: Fix building when pjproject is not used.
........

Merged revisions 423207 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 423208 from http://svn.asterisk.org/svn/asterisk/branches/12

Modified:
    branches/13/   (props changed)
    branches/13/res/res_rtp_asterisk.c

Propchange: branches/13/
------------------------------------------------------------------------------
Binary property 'branch-12-merged' - no diff available.

Modified: branches/13/res/res_rtp_asterisk.c
URL: http://svnview.digium.com/svn/asterisk/branches/13/res/res_rtp_asterisk.c?view=diff&rev=423209&r1=423208&r2=423209
==============================================================================
--- branches/13/res/res_rtp_asterisk.c (original)
+++ branches/13/res/res_rtp_asterisk.c Tue Sep 16 15:35:34 2014
@@ -2476,8 +2476,10 @@
 static int ast_rtp_destroy(struct ast_rtp_instance *instance)
 {
 	struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
+#ifdef HAVE_PJPROJECT
 	struct timeval wait = ast_tvadd(ast_tvnow(), ast_samp2tv(TURN_STATE_WAIT_TIME, 1000));
 	struct timespec ts = { .tv_sec = wait.tv_sec, .tv_nsec = wait.tv_usec * 1000, };
+#endif
 
 	/* Destroy the smoother that was smoothing out audio if present */
 	if (rtp->smoother) {




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