Author: rizzo
Date: Fri Jul 20 17:55:15 2007
New Revision: 76218
URL: http://svn.digium.com/view/asterisk?view=rev&rev=76218
Log:
another round of diff reductions wrt trunk
Modified:
team/rizzo/astobj2/channels/chan_sip.c
Modified: team/rizzo/astobj2/channels/chan_sip.c
URL:
http://svn.digium.com/view/asterisk/team/rizzo/astobj2/channels/chan_sip.c?view=diff&rev=76218&r1=76217&r2=76218
==============================================================================
--- team/rizzo/astobj2/channels/chan_sip.c (original)
+++ team/rizzo/astobj2/channels/chan_sip.c Fri Jul 20 17:55:15 2007
@@ -15349,7 +15349,7 @@
pbx_builtin_setvar_helper(current.chan2,
"_SIPTRANSFER_REFERER", p->refer->referred_by);
}
/* Generate a Replaces string to be used in the INVITE during attended
transfer */
- if (p->refer->replaces_callid &&
!ast_strlen_zero(p->refer->replaces_callid)) {
+ if (!ast_strlen_zero(p->refer->replaces_callid)) {
char tempheader[BUFSIZ];
snprintf(tempheader, sizeof(tempheader), "%s%s%s%s%s",
p->refer->replaces_callid,
p->refer->replaces_callid_totag ? ";to-tag=" :
"",
@@ -16400,7 +16400,8 @@
return;
/* Check AUDIO RTP keepalives */
- if (dialog->lastrtptx && ast_rtp_get_rtpkeepalive(dialog->rtp) && (t >
dialog->lastrtptx + ast_rtp_get_rtpkeepalive(dialog->rtp))) {
+ if (dialog->lastrtptx && ast_rtp_get_rtpkeepalive(dialog->rtp) &&
+ (t > dialog->lastrtptx +
ast_rtp_get_rtpkeepalive(dialog->rtp))) {
/* Need to send an empty RTP packet */
dialog->lastrtptx = time(NULL);
ast_rtp_sendcng(dialog->rtp, 0);
@@ -18808,7 +18809,7 @@
#endif
ast_mutex_lock(&monlock);
- if (monitor_thread && (monitor_thread != AST_PTHREADT_STOP)) {
+ if (monitor_thread && (monitor_thread != AST_PTHREADT_STOP) &&
(monitor_thread != AST_PTHREADT_NULL)) {
pthread_cancel(monitor_thread);
pthread_kill(monitor_thread, SIGURG);
pthread_join(monitor_thread, NULL);
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