Issue #5954 revisited. Thanks rizzo.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@8731 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 2ddf23e..aa5c23f 100644 (file)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -8671,7 +8671,7 @@ static int sip_show_history(int fd, int argc, char *argv[])
                                ast_cli(fd, "  * SIP Call\n");
                        if (cur->history)
                                AST_LIST_TRAVERSE(cur->history, hist, list)
-                                       ast_cli(fd, "%d. %s\n", x++, hist->event);
+                                       ast_cli(fd, "%d. %s\n", ++x, hist->event);
                        if (x == 0)
                                ast_cli(fd, "Call '%s' has no history\n", cur->callid);
                        found++;
@@ -8700,7 +8700,7 @@ void sip_dump_history(struct sip_pvt *dialog)
                ast_log(LOG_DEBUG, "  * SIP Call\n");
        if (dialog->history)
                AST_LIST_TRAVERSE(dialog->history, hist, list)
-                       ast_log(LOG_DEBUG, "  %d. %s\n", x++, hist->event);
+                       ast_log(LOG_DEBUG, "  %d. %s\n", ++x, hist->event);
        if (!x)
                ast_log(LOG_DEBUG, "Call '%s' has no history\n", dialog->callid);
        ast_log(LOG_DEBUG, "\n---------- END SIP HISTORY for '%s' \n", dialog->callid);