adding new code should require following the formatting guidelines :-)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@28214 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index db4bedae824038cafc9654c58ee41a1c82d808c1..098e5503394cea903935f3a30b1a9cda93d6b3f9 100644 (file)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -13752,8 +13752,8 @@ static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struc
        }
        if (changed && !ast_test_flag(&p->flags[0], SIP_GOTREFER)) {
                if (chan->_state != AST_STATE_UP) {
-                               char iabuf[INET_ADDRSTRLEN];
-                               ast_log(LOG_DEBUG, "Early media setting SIP '%s' - Sending early media to %s\n", p->callid, ast_inet_ntoa(iabuf, sizeof(iabuf), rtp ? p->redirip.sin_addr : p->ourip));
+                       char iabuf[INET_ADDRSTRLEN];
+                       ast_log(LOG_DEBUG, "Early media setting SIP '%s' - Sending early media to %s\n", p->callid, ast_inet_ntoa(iabuf, sizeof(iabuf), rtp ? p->redirip.sin_addr : p->ourip));
                } else if (!p->pendinginvite) {
                        if (option_debug > 2) {
                                char iabuf[INET_ADDRSTRLEN];