Small fix

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@18797 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 52aa6288495a0e0d86cb2b01892d3f3e5c47a372..6a1c6c1591b260ccfa3d9db54a473fc71aa3f192 100644 (file)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -4730,7 +4730,7 @@ static int transmit_reinvite_with_sdp(struct sip_pvt *p)
        if (sipdebug)
                add_header(&req, "X-asterisk-info", "SIP re-invite (RTP bridge)");
        if (recordhistory)
-               append_history(p, "%s", "Re-invite sent");
+               append_history(p, "ReInv", "Re-invite sent");
        add_sdp(&req, p);
        /* Use this as the basis */
        copy_request(&p->initreq, &req);
@@ -10151,6 +10151,9 @@ static void handle_response(struct sip_pvt *p, int resp, char *rest, struct sip_
        } else {        
                /* Responses to OUTGOING SIP requests on INCOMING calls 
                   get handled here. As well as out-of-call message responses */
+               if (ast_test_flag(req, SIP_PKT_DEBUG))
+                       ast_verbose("SIP Response message for INCOMING dialog %s arrived\n", msg);
+
                if (resp == 200) {
                        /* Tags in early session is replaced by the tag in 200 OK, which is 
                        the final reply to our INVITE */