Add history events for re-invites
authorOlle Johansson <oej@edvina.net>
Fri, 7 Apr 2006 19:46:50 +0000 (19:46 +0000)
committerOlle Johansson <oej@edvina.net>
Fri, 7 Apr 2006 19:46:50 +0000 (19:46 +0000)
(need to nail this issue...)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@18373 65c4cc65-6c06-0410-ace0-fbb531ad65f3

channels/chan_sip.c

index 63b117b..85ec6dd 100644 (file)
@@ -2078,7 +2078,8 @@ static int sip_call(struct ast_channel *ast, char *dest, int timeout)
        
        res = 0;
        ast_set_flag(&p->flags[0], SIP_OUTGOING);
-       ast_log(LOG_DEBUG, "Outgoing Call for %s\n", p->username);
+       if (option_debug)
+               ast_log(LOG_DEBUG, "Outgoing Call for %s\n", p->username);
        res = update_call_counter(p, INC_CALL_LIMIT);
        if ( res != -1 ) {
                p->callingpres = ast->cid.cid_pres;
@@ -4731,6 +4732,8 @@ static int transmit_reinvite_with_sdp(struct sip_pvt *p)
        add_header(&req, "Allow", ALLOWED_METHODS);
        if (sipdebug)
                add_header(&req, "X-asterisk-info", "SIP re-invite (RTP bridge)");
+       if (recordhistory)
+               append_history(p, "%s", "Re-invite sent");
        add_sdp(&req, p);
        /* Use this as the basis */
        copy_request(&p->initreq, &req);
@@ -10701,6 +10704,8 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
                                p->jointcapability = p->capability;
                                ast_log(LOG_DEBUG, "Hm....  No sdp for the moment\n");
                        }
+                       if (recordhistory) /* This is a response, note what it was for */
+                               append_history(p, "%s", "Re-invite received");
                }
        } else if (debug)
                ast_verbose("Ignoring this INVITE request\n");