]> git.ipfire.org Git - thirdparty/asterisk.git/commitdiff
Add a log message that was requested by the masses in the developer tutorial
authorRussell Bryant <russell@russellbryant.com>
Wed, 26 Sep 2007 21:15:15 +0000 (21:15 +0000)
committerRussell Bryant <russell@russellbryant.com>
Wed, 26 Sep 2007 21:15:15 +0000 (21:15 +0000)
session at Astricon.  chan_sip did not output any message when a call was
rejected because the extension was not found.  This adds a verbose message
(at verbose level 3) to note when this happens.

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@83941 65c4cc65-6c06-0410-ace0-fbb531ad65f3

channels/chan_sip.c

index ec06ca6499a5c80e517fccb6ee782a8beaed23f6..21060ec4e879bb9e148ad85d3f8aaef5513a29e8 100644 (file)
@@ -13659,8 +13659,14 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
                if (!replace_id && gotdest) {   /* No matching extension found */
                        if (gotdest == 1 && ast_test_flag(&p->flags[1], SIP_PAGE2_ALLOWOVERLAP))
                                transmit_response_reliable(p, "484 Address Incomplete", req);
-                       else
+                       else {
                                transmit_response_reliable(p, "404 Not Found", req);
+                               if (option_verbose > 2) {
+                                       ast_verbose(VERBOSE_PREFIX_3 "Call from '%s' to extension"
+                                               " '%s' rejected because extension not found.\n",
+                                               S_OR(p->username, p->peername), p->exten);
+                               }
+                       }
                        p->invitestate = INV_COMPLETED; 
                        update_call_counter(p, DEC_CALL_LIMIT);
                        sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);