Want to stop incoming traffic to FreeSwitch. Followed instructions at :http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#sip-options-respon...: Added <param name="sip-options-respond-503-on-busy" value="true"/> under <global_settings> in sofial.comf.xml, called "fsctl pause inbound". Freeswitch replies with 500 instead of 503 to OPTIONS. As a result, SBC continues to send INVITEs to FreeSwitch, which are rejected with 503. Did I miss something? If not, how can I debug this? Thank you. A Running FreeSWITCH Version 1.3.0 (Tue, 28 Aug 2012). INVITE - gets a 503 INVITE sip:yyyyyyy@devsip01-5070:5070;maddr=dev302-servers SIP/2.0 Via: SIP/2.0/UDP 10.3.220.52:5060;branch=z9hG4bKabu9dq10agch4ioi2300.2 Max-Forwards: 69 From: "Adelia C" <sip:xxxxxxx@10.3.220.52>;tag=as719a49d0 To: <sip:yyyyyyy@10.3.212.211> Contact: <sip:xxxxxxx@10.3.220.52:5060;transport=udp> Call-ID: 5c60645f045c6f9f62dd14b37d1eabfb@10.3.10.94 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.1 Date: Fri, 01 Mar 2013 02:15:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Type: application/sdp Content-Length: 259 v=0 o=root 969673395 969673395 IN IP4 10.3.220.52 s=Asterisk PBX 1.6.1.1 c=IN IP4 10.3.220.52 t=0 0 m=audio 50452 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv OPTIONS - gets a 500 OPTIONS sip:ashdevsip01-5070:5070 SIP/2.0 Via: SIP/2.0/UDP 10.3.220.41:5060;branch=z9hG4bK1rl6s2109ogg1jk9b2a0 Call-ID: c0b302e8be810ef50a9cadf8798911080g7ukv0@10.3.220.41 To: sip:ping@10.3.212.211 From: <sip:ping@10.3.220.41>;tag=e99174716b908c0877e5a74acceb78f30g7ukv0 Max-Forwards: 70 CSeq: 1986367 OPTIONS Route: <sip:10.3.212.211:5070;lr>