Below is a 'sip debug' dump from my Asterisk 1.2 (behind NAT) to the TalkShoe service. The talkshoe peer: [talkshoe] type = peer host = 66.212.134.192 fromuser = 123 disallow = all allow=ulaw qualify=yes dtmfmode=auto insecure = very The dialplan line: exten => 733,n,Dial(SIP/talkshoe,60,) -- Hungup 'Zap/3-1' -- Starting simple switch on 'Zap/3-1' -- Executing NoOp("Zap/3-1", "") in new stack -- Executing Dial("Zap/3-1", "SIP/talkshoe|60|") in new stack We're at 192.168.67.18 port 14828 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 10 lines Reliably Transmitting (no NAT) to 66.212.134.192:5060: INVITE sip:66.212.134.192 SIP/2.0 Via: SIP/2.0/UDP 192.168.67.18:5060;branch=z9hG4bK77b0646e;rport From: "Channel 3" ;tag=as030a5bf1 To: Contact: Call-ID: 2705765123bad3291632549955ad21f3@192.168.67.18 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 06 Jul 2007 11:29:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 218 v=0 o=root 29760 29760 IN IP4 192.168.67.18 s=session c=IN IP4 192.168.67.18 t=0 0 m=audio 14828 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Called talkshoe pungenday*CLI> <-- SIP read from 66.212.134.192:5060: SIP/2.0 180 Ringing From: "Channel 3";tag=as030a5bf1 To: ;tag=a000217-13c4-468df02d-378793b7-532ab8 Call-ID: 2705765123bad3291632549955ad21f3@192.168.67.18 CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.67.18:5060;received=212.179.75.202;rport=5060;branch=z9hG4bK77b0646e Contact: Content-Length: 0 --- (8 headers 0 lines) --- -- SIP/talkshoe-08149a88 is ringing pungenday*CLI> <-- SIP read from 66.212.134.192:5060: SIP/2.0 403 Forbidden From: "Channel 3";tag=as030a5bf1 To: ;tag=a000217-13c4-468df02d-378793b7-532ab8 Call-ID: 2705765123bad3291632549955ad21f3@192.168.67.18 CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.67.18:5060;received=212.179.75.202;rport=5060;branch=z9hG4bK77b0646e Content-Length: 0 --- (7 headers 0 lines) --- Transmitting (no NAT) to 66.212.134.192:5060: ACK sip:66.212.134.192 SIP/2.0 Via: SIP/2.0/UDP 192.168.67.18:5060;branch=z9hG4bK77b0646e;rport From: "Channel 3" ;tag=as030a5bf1 To: ;tag=a000217-13c4-468df02d-378793b7-532ab8 Contact: Call-ID: 2705765123bad3291632549955ad21f3@192.168.67.18 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Jul 6 14:29:59 WARNING[29774]: chan_sip.c:9847 handle_response_invite: Forbidden - wrong password on authentication for INVITE to '"Channel 3" ;tag=as030a5bf1' -- SIP/talkshoe-08149a88 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) == Auto fallthrough, channel 'Zap/3-1' status is 'CONGESTION' Destroying call '2705765123bad3291632549955ad21f3@192.168.67.18' -- Hungup 'Zap/3-1' pungenday*CLI> sip no debug SIP Debugging Disabled