-- Starting simple switch on 'Zap/1-1' -- Executing [914232405159@numberplan-custom-1:1] Macro("Zap/1-1", "trunkdial|SIP/trunk_1/14232405159") in new stack -- Executing [s@macro-trunkdial:1] Dial("Zap/1-1", "SIP/trunk_1/14232405159") in new stack Audio is at 10.18.18.94 port 12754 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x10 (g726aal2) to SDP Adding codec 0x40 (slin) to SDP Adding codec 0x800 (g726) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 64.61.93.87:5060: INVITE sip:14232405159@connect01.voicepulse.com SIP/2.0 Via: SIP/2.0/UDP 10.18.18.94:5060;branch=z9hG4bK5fbe8397;rport From: "New User" ;tag=as4678a846 To: Contact: Call-ID: 29e9f45c03d6317204453df72034549a@10.18.18.94 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 26 Jun 2007 22:06:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 373 v=0 o=root 923 923 IN IP4 10.18.18.94 s=session c=IN IP4 10.18.18.94 t=0 0 m=audio 12754 RTP/AVP 0 3 8 112 10 111 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:112 AAL2-G726-32/8000 a=rtpmap:10 L16/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called trunk_1/14232405159 asteriskpbx*CLI> <--- SIP read from 64.61.93.87:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.18.18.94:5060;branch=z9hG4bK5fbe8397;received=216.207.245.1;rport=5060 From: "New User" ;tag=as4678a846 To: ;tag=as3ff7a6e0 Call-ID: 29e9f45c03d6317204453df72034549a@10.18.18.94 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Proxy-Authenticate: Digest realm="asterisk", nonce="6afabf39" Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Transmitting (no NAT) to 64.61.93.87:5060: ACK sip:14232405159@connect01.voicepulse.com SIP/2.0 Via: SIP/2.0/UDP 10.18.18.94:5060;branch=z9hG4bK5fbe8397;rport From: "New User" ;tag=as4678a846 To: ;tag=as3ff7a6e0 Contact: Call-ID: 29e9f45c03d6317204453df72034549a@10.18.18.94 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Access-URL: Content-Length: 0 --- Audio is at 10.18.18.94 port 12754 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x10 (g726aal2) to SDP Adding codec 0x40 (slin) to SDP Adding codec 0x800 (g726) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 64.61.93.87:5060: INVITE sip:14232405159@connect01.voicepulse.com SIP/2.0 Via: SIP/2.0/UDP 10.18.18.94:5060;branch=z9hG4bK1a97a88e;rport From: "New User" ;tag=as4678a846 To: Contact: Call-ID: 29e9f45c03d6317204453df72034549a@10.18.18.94 CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Access-URL: Proxy-Authorization: Digest username="ueD6XhR29T", realm="asterisk", algorithm=MD5, uri="sip:14232405159@connect01.voicepulse.com", nonce="6afabf39", response="d4b71bd9f76105feeaabc666b87c6042", opaque="" Date: Tue, 26 Jun 2007 22:06:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 373 v=0 o=root 923 924 IN IP4 10.18.18.94 s=session c=IN IP4 10.18.18.94 t=0 0 m=audio 12754 RTP/AVP 0 3 8 112 10 111 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:112 AAL2-G726-32/8000 a=rtpmap:10 L16/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- asteriskpbx*CLI> <--- SIP read from 64.61.93.87:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.18.18.94:5060;branch=z9hG4bK1a97a88e;received=216.207.245.1;rport=5060 From: "New User" ;tag=as4678a846 To: ;tag=as3ff7a6e0 Call-ID: 29e9f45c03d6317204453df72034549a@10.18.18.94 CSeq: 103 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Transmitting (no NAT) to 64.61.93.87:5060: ACK sip:14232405159@connect01.voicepulse.com SIP/2.0 Via: SIP/2.0/UDP 10.18.18.94:5060;branch=z9hG4bK1a97a88e;rport From: "New User" ;tag=as4678a846 To: ;tag=as3ff7a6e0 Contact: Call-ID: 29e9f45c03d6317204453df72034549a@10.18.18.94 CSeq: 103 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Access-URL: Content-Length: 0 --- -- SIP/trunk_1-008f06a0 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Executing [s@macro-trunkdial:2] Goto("Zap/1-1", "s-CONGESTION|1") in new stack -- Goto (macro-trunkdial,s-CONGESTION,1) -- Executing [s-CONGESTION@macro-trunkdial:1] NoOp("Zap/1-1", "") in new stack == Auto fallthrough, channel 'Zap/1-1' status is 'CONGESTION' Really destroying SIP dialog '29e9f45c03d6317204453df72034549a@10.18.18.94' Method: INVITE -- Hungup 'Zap/1-1'