Verbosity is at least 10 -- Remote UNIX connection Reliably Transmitting (no NAT) to 64.61.93.87:5060: OPTIONS sip:connect01.voicepulse.com SIP/2.0 Via: SIP/2.0/UDP 10.18.18.94:5060;branch=z9hG4bK24de7733;rport From: "asterisk" ;tag=as54f4eebb To: Contact: Call-ID: 59af2972309e712b3f304d4a457f2f9b@10.18.18.94 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 26 Jun 2007 22:00:27 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- asteriskpbx*CLI> <--- SIP read from 64.61.93.87:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.18.18.94:5060;branch=z9hG4bK24de7733;received=216.207.245.1;rport=5060 From: "asterisk" ;tag=as54f4eebb To: ;tag=as4dd37b10 Call-ID: 59af2972309e712b3f304d4a457f2f9b@10.18.18.94 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Accept: application/sdp Content-Length: 0 <-------------> --- (11 headers 0 lines) --- [Jun 26 22:00:27] NOTICE[314]: chan_sip.c:12280 handle_response_peerpoke: Peer 'trunk_1' is now Reachable. (43ms / 2000ms) Really destroying SIP dialog '59af2972309e712b3f304d4a457f2f9b@10.18.18.94' Method: OPTIONS asteriskpbx*CLI> <--- SIP read from 192.168.69.52:5060 ---> INVITE sip:914232405159@192.168.69.1:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.69.52;branch=z9hG4bK3d1d9db1740104D6 From: "New User" ;tag=3AC39282-E22FAE3 To: CSeq: 1 INVITE Call-ID: 6c383de7-b0638734-14af0dcd@192.168.69.52 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundStationIP-SSIP_4000-UA/2.1.1.0052 Supported: 100rel,replaces Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 225 v=0 o=- 1182895236 1182895236 IN IP4 192.168.69.52 s=Polycom IP Phone c=IN IP4 192.168.69.52 t=0 0 m=audio 2234 RTP/AVP 0 8 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 <-------------> --- (14 headers 10 lines) --- Sending to 192.168.69.52 : 5060 (no NAT) Using INVITE request as basis request - 6c383de7-b0638734-14af0dcd@192.168.69.52 Found peer '6001' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 192.168.69.52:2234 Found description format PCMU for ID 0 Found description format PCMA for ID 8 Found description format telephone-event for ID 101 Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.69.52:2234 Looking for 914232405159 in numberplan-custom-1 (domain 192.168.69.1) list_route: hop: <--- Transmitting (no NAT) to 192.168.69.52:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.69.52;branch=z9hG4bK3d1d9db1740104D6;received=192.168.69.52 From: "New User" ;tag=3AC39282-E22FAE3 To: Call-ID: 6c383de7-b0638734-14af0dcd@192.168.69.52 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Access-URL: Content-Length: 0 <------------> -- Executing [914232405159@numberplan-custom-1:1] Macro("SIP/6001-008ea26c", "trunkdial|SIP/trunk_1/14232405159") in new stack -- Executing [s@macro-trunkdial:1] Dial("SIP/6001-008ea26c", "SIP/trunk_1/14232405159") in new stack Audio is at 10.18.18.94 port 17640 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=z9hG4bK4b7c7dd2;rport From: "New User" ;tag=as46e9f30d To: Contact: Call-ID: 6586323f20b32df879fa4923728fa9ca@10.18.18.94 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 26 Jun 2007 22:00:43 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 373 v=0 o=root 893 893 IN IP4 10.18.18.94 s=session c=IN IP4 10.18.18.94 t=0 0 m=audio 17640 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=z9hG4bK4b7c7dd2;received=216.207.245.1;rport=5060 From: "New User" ;tag=as46e9f30d To: ;tag=as1f0ace78 Call-ID: 6586323f20b32df879fa4923728fa9ca@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="3e54ed8a" 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=z9hG4bK4b7c7dd2;rport From: "New User" ;tag=as46e9f30d To: ;tag=as1f0ace78 Contact: Call-ID: 6586323f20b32df879fa4923728fa9ca@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 17640 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=z9hG4bK1d093484;rport From: "New User" ;tag=as46e9f30d To: Contact: Call-ID: 6586323f20b32df879fa4923728fa9ca@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="3e54ed8a", response="1d453577ab427d302df38a39327c8ed5", opaque="" Date: Tue, 26 Jun 2007 22:00:43 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 373 v=0 o=root 893 894 IN IP4 10.18.18.94 s=session c=IN IP4 10.18.18.94 t=0 0 m=audio 17640 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=z9hG4bK1d093484;received=216.207.245.1;rport=5060 From: "New User" ;tag=as46e9f30d To: ;tag=as1f0ace78 Call-ID: 6586323f20b32df879fa4923728fa9ca@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=z9hG4bK1d093484;rport From: "New User" ;tag=as46e9f30d To: ;tag=as1f0ace78 Contact: Call-ID: 6586323f20b32df879fa4923728fa9ca@10.18.18.94 CSeq: 103 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Access-URL: Content-Length: 0 --- -- SIP/trunk_1-008eb7f0 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Executing [s@macro-trunkdial:2] Goto("SIP/6001-008ea26c", "s-CONGESTION|1") in new stack -- Goto (macro-trunkdial,s-CONGESTION,1) -- Executing [s-CONGESTION@macro-trunkdial:1] NoOp("SIP/6001-008ea26c", "") in new stack == Auto fallthrough, channel 'SIP/6001-008ea26c' status is 'CONGESTION' <--- Transmitting (no NAT) to 192.168.69.52:5060 ---> SIP/2.0 503 Service Unavailable Via: SIP/2.0/UDP 192.168.69.52;branch=z9hG4bK3d1d9db1740104D6;received=192.168.69.52 From: "New User" ;tag=3AC39282-E22FAE3 To: ;tag=as6ae5e9e7 Call-ID: 6c383de7-b0638734-14af0dcd@192.168.69.52 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Access-URL: Content-Length: 0 X-Asterisk-HangupCause: Unallocated (unassigned) number X-Asterisk-HangupCauseCode: 1 <------------> Really destroying SIP dialog '6586323f20b32df879fa4923728fa9ca@10.18.18.94' Method: INVITE Really destroying SIP dialog '6c383de7-b0638734-14af0dcd@192.168.69.52' Method: INVITE asteriskpbx*CLI> <--- SIP read from 192.168.69.52:5060 ---> ACK sip:914232405159@192.168.69.1:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.69.52;branch=z9hG4bK3d1d9db1740104D6 From: "New User" ;tag=3AC39282-E22FAE3 To: ;tag=as6ae5e9e7 CSeq: 1 ACK Call-ID: 6c383de7-b0638734-14af0dcd@192.168.69.52 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundStationIP-SSIP_4000-UA/2.1.1.0052 Max-Forwards: 70 Content-Length: 0 <-------------> --- (11 headers 0 lines) ---