-- Executing NoOp("SIP/jon-0817d508", "Exchange Test: 666666") in new stack -- Executing Dial("SIP/jon-0817d508", "SIP/pbx.elephantoutlook.com:5070/666666") in new stack We're at 66.192.107.225 port 13976 Video is at 66.192.107.225 port 14142 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x80000 (h263) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 13 lines Reliably Transmitting (no NAT) to 66.192.107.225:5070: INVITE sip:666666@pbx.elephantoutlook.com:5070 SIP/2.0 Via: SIP/2.0/UDP 66.192.107.225:5060;branch=z9hG4bK313e9ff7;rport From: "Jon Webster" ;tag=as5c3450e3 To: Contact: Call-ID: 4d1f33b47f8cbc2237b44eb051f1f17e@66.192.107.225 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 07 Dec 2006 07:19:50 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 291 v=0 o=root 8640 8640 IN IP4 66.192.107.225 s=session c=IN IP4 66.192.107.225 t=0 0 m=audio 13976 RTP/AVP 0 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - m=video 14142 RTP/AVP 34 a=rtpmap:34 H263/90000 --- -- Called pbx.elephantoutlook.com:5070/666666 pbx*CLI> <-- SIP read from 66.192.107.225:5070: SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 66.192.107.225:5060;branch=z9hG4bK313e9ff7;rport=5060 From: "Jon Webster" ;tag=as5c3450e3 To: Call-ID: 4d1f33b47f8cbc2237b44eb051f1f17e@66.192.107.225 CSeq: 102 INVITE Server: OpenSer (1.1.0-notls (i386/linux)) Content-Length: 0 Warning: 392 66.192.107.225:5070 "Noisy feedback tells: pid=20826 req_src_ip=66.192.107.225 req_src_port=5060 in_uri=sip:666666@pbx.elephantoutlook.com:5070 out_uri=sip:666666@pbx.elephantoutlook.com:5070 via_cnt==1" --- (9 headers 0 lines)--- pbx*CLI> <-- SIP read from 66.192.107.225:5070: SIP/2.0 500 I'm terribly sorry, server error occurred (1/SL) Via: SIP/2.0/UDP 66.192.107.225:5060;branch=z9hG4bK313e9ff7;rport=5060 From: "Jon Webster" ;tag=as5c3450e3 To: ;tag=ff0593a3ce80005043f79ea67dccae1c.3701 Call-ID: 4d1f33b47f8cbc2237b44eb051f1f17e@66.192.107.225 CSeq: 102 INVITE Server: OpenSer (1.1.0-notls (i386/linux)) Content-Length: 0 Warning: 392 66.192.107.225:5070 "Noisy feedback tells: pid=20826 req_src_ip=66.192.107.225 req_src_port=5060 in_uri=sip:666666@pbx.elephantoutlook.com:5070 out_uri=sip:666666@pbx.elephantoutlook.com:5070 via_cnt==1" --- (9 headers 0 lines)--- -- Got SIP response 500 "I'm terribly sorry, server error occurred (1/SL)" back from 66.192.107.225 Transmitting (no NAT) to 66.192.107.225:5070: ACK sip:666666@pbx.elephantoutlook.com:5070 SIP/2.0 Via: SIP/2.0/UDP 66.192.107.225:5060;branch=z9hG4bK313e9ff7;rport From: "Jon Webster" ;tag=as5c3450e3 To: ;tag=ff0593a3ce80005043f79ea67dccae1c.3701 Contact: Call-ID: 4d1f33b47f8cbc2237b44eb051f1f17e@66.192.107.225 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- SIP/pbx.elephantoutlook.com:5070-0823d1a0 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) == Auto fallthrough, channel 'SIP/jon-0817d508' status is 'CONGESTION' pbx*CLI> <-- SIP read from 66.192.107.225:5070: SIP/2.0 302 Moved Temporarily FROM: "Jon Webster";tag=as5c3450e3 TO: ;tag=3c2de362d0 CSEQ: 102 INVITE CALL-ID: 4d1f33b47f8cbc2237b44eb051f1f17e@66.192.107.225 MAX-FORWARDS: 70 VIA: SIP/2.0/UDP 66.192.107.225:5060;branch=z9hG4bK313e9ff7;rport=5060 CONTENT-LENGTH: 0 SERVER: RTCC/2.0.6010.0 CONTACT: --- (10 headers 0 lines)--- -- Got SIP response 302 "Moved Temporarily" back from 66.192.107.225 Transmitting (no NAT) to 66.192.107.225:5070: ACK sip:666666@pbx.elephantoutlook.com:5070 SIP/2.0 Via: SIP/2.0/UDP 66.192.107.225:5060;branch=z9hG4bK313e9ff7;rport From: "Jon Webster" ;tag=as5c3450e3 To: ;tag=3c2de362d0 Contact: Call-ID: 4d1f33b47f8cbc2237b44eb051f1f17e@66.192.107.225 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Destroying call '4d1f33b47f8cbc2237b44eb051f1f17e@66.192.107.225' pbx*CLI> <-- SIP read from 66.192.107.225:5070: OPTIONS sip:pbx.elephantoutlook.com:5060;transport=TCP SIP/2.0 Record-Route: FROM: ;epid=8C-D4-0D-19-22;tag=1ed053927e TO: CSEQ: 312 OPTIONS CALL-ID: 2b0405658d76427999f7dec5c813028b MAX-FORWARDS: 69 Via: SIP/2.0/UDP 66.192.107.225:5070;branch=z9hG4bKc87c.cfc2eb.0;i=9 VIA: SIP/2.0/TCP 66.192.106.194:3437;branch=z9hG4bKffb45234 ACCEPT: application/sdp CONTACT: CONTENT-LENGTH: 0 USER-AGENT: RTCC/2.0.6010.0 --- (13 headers 0 lines)--- Looking for s in local (domain pbx.elephantoutlook.com) Transmitting (no NAT) to 66.192.107.225:5070: SIP/2.0 200 OK Via: SIP/2.0/UDP 66.192.107.225:5070;branch=z9hG4bKc87c.cfc2eb.0;i=9;received=66.192.107.225 Via: SIP/2.0/TCP 66.192.106.194:3437;branch=z9hG4bKffb45234 Record-Route: From: ;epid=8C-D4-0D-19-22;tag=1ed053927e To: ;tag=as5f48131a Call-ID: 2b0405658d76427999f7dec5c813028b CSeq: 312 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Accept: application/sdp Content-Length: 0 --- Destroying call '2b0405658d76427999f7dec5c813028b' pbx*CLI> <-- SIP read from 66.192.107.225:5070: OPTIONS sip:pbx.elephantoutlook.com:5060;transport=TCP SIP/2.0 Record-Route: FROM: ;epid=8C-D4-0D-19-22;tag=7e9f7b41ce TO: CSEQ: 313 OPTIONS CALL-ID: b63c8e8c7d6c40f8825c34c689937a10 MAX-FORWARDS: 69 Via: SIP/2.0/UDP 66.192.107.225:5070;branch=z9hG4bK9b25.c3bddd12.0;i=a VIA: SIP/2.0/TCP 66.192.106.194:3446;branch=z9hG4bK68c390d5 ACCEPT: application/sdp CONTACT: CONTENT-LENGTH: 0 USER-AGENT: RTCC/2.0.6010.0 --- (13 headers 0 lines)--- Looking for s in local (domain pbx.elephantoutlook.com) Transmitting (no NAT) to 66.192.107.225:5070: SIP/2.0 200 OK Via: SIP/2.0/UDP 66.192.107.225:5070;branch=z9hG4bK9b25.c3bddd12.0;i=a;received=66.192.107.225 Via: SIP/2.0/TCP 66.192.106.194:3446;branch=z9hG4bK68c390d5 Record-Route: From: ;epid=8C-D4-0D-19-22;tag=7e9f7b41ce To: ;tag=as5b3ab11b Call-ID: b63c8e8c7d6c40f8825c34c689937a10 CSeq: 313 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Accept: application/sdp Content-Length: 0 --- Destroying call 'b63c8e8c7d6c40f8825c34c689937a10' pbx*CLI> <-- SIP read from 66.192.107.225:5070: OPTIONS sip:pbx.elephantoutlook.com:5060;transport=TCP SIP/2.0 Record-Route: FROM: ;epid=8C-D4-0D-19-22;tag=8a6030c58c TO: CSEQ: 314 OPTIONS CALL-ID: 238060bb462b4b6fbdee9352fb924389 MAX-FORWARDS: 69 Via: SIP/2.0/UDP 66.192.107.225:5070;branch=z9hG4bKf87e.5be96a12.0;i=b VIA: SIP/2.0/TCP 66.192.106.194:3453;branch=z9hG4bK197a6ad9 ACCEPT: application/sdp CONTACT: CONTENT-LENGTH: 0 USER-AGENT: RTCC/2.0.6010.0 --- (13 headers 0 lines)--- Looking for s in local (domain pbx.elephantoutlook.com) Transmitting (no NAT) to 66.192.107.225:5070: SIP/2.0 200 OK Via: SIP/2.0/UDP 66.192.107.225:5070;branch=z9hG4bKf87e.5be96a12.0;i=b;received=66.192.107.225 Via: SIP/2.0/TCP 66.192.106.194:3453;branch=z9hG4bK197a6ad9 Record-Route: From: ;epid=8C-D4-0D-19-22;tag=8a6030c58c To: ;tag=as4226570b Call-ID: 238060bb462b4b6fbdee9352fb924389 CSeq: 314 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Accept: application/sdp Content-Length: 0 --- Destroying call '238060bb462b4b6fbdee9352fb924389' pbx*CLI>