*CLI> Reliably Transmitting (no NAT) to 10.0.2.201:5060: OPTIONS sip:jason@10.0.2.201:5060 SIP/2.0 Via: SIP/2.0/UDP 10.0.2.231:5060;branch=z9hG4bK6cf925c5;rport From: "asterisk" ;tag=as36918c0a To: Contact: Call-ID: 76b6b5287d554a9825f52ba570884595@10.0.2.231 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 01 Feb 2007 03:11:47 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- <--- SIP read from 10.0.2.201:5060 ---> SIP/2.0 200 OK To: ;tag=3ca707916c62eb32i0 From: "asterisk" ;tag=as36918c0a Call-ID: 76b6b5287d554a9825f52ba570884595@10.0.2.231 CSeq: 102 OPTIONS Via: SIP/2.0/UDP 10.0.2.231:5060;branch=z9hG4bK6cf925c5 Server: Linksys/SPA942-5.1.5 Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces <-------------> --- (10 headers 0 lines) --- Really destroying SIP dialog '76b6b5287d554a9825f52ba570884595@10.0.2.231' Method: OPTIONS Reliably Transmitting (NAT) to 203.2.134.1:5060: OPTIONS sip:sip.internode.on.net SIP/2.0 Via: SIP/2.0/UDP 10.0.2.231:5060;branch=z9hG4bK483b77e1;rport From: "asterisk" ;tag=as1121982d To: Contact: Call-ID: 24f5716a19ee682615492ca4625a01cb@10.0.2.231 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 01 Feb 2007 03:11:47 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- <--- SIP read from 203.2.134.1:5060 ---> SIP/2.0 405 Method Not Allowed Via: SIP/2.0/UDP 10.0.2.231:5060;received=59.167.244.232;branch=z9hG4bK483b77e1;rport=1088 From: "asterisk" ;tag=as1121982d To: ;tag=aprqngfrt-2ed8f700000c6 Call-ID: 24f5716a19ee682615492ca4625a01cb@10.0.2.231 CSeq: 102 OPTIONS Allow: INVITE,ACK,BYE,REGISTER,CANCEL,PRACK,INFO,SUBSCRIBE,NOTIFY,UPDATE <-------------> --- (7 headers 0 lines) --- Really destroying SIP dialog '24f5716a19ee682615492ca4625a01cb@10.0.2.231' Method: OPTIONS <--- SIP read from 10.0.2.201:5060 ---> INVITE sip:2@10.0.2.231 SIP/2.0 Via: SIP/2.0/UDP 10.0.2.201:5060;branch=z9hG4bK-1d940943 From: "Jason" ;tag=61e2bb0f5a8e7e1bo0 To: Call-ID: 3e96251f-b51b9b4b@10.0.2.201 CSeq: 101 INVITE Max-Forwards: 70 Contact: "Jason" Expires: 240 User-Agent: Linksys/SPA942-5.1.5 Content-Length: 393 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces Content-Type: application/sdp v=0 o=- 7791456 7791456 IN IP4 10.0.2.201 s=- c=IN IP4 10.0.2.201 t=0 0 m=audio 16394 RTP/AVP 18 0 2 4 8 96 97 98 101 a=rtpmap:18 G729a/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv <-------------> --- (14 headers 18 lines) --- Sending to 10.0.2.201 : 5060 (no NAT) Using INVITE request as basis request - 3e96251f-b51b9b4b@10.0.2.201 <--- Reliably Transmitting (no NAT) to 10.0.2.201:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.0.2.201:5060;branch=z9hG4bK-1d940943;received=10.0.2.201 From: "Jason" ;tag=61e2bb0f5a8e7e1bo0 To: ;tag=as68029e8d Call-ID: 3e96251f-b51b9b4b@10.0.2.201 CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="sip.internode.on.net", nonce="66f141e1" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '3e96251f-b51b9b4b@10.0.2.201' in 32000 ms (Method: INVITE) Found user 'jason' <--- SIP read from 10.0.2.201:5060 ---> ACK sip:2@10.0.2.231 SIP/2.0 Via: SIP/2.0/UDP 10.0.2.201:5060;branch=z9hG4bK-1d940943 From: "Jason" ;tag=61e2bb0f5a8e7e1bo0 To: ;tag=as68029e8d Call-ID: 3e96251f-b51b9b4b@10.0.2.201 CSeq: 101 ACK Max-Forwards: 70 Contact: "Jason" User-Agent: Linksys/SPA942-5.1.5 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 10.0.2.201:5060 ---> INVITE sip:2@10.0.2.231 SIP/2.0 Via: SIP/2.0/UDP 10.0.2.201:5060;branch=z9hG4bK-fcb1c6bb From: "Jason" ;tag=61e2bb0f5a8e7e1bo0 To: Call-ID: 3e96251f-b51b9b4b@10.0.2.201 CSeq: 102 INVITE Max-Forwards: 70 Proxy-Authorization: Digest username="jason",realm="sip.internode.on.net",nonce="66f141e1",uri="sip:2@10.0.2.231",algorithm=MD5,response="7401b0554c9c8132263bfeca8e3a96de" Contact: "Jason" Expires: 240 User-Agent: Linksys/SPA942-5.1.5 Content-Length: 393 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces Content-Type: application/sdp v=0 o=- 7791456 7791456 IN IP4 10.0.2.201 s=- c=IN IP4 10.0.2.201 t=0 0 m=audio 16394 RTP/AVP 18 0 2 4 8 96 97 98 101 a=rtpmap:18 G729a/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv <-------------> --- (15 headers 18 lines) --- Sending to 10.0.2.201 : 5060 (no NAT) Using INVITE request as basis request - 3e96251f-b51b9b4b@10.0.2.201 Found user 'jason' Found RTP audio format 18 Found RTP audio format 0 Found RTP audio format 2 Found RTP audio format 4 Found RTP audio format 8 Found RTP audio format 96 Found RTP audio format 97 Found RTP audio format 98 Found RTP audio format 101 Peer audio RTP is at port 10.0.2.201:16394 Found description format G729a for ID 18 Found description format PCMU for ID 0 Found description format G726-32 for ID 2 Found description format G723 for ID 4 Found description format PCMA for ID 8 Found description format G726-40 for ID 96 Found description format G726-24 for ID 97 Found description format G726-16 for ID 98 Found description format telephone-event for ID 101 Capabilities: us - 0x8 (alaw), peer - audio=0xd0d (g723|ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.0.2.201:16394 Looking for 2 in myphones (domain 10.0.2.231) list_route: hop: <--- Transmitting (no NAT) to 10.0.2.201:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.0.2.201:5060;branch=z9hG4bK-fcb1c6bb;received=10.0.2.201 From: "Jason" ;tag=61e2bb0f5a8e7e1bo0 To: Call-ID: 3e96251f-b51b9b4b@10.0.2.201 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> -- Executing [2@myphones:1] Dial("SIP/jason-081d7cf8", "SIP/nodephone/0871271204") in new stack Audio is at 10.0.2.231 port 15596 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 203.2.134.1:5060: INVITE sip:0871271204@sip.internode.on.net SIP/2.0 Via: SIP/2.0/UDP 10.0.2.231:5060;branch=z9hG4bK2e1a1505;rport From: "Jason" ;tag=as7774e54b To: Contact: Call-ID: 4b0247f9697a5651296659876c7edbae@sip.internode.on.net CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Remote-Party-ID: "Jason" ;privacy=off;screen=no Date: Thu, 01 Feb 2007 03:11:47 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 234 v=0 o=root 6406 6406 IN IP4 10.0.2.231 s=session c=IN IP4 10.0.2.231 t=0 0 m=audio 15596 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called nodephone/0871271204 <--- SIP read from 203.2.134.1:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.0.2.231:5060;received=59.167.244.232;branch=z9hG4bK2e1a1505;rport=1088 From: "Jason" ;tag=as7774e54b To: Call-ID: 4b0247f9697a5651296659876c7edbae@sip.internode.on.net CSeq: 102 INVITE <-------------> --- (6 headers 0 lines) --- <--- SIP read from 203.2.134.1:5060 ---> SIP/2.0 408 Request Timeout Via: SIP/2.0/UDP 10.0.2.231:5060;received=59.167.244.232;branch=z9hG4bK2e1a1505;rport=1088 From: "Jason" ;tag=as7774e54b To: ;tag=aprqngfrt-10mq6t30000c6 Call-ID: 4b0247f9697a5651296659876c7edbae@sip.internode.on.net CSeq: 102 INVITE Content-Length: 0 <-------------> --- (7 headers 0 lines) --- -- Got SIP response 408 "Request Timeout" back from 203.2.134.1 Transmitting (NAT) to 203.2.134.1:5060: ACK sip:0871271204@sip.internode.on.net SIP/2.0 Via: SIP/2.0/UDP 10.0.2.231:5060;branch=z9hG4bK2e1a1505;rport From: "Jason" ;tag=as7774e54b To: ;tag=aprqngfrt-10mq6t30000c6 Contact: Call-ID: 4b0247f9697a5651296659876c7edbae@sip.internode.on.net CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Remote-Party-ID: "Jason" ;privacy=off;screen=no Content-Length: 0 --- == Everyone is busy/congested at this time (1:0/0/1) Really destroying SIP dialog '4b0247f9697a5651296659876c7edbae@sip.internode.on.net' Method: INVITE <--- SIP read from 10.0.2.201:5060 ---> CANCEL sip:2@10.0.2.231 SIP/2.0 Via: SIP/2.0/UDP 10.0.2.201:5060;branch=z9hG4bK-fcb1c6bb From: "Jason" ;tag=61e2bb0f5a8e7e1bo0 To: Call-ID: 3e96251f-b51b9b4b@10.0.2.201 CSeq: 102 CANCEL Max-Forwards: 70 Proxy-Authorization: Digest username="jason",realm="sip.internode.on.net",nonce="66f141e1",uri="sip:2@10.0.2.231",algorithm=MD5,response="ab6cae345fa4ae0667a4542c59d840e3" User-Agent: Linksys/SPA942-5.1.5 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 10.0.2.201 : 5060 (no NAT) <--- Reliably Transmitting (no NAT) to 10.0.2.201:5060 ---> SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 10.0.2.201:5060;branch=z9hG4bK-fcb1c6bb;received=10.0.2.201 From: "Jason" ;tag=61e2bb0f5a8e7e1bo0 To: ;tag=as139af391 Call-ID: 3e96251f-b51b9b4b@10.0.2.201 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> <--- Transmitting (no NAT) to 10.0.2.201:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.2.201:5060;branch=z9hG4bK-fcb1c6bb;received=10.0.2.201 From: "Jason" ;tag=61e2bb0f5a8e7e1bo0 To: ;tag=as139af391 Call-ID: 3e96251f-b51b9b4b@10.0.2.201 CSeq: 102 CANCEL User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> <--- SIP read from 10.0.2.201:5060 ---> ACK sip:2@10.0.2.231 SIP/2.0 Via: SIP/2.0/UDP 10.0.2.201:5060;branch=z9hG4bK-fcb1c6bb From: "Jason" ;tag=61e2bb0f5a8e7e1bo0 To: ;tag=as139af391 Call-ID: 3e96251f-b51b9b4b@10.0.2.201 CSeq: 102 ACK Max-Forwards: 70 Proxy-Authorization: Digest username="jason",realm="sip.internode.on.net",nonce="66f141e1",uri="sip:2@10.0.2.231",algorithm=MD5,response="246adcfd5d54d712e675a63fcad7b061" Contact: "Jason" User-Agent: Linksys/SPA942-5.1.5 Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Really destroying SIP dialog '3e96251f-b51b9b4b@10.0.2.201' Method: ACK