<------------> -- Executing [0871271201@myphones:1] Dial("SIP/jason-081da620", "SIP/nodephone/0871271201") in new stack Audio is at 59.167.244.232 port 10238 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:0871271201@sip.internode.on.net SIP/2.0 Via: SIP/2.0/UDP 59.167.244.232:5060;branch=z9hG4bK4c7d2a93;rport From: "Jason" ;tag=as07c0d7d8 To: Contact: Call-ID: 4e6ec6e73fa284d5316ad40141d0daef@59.167.244.232 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 02 Feb 2007 04:07:34 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 242 v=0 o=root 3152 3152 IN IP4 59.167.244.232 s=session c=IN IP4 59.167.244.232 t=0 0 m=audio 10238 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/0871271201 <--- SIP read from 203.2.134.1:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 59.167.244.232:5060;branch=z9hG4bK4c7d2a93;rport=5060 From: "Jason" ;tag=as07c0d7d8 To: Call-ID: 4e6ec6e73fa284d5316ad40141d0daef@59.167.244.232 CSeq: 102 INVITE <-------------> --- (6 headers 0 lines) --- Really destroying SIP dialog '31000642-8e5212ca@10.0.2.201' Method: REGISTER <--- SIP read from 203.2.134.1:5060 ---> SIP/2.0 408 Request Timeout Via: SIP/2.0/UDP 59.167.244.232:5060;branch=z9hG4bK4c7d2a93;rport=5060 From: "Jason" ;tag=as07c0d7d8 To: ;tag=aprqngfrt-5vs09500000c6 Call-ID: 4e6ec6e73fa284d5316ad40141d0daef@59.167.244.232 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:0871271201@sip.internode.on.net SIP/2.0 Via: SIP/2.0/UDP 59.167.244.232:5060;branch=z9hG4bK4c7d2a93;rport From: "Jason" ;tag=as07c0d7d8 To: ;tag=aprqngfrt-5vs09500000c6 Contact: Call-ID: 4e6ec6e73fa284d5316ad40141d0daef@59.167.244.232 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- == Everyone is busy/congested at this time (1:0/0/1) Really destroying SIP dialog '4e6ec6e73fa284d5316ad40141d0daef@59.167.244.232' Method: INVITE [Feb 2 15:08:05] WARNING[4424]: pbx.c:2460 __ast_pbx_run: Timeout, but no rule 't' in context 'myphones' Scheduling destruction of SIP dialog 'ea577ab4-312aa56a@10.0.2.201' in 32000 ms (Method: INVITE) <--- Reliably Transmitting (NAT) to 10.0.2.201:5060 ---> SIP/2.0 408 Request Timeout Via: SIP/2.0/UDP 10.0.2.201:5060;branch=z9hG4bK-4181cf0e;received=10.0.2.201 From: "Jason" ;tag=a35f8a2dd5c4b446o0 To: ;tag=as1a91d9f2 Call-ID: ea577ab4-312aa56a@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 <------------> <--- SIP read from 10.0.2.201:5060 ---> ACK sip:0871271201@10.0.2.231 SIP/2.0 Via: SIP/2.0/UDP 10.0.2.201:5060;branch=z9hG4bK-4181cf0e From: "Jason" ;tag=a35f8a2dd5c4b446o0 To: ;tag=as1a91d9f2 Call-ID: ea577ab4-312aa56a@10.0.2.201 CSeq: 102 ACK Max-Forwards: 70 Proxy-Authorization: Digest username="jason",realm="asterisk",nonce="4d650288",uri="sip:0871271201@10.0.2.231",algorithm=MD5,response="f7f88b4f77a8d57796887cff3338b491" Contact: "Jason" User-Agent: Linksys/SPA942-5.1.5 Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Reliably Transmitting (NAT) to 203.2.134.1:5060: OPTIONS sip:sip.internode.on.net SIP/2.0 Via: SIP/2.0/UDP 59.167.244.232:5060;branch=z9hG4bK6b0a9deb;rport From: "asterisk" ;tag=as4b210172 To: Contact: Call-ID: 7badfc7b590b86c2580269f3087c2e55@59.167.244.232 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 02 Feb 2007 04:08:43 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- Reliably Transmitting (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=z9hG4bK4259a824;rport From: "asterisk" ;tag=as7101319b To: Contact: Call-ID: 29078e914e1e20157425ee4371264b39@10.0.2.231 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 02 Feb 2007 04:08:43 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- Really destroying SIP dialog 'ea577ab4-312aa56a@10.0.2.201' Method: ACK <--- SIP read from 10.0.2.201:5060 ---> SIP/2.0 200 OK To: ;tag=3ca707916c62eb32i0 From: "asterisk" ;tag=as7101319b Call-ID: 29078e914e1e20157425ee4371264b39@10.0.2.231 CSeq: 102 OPTIONS Via: SIP/2.0/UDP 10.0.2.231:5060;branch=z9hG4bK4259a824 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 '29078e914e1e20157425ee4371264b39@10.0.2.231' Method: OPTIONS <--- SIP read from 203.2.134.1:5060 ---> SIP/2.0 405 Method Not Allowed Via: SIP/2.0/UDP 59.167.244.232:5060;branch=z9hG4bK6b0a9deb;rport=5060 From: "asterisk" ;tag=as4b210172 To: ;tag=aprqngfrt-o1dtfe10000c6 Call-ID: 7badfc7b590b86c2580269f3087c2e55@59.167.244.232 CSeq: 102 OPTIONS Allow: INVITE,ACK,BYE,REGISTER,CANCEL,PRACK,INFO,SUBSCRIBE,NOTIFY,UPDATE <-------------> --- (7 headers 0 lines) --- Really destroying SIP dialog '7badfc7b590b86c2580269f3087c2e55@59.167.244.232' Method: OPTIONS