<------------> -- Executing [0748881658@numberplan-custom-1:1] Dial("SIP/23003-08272f80", "SIP/0748881658@10.0.0.1||T") in new stack Audio is at 10.0.0.2 port 10682 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.0.0.1:5060: INVITE sip:0748881658@10.0.0.1 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bK5e95a363;rport From: "Znuff" ;tag=as3aad0344 To: Contact: Call-ID: 2927ec7533ce342326e5fd1d0f61a758@10.0.0.2 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 05 Mar 2007 04:59:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 236 v=0 o=root 8856 8856 IN IP4 89.35.79.69 s=session c=IN IP4 89.35.79.69 t=0 0 m=audio 38282 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 0748881658@10.0.0.1 voip*CLI> <--- SIP read from 10.0.0.1:5060 ---> SIP/2.0 100 Trying From: "Znuff";tag=as3aad0344 To: ;tag=a000001-13c4-2cf7-afc142-6fb5 Call-ID: 2927ec7533ce342326e5fd1d0f61a758@10.0.0.2 CSeq: 102 INVITE Via: SIP/2.0/UDP 10.0.0.2:5060;received=10.0.0.2;branch=z9hG4bK5e95a363;rport Contact: Content-Length:0 <-------------> --- (8 headers 0 lines) --- voip*CLI> <--- SIP read from 10.0.0.1:5060 ---> SIP/2.0 415 Unsupported Media Type From: "Znuff";tag=as3aad0344 To: ;tag=a000001-13c4-2cf7-afc142-6fb5 Call-ID: 2927ec7533ce342326e5fd1d0f61a758@10.0.0.2 CSeq: 102 INVITE Via: SIP/2.0/UDP 10.0.0.2:5060;received=10.0.0.2;branch=z9hG4bK5e95a363;rport Content-Length:0 <-------------> --- (7 headers 0 lines) --- -- Got SIP response 415 "Unsupported Media Type" back from 10.0.0.1 Transmitting (no NAT) to 10.0.0.1:5060: ACK sip:0748881658@10.0.0.1 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bK5e95a363;rport From: "Znuff" ;tag=as3aad0344 To: ;tag=a000001-13c4-2cf7-afc142-6fb5 Contact: Call-ID: 2927ec7533ce342326e5fd1d0f61a758@10.0.0.2 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- == Everyone is busy/congested at this time (1:0/0/1) -- Executing [0748881658@numberplan-custom-1:2] Congestion("SIP/23003-08272f80", "") in new stack voip*CLI> <--- Transmitting (no NAT) to 89.35.79.69:61286 ---> SIP/2.0 503 Service Unavailable Via: SIP/2.0/UDP 89.35.79.69:61286;branch=z9hG4bK-d87543-1e3d85354a73334a-1--d87543-;received=89.35.79.69;rport=61286 From: "Znuff";tag=b0366644 To: "0748881658";tag=as66bec888 Call-ID: MGI4ZTcxYmE3NTAwZmM0NGI1NTg5MDIwNzAwYzA5YTU. CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 X-Asterisk-HangupCause: Interworking, unspecified X-Asterisk-HangupCauseCode: 127 <------------> == Spawn extension (numberplan-custom-1, 0748881658, 2) exited non-zero on 'SIP/23003-08272f80' voip*CLI> <--- SIP read from 89.35.79.69:61286 ---> ACK sip:0748881658@voip.telefonet.ro SIP/2.0 Via: SIP/2.0/UDP 89.35.79.69:61286;branch=z9hG4bK-d87543-1e3d85354a73334a-1--d87543-;rport To: "0748881658";tag=as66bec888 From: "Znuff";tag=b0366644 Call-ID: MGI4ZTcxYmE3NTAwZmM0NGI1NTg5MDIwNzAwYzA5YTU. CSeq: 2 ACK Content-Length: 0 <-------------> --- (7 headers 0 lines) --- Really destroying SIP dialog '2927ec7533ce342326e5fd1d0f61a758@10.0.0.2' Method: INVITE Really destroying SIP dialog 'MGI4ZTcxYmE3NTAwZmM0NGI1NTg5MDIwNzAwYzA5YTU.' Method: ACK Really destroying SIP dialog 'XfLu8uaWvRwQUWiM@89.35.79.247' Method: REGISTER Really destroying SIP dialog 'owiz95Hz9XJxvP8b@89.35.79.94' Method: REGISTER voip*CLI> <--- SIP read from 89.35.79.69:61286 ---> <------------->