Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Proxy-Authenticate: Digest realm="asterisk", nonce="46bb78ef" Content-Length: 0 Nov 13 14:38:04 VERBOSE[8868] logger.c: --- (11 headers 0 lines)Nov 13 14:38:04 VERBOSE[8868] logger.c: --- (11 headers 0 lines)--- Nov 13 14:38:04 DEBUG[8868] chan_sip.c: Acked pending invite 102 Nov 13 14:38:04 DEBUG[8868] chan_sip.c: Stopping retransmission on '372385330dcc4f5d6bba8e5b34dd95d9@70.90.149.177' of Request 102: Match Found Nov 13 14:38:04 VERBOSE[8868] logger.c: Transmitting (no NAT) to 208.100.6.10:5060:UDP with fd 12 ACK sip:15742042820@chiv1.voipstreet.com SIP/2.0 Via: SIP/2.0/UDP 70.90.149.177:5060;branch=z9hG4bK0bc67443;rport From: "OnsiteServiceNow" ;tag=as24196c6e To: ;tag=as58470746 Contact: Call-ID: 372385330dcc4f5d6bba8e5b34dd95d9@70.90.149.177 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Nov 13 14:38:04 VERBOSE[8868] logger.c: UDP write: fd 12 Nov 13 14:38:04 VERBOSE[8868] logger.c: We're at 70.90.149.177 port 16668 Nov 13 14:38:04 VERBOSE[8868] logger.c: Adding codec 0x4 (ulaw) to SDP Nov 13 14:38:04 VERBOSE[8868] logger.c: Adding codec 0x8 (alaw) to SDP Nov 13 14:38:04 VERBOSE[8868] logger.c: Adding non-codec 0x1 (telephone-event) to SDP Nov 13 14:38:04 VERBOSE[8868] logger.c: Reliably Transmitting (no NAT) to 208.100.6.10:5060:UDP with fd 12 INVITE sip:15742042820@chiv1.voipstreet.com SIP/2.0 Via: SIP/2.0/UDP 70.90.149.177:5060;branch=z9hG4bK69971141;rport From: "OnsiteServiceNow" ;tag=as24196c6e To: Contact: Call-ID: 372385330dcc4f5d6bba8e5b34dd95d9@70.90.149.177 CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Proxy-Authorization: Digest username="dfrohwerk-17891", realm="asterisk", algorithm=MD5, uri="sip:15742042820@chiv1.voipstreet.com", nonce="46bb78ef", response="282a4f2840c689b13f6a71f5c54a64f5", opaque="" Date: Mon, 13 Nov 2006 20:38:04 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 240 v=0 o=root 8858 8859 IN IP4 70.90.149.177 s=session c=IN IP4 70.90.149.177 t=0 0 m=audio 16668 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Nov 13 14:38:04 VERBOSE[8868] logger.c: UDP write: fd 12 Nov 13 14:38:04 VERBOSE[8868] logger.c: <-- Sip read from 208.100.6.10:5060:UDP SIP/2.0 100 Trying Via: SIP/2.0/UDP 70.90.149.177:5060;branch=z9hG4bK69971141;received=70.90.149.177;rport=5060 From: "OnsiteServiceNow" ;tag=as24196c6e To: Call-ID: 372385330dcc4f5d6bba8e5b34dd95d9@70.90.149.177 CSeq: 103 INVITE User-Agent: Lightwave Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 Nov 13 14:38:04 VERBOSE[8868] logger.c: --- (10 headers 0 lines)Nov 13 14:38:04 VERBOSE[8868] logger.c: --- (10 headers 0 lines)--- Nov 13 14:38:04 DEBUG[8868] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '372385330dcc4f5d6bba8e5b34dd95d9@70.90.149.177' Request 103: Found Nov 13 14:38:04 VERBOSE[8868] logger.c: <-- Sip read from 208.100.6.10:5060:UDP SIP/2.0 200 OK Via: SIP/2.0/UDP 70.90.149.177:5060;branch=z9hG4bK69971141;received=70.90.149.177;rport=5060 From: "OnsiteServiceNow" ;tag=as24196c6e To: ;tag=as33dba6e8 Call-ID: 372385330dcc4f5d6bba8e5b34dd95d9@70.90.149.177 CSeq: 103 INVITE User-Agent: Lightwave Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 214 v=0 o=root 4534 4534 IN IP4 208.100.6.10 s=session c=IN IP4 208.100.6.10 t=0 0 m=audio 14226 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - Nov 13 14:38:04 VERBOSE[8868] logger.c: --- (11 headers 10 lines)Nov 13 14:38:04 VERBOSE[8868] logger.c: --- (11 headers 10 lines)--- Nov 13 14:38:04 DEBUG[8868] chan_sip.c: Acked pending invite 103 Nov 13 14:38:04 DEBUG[8868] chan_sip.c: Stopping retransmission on '372385330dcc4f5d6bba8e5b34dd95d9@70.90.149.177' of Request 103: Match Found Nov 13 14:38:04 VERBOSE[8868] logger.c: Found RTP audio format 0 Nov 13 14:38:04 VERBOSE[8868] logger.c: Found RTP audio format 101 Nov 13 14:38:04 VERBOSE[8868] logger.c: Peer audio RTP is at port 208.100.6.10:14226 Nov 13 14:38:04 DEBUG[8868] chan_sip.c: Peer audio RTP is at port 208.100.6.10:14226 Nov 13 14:38:04 VERBOSE[8868] logger.c: Found description format PCMU Nov 13 14:38:04 VERBOSE[8868] logger.c: Found description format telephone-event Nov 13 14:38:04 VERBOSE[8868] logger.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Nov 13 14:38:04 VERBOSE[8868] logger.c: Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Nov 13 14:38:04 DEBUG[8868] chan_sip.c: build_route: Contact hop: Nov 13 14:38:04 VERBOSE[8868] logger.c: list_route: hop: Nov 13 14:38:04 VERBOSE[8868] logger.c: set_destination: Parsing for address/port/transport to send to Nov 13 14:38:04 WARNING[8868] chan_sip.c: Unsupported transport 'UDP' should be 'UDP', 'TCP', or 'TLS' Nov 13 14:38:04 VERBOSE[8868] logger.c: Transmitting (no NAT) to 208.100.6.10:5060:UDP with fd 12 ACK sip:15742042820@208.100.6.10 SIP/2.0 Via: SIP/2.0/UDP 70.90.149.177:5060;branch=z9hG4bK3f4f13b0;rport From: "OnsiteServiceNow" ;tag=as24196c6e To: ;tag=as33dba6e8 Contact: Call-ID: 372385330dcc4f5d6bba8e5b34dd95d9@70.90.149.177 CSeq: 103 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Nov 13 14:38:04 VERBOSE[8868] logger.c: UDP write: fd 12 Nov 13 14:38:04 VERBOSE[10005] logger.c: -- SIP/voipstreet-082b98f0 answered SIP/201-08294c38 Nov 13 14:38:04 VERBOSE[10005] logger.c: We're at 70.90.149.177 port 10812 Nov 13 14:38:04 VERBOSE[10005] logger.c: Adding codec 0x4 (ulaw) to SDP Nov 13 14:38:04 VERBOSE[10005] logger.c: Adding codec 0x8 (alaw) to SDP Nov 13 14:38:04 VERBOSE[10005] logger.c: Adding non-codec 0x1 (telephone-event) to SDP Nov 13 14:38:04 VERBOSE[10005] logger.c: Reliably Transmitting (no NAT) to 70.90.149.179:5060:UDP with fd 12 SIP/2.0 200 OK Via: SIP/2.0/UDP 70.90.149.179:5060;branch=z9hG4bK-f003f7d8;received=70.90.149.179 From: Onsite Service Now,Inc. ;tag=4a2a0c47cf2fb0aco0 To: ;tag=as60d218e5 Call-ID: 33c0cca3-6328c00@70.90.149.179 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 240 v=0 o=root 8858 8858 IN IP4 70.90.149.177 s=session c=IN IP4 70.90.149.177 t=0 0 m=audio 10812 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Nov 13 14:38:04 VERBOSE[10005] logger.c: UDP write: fd 12 Nov 13 14:38:04 VERBOSE[10005] logger.c: -- Attempting native bridge of SIP/201-08294c38 and SIP/voipstreet-082b98f0 Nov 13 14:38:04 VERBOSE[8868] logger.c: <-- Sip read from 70.90.149.179:5060:UDP ACK sip:15742042820@70.90.149.177;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 70.90.149.179:5060;branch=z9hG4bK-53d7d29d From: Onsite Service Now,Inc. ;tag=4a2a0c47cf2fb0aco0 To: ;tag=as60d218e5 Call-ID: 33c0cca3-6328c00@70.90.149.179 CSeq: 102 ACK Max-Forwards: 70 Proxy-Authorization: Digest username="201",realm="asterisk",nonce="035c93d5",uri="sip:15742042820@70.90.149.177",algorithm=MD5,response="ff81d0af59a1914336f4aca9c774dce9" Contact: Onsite Service Now,Inc. User-Agent: Sipura/SPA1000-2.0.10(e) Content-Length: 0 Nov 13 14:38:04 VERBOSE[8868] logger.c: --- (11 headers 0 lines)Nov 13 14:38:04 VERBOSE[8868] logger.c: --- (11 headers 0 lines)--- Nov 13 14:38:04 DEBUG[8868] chan_sip.c: Stopping retransmission on '33c0cca3-6328c00@70.90.149.179' of Response 102: Match Found Nov 13 14:38:06 VERBOSE[8868] logger.c: <-- Sip read from 70.90.149.179:5060:UDP NOTIFY sip:70.90.149.177 SIP/2.0 Via: SIP/2.0/UDP 70.90.149.179:5060;branch=z9hG4bK-fc91af68 From: Onsite Service Now,Inc. ;tag=9d074f5b40d37cfo0 To: Call-ID: 43d8809b-666bb78f@70.90.149.179 CSeq: 933 NOTIFY Max-Forwards: 70 Event: keep-alive User-Agent: Sipura/SPA1000-2.0.10(e) Content-Length: 0 Nov 13 14:38:06 VERBOSE[8868] logger.c: --- (10 headers 0 lines)Nov 13 14:38:06 VERBOSE[8868] logger.c: --- (10 headers 0 lines)--- Nov 13 14:38:06 VERBOSE[8868] logger.c: Transmitting (no NAT) to 70.90.149.179:5060:UDP with fd 12 SIP/2.0 200 OK Via: SIP/2.0/UDP 70.90.149.179:5060;branch=z9hG4bK-fc91af68;received=70.90.149.179 From: Onsite Service Now,Inc. ;tag=9d074f5b40d37cfo0 To: ;tag=as34aec288 Call-ID: 43d8809b-666bb78f@70.90.149.179 CSeq: 933 NOTIFY User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- Nov 13 14:38:06 VERBOSE[8868] logger.c: UDP write: fd 12 Nov 13 14:38:06 VERBOSE[8868] logger.c: Destroying call '43d8809b-666bb78f@70.90.149.179' [2]+ Stopped tail -f /var/log/asterisk/full