Peer audio RTP is at port 89.35.79.69:30498 Found description format telephone-event for ID 101 Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/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 89.35.79.69:30498 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 89.35.79.69, port 16088 Transmitting (no NAT) to 89.35.79.69:16088: ACK sip:23003@89.35.79.69:16088;rinstance=7ed7b782ad482eab SIP/2.0 Via: SIP/2.0/UDP 89.35.79.5:5060;branch=z9hG4bK3d73da63;rport From: "ORAS" ;tag=as5e1926f2 To: ;tag=1b3d7009 Contact: Call-ID: 693ae16d42b918627a4ad5da07bb9361@89.35.79.5 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- Call on SIP/23003-0823b6c0 left from hold -- SIP/23003-0823b6c0 answered SIP/5060-0822ef78 Audio is at 10.0.0.2 port 11524 Adding codec 0x8 (alaw) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (NAT) to 10.0.0.1:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK-163db-56e2fa4-3108;received=10.0.0.1 From: ORAS;tag=a000001-13c4-163db-56e2fa4-56ab To: ;tag=as6db5b353 Call-ID: 97949eec-a000001-13c4-163db-56e2fa4-1725@10.0.0.1 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 300 v=0 o=root 24756 24756 IN IP4 10.0.0.2 s=session c=IN IP4 10.0.0.2 t=0 0 m=audio 11524 RTP/AVP 8 0 18 97 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> -- Packet2Packet bridging SIP/5060-0822ef78 and SIP/23003-0823b6c0 voip*CLI> <--- SIP read from 10.0.0.1:5060 ---> ACK sip:23003@10.0.0.2 SIP/2.0 From: ORAS;tag=a000001-13c4-163db-56e2fa4-56ab To: ;tag=as6db5b353 Call-ID: 97949eec-a000001-13c4-163db-56e2fa4-1725@10.0.0.1 CSeq: 1 ACK Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK-163e3-56e4e72-a1 Max-Forwards: 70 Contact: Content-Length:0 <-------------> --- (9 headers 0 lines) --- voip*CLI> <--- SIP read from 10.0.0.1:5060 ---> BYE sip:23003@10.0.0.2;TCID-29 SIP/2.0 From: ORAS;tag=a000001-13c4-163db-56e2fa4-56ab To: ;tag=as6db5b353 Call-ID: 97949eec-a000001-13c4-163db-56e2fa4-1725@10.0.0.1 CSeq: 2 BYE Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK-163e3-56e4e7a-3287 Max-Forwards: 70 Content-Length:0 <-------------> --- (8 headers 0 lines) --- Sending to 10.0.0.1 : 5060 (NAT) <--- Transmitting (NAT) to 10.0.0.1:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK-163e3-56e4e7a-3287;received=10.0.0.1 From: ORAS;tag=a000001-13c4-163db-56e2fa4-56ab To: ;tag=as6db5b353 Call-ID: 97949eec-a000001-13c4-163db-56e2fa4-1725@10.0.0.1 CSeq: 2 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> Scheduling destruction of SIP dialog '693ae16d42b918627a4ad5da07bb9361@89.35.79.5' in 32000 ms (Method: INVITE) set_destination: Parsing for address/port to send to set_destination: set destination to 89.35.79.69, port 16088 Reliably Transmitting (no NAT) to 89.35.79.69:16088: BYE sip:23003@89.35.79.69:16088;rinstance=7ed7b782ad482eab SIP/2.0 Via: SIP/2.0/UDP 89.35.79.5:5060;branch=z9hG4bK55851551;rport From: "ORAS" ;tag=as5e1926f2 To: ;tag=1b3d7009 Call-ID: 693ae16d42b918627a4ad5da07bb9361@89.35.79.5 CSeq: 103 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- == Spawn extension (default, 23003, 1) exited non-zero on 'SIP/5060-0822ef78' voip*CLI> <--- SIP read from 89.35.79.69:16088 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 89.35.79.5:5060;branch=z9hG4bK55851551;rport=5060 Contact: To: ;tag=1b3d7009 From: "ORAS";tag=as5e1926f2 Call-ID: 693ae16d42b918627a4ad5da07bb9361@89.35.79.5 CSeq: 103 BYE User-Agent: X-Lite release 1006e stamp 34025 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Really destroying SIP dialog '693ae16d42b918627a4ad5da07bb9361@89.35.79.5' Method: INVITE Really destroying SIP dialog '97949eec-a000001-13c4-163db-56e2fa4-1725@10.0.0.1' Method: BYE voip*CLI> <--- SIP read from 89.35.79.247:5060 ---> REGISTER sip:voip.telefonet.ro SIP/2.0 Via: SIP/2.0/UDP 89.35.79.247:5060;branch=z9hG4bKt8463hHZe Max-Forwards: 70 User-Agent: Wu Chuan From: "23001" ;tag=VWtuxzrHvah1XTdZ To: "23001" Call-ID: XfLu8uaWvRwQUWiM@89.35.79.247 CSeq: 10990 REGISTER Contact: Expires: 60 Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 89.35.79.247 : 5060 (no NAT) <--- Transmitting (no NAT) to 89.35.79.247:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 89.35.79.247:5060;branch=z9hG4bKt8463hHZe;received=89.35.79.247 From: "23001" ;tag=VWtuxzrHvah1XTdZ To: "23001" Call-ID: XfLu8uaWvRwQUWiM@89.35.79.247 CSeq: 10990 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> <--- Transmitting (no NAT) to 89.35.79.247:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 89.35.79.247:5060;branch=z9hG4bKt8463hHZe;received=89.35.79.247 From: "23001" ;tag=VWtuxzrHvah1XTdZ To: "23001" ;tag=as4f469b8f Call-ID: XfLu8uaWvRwQUWiM@89.35.79.247 CSeq: 10990 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="voip.telefonet.ro", nonce="4476e144" Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'XfLu8uaWvRwQUWiM@89.35.79.247' in 32000 ms (Method: REGISTER) voip*CLI> <--- SIP read from 89.35.79.247:5060 ---> REGISTER sip:voip.telefonet.ro SIP/2.0 Via: SIP/2.0/UDP 89.35.79.247:5060;branch=z9hG4bKvTVL4kPSj Max-Forwards: 70 User-Agent: Wu Chuan From: "23001" ;tag=vZUv2gTxnjXNfR5q To: "23001" Call-ID: XfLu8uaWvRwQUWiM@89.35.79.247 CSeq: 10991 REGISTER Contact: Expires: 60 Authorization: Digest username="23001", realm="voip.telefonet.ro", nonce="4476e144", uri="sip:voip.telefonet.ro", response="5f9e1f6376d57706d89409a92d1754fc" Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 89.35.79.247 : 5060 (no NAT) <--- Transmitting (no NAT) to 89.35.79.247:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 89.35.79.247:5060;branch=z9hG4bKvTVL4kPSj;received=89.35.79.247 From: "23001" ;tag=vZUv2gTxnjXNfR5q To: "23001" Call-ID: XfLu8uaWvRwQUWiM@89.35.79.247 CSeq: 10991 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> voip*CLI> <--- Transmitting (no NAT) to 89.35.79.247:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 89.35.79.247:5060;branch=z9hG4bKvTVL4kPSj;received=89.35.79.247 From: "23001" ;tag=vZUv2gTxnjXNfR5q To: "23001" ;tag=as4f469b8f Call-ID: XfLu8uaWvRwQUWiM@89.35.79.247 CSeq: 10991 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 60 Contact: ;expires=60 Date: Tue, 06 Mar 2007 03:05:54 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'XfLu8uaWvRwQUWiM@89.35.79.247' in 32000 ms (Method: REGISTER) voip*CLI>