<-- SIP read from 83.149.106.71:5060: --- (0 headers 0 lines) Nat keepalive --- Destroying call '63cf632a543857f04b80430818071bd0@192.168.2.202' borys*CLI> <-- SIP read from 83.149.106.71:5060: --- (0 headers 0 lines) Nat keepalive --- borys*CLI> <-- SIP read from 83.149.106.71:5060: --- (0 headers 0 lines) Nat keepalive --- borys*CLI> <-- SIP read from 83.149.106.71:5060: --- (0 headers 0 lines) Nat keepalive --- We're at 192.168.2.202 port 10002 Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 10 lines Reliably Transmitting (no NAT) to 83.149.106.71:5060: INVITE sip:01236777@sip.provider.com SIP/2.0 Via: SIP/2.0/UDP 192.168.2.202:5060;branch=z9hG4bK777cccff;rport From: "paytest" ;tag=as77bbb80a To: Contact: Call-ID: 5090e816200363b77d6f2b7845e150de@sip.provider.com CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 15 Feb 2006 08:41:41 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 217 v=0 o=root 21128 21128 IN IP4 192.168.2.202 s=session c=IN IP4 192.168.2.202 t=0 0 m=audio 10002 RTP/AVP 3 101 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- borys*CLI> <-- SIP read from 83.149.106.71:5060: SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 192.168.2.202:5060;branch=z9hG4bK777cccff;rport=5060;received=6 2.233.163.154 From: "paytest" ;tag=as77bbb80a To: Call-ID: 5090e816200363b77d6f2b7845e150de@sip.provider.com CSeq: 102 INVITE Content-Length: 0 Warning: 392 sip.provider.com:5060 "Noisy feedback tells: pid=6005 req_src_ip=62 .233.163.154 req_src_port=5060 in_uri=sip:01236777@sip.provider.com out_uri=sip :01236777@83.149.106.7:5060 via_cnt==1" --- (8 headers 0 lines)--- borys*CLI> <-- SIP read from 83.149.106.71:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.2.202:5060;received=62.233.163.154;branch=z9hG4bK777ccc ff From: "paytest" ;tag=as77bbb80a To: ;tag=as159c776b Call-ID: 5090e816200363b77d6f2b7845e150de@sip.provider.com CSeq: 102 INVITE User-Agent: foo Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Proxy-Authenticate: Digest realm="asterisk", nonce="336baa54" Content-Length: 0 --- (11 headers 0 lines)--- Transmitting (no NAT) to 83.149.106.71:5060: ACK sip:01236777@sip.provider.com SIP/2.0 Via: SIP/2.0/UDP 192.168.2.202:5060;branch=z9hG4bK777cccff;rport From: "paytest" ;tag=as77bbb80a To: ;tag=as159c776b Contact: Call-ID: 5090e816200363b77d6f2b7845e150de@sip.provider.com CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- We're at 192.168.2.202 port 10002 Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 83.149.106.71:5060: INVITE sip:01236777@sip.provider.com SIP/2.0 Via: SIP/2.0/UDP 192.168.2.202:5060;branch=z9hG4bK6c5c0b3d;rport From: "paytest" ;tag=as77bbb80a To: Contact: Call-ID: 5090e816200363b77d6f2b7845e150de@sip.provider.com CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Proxy-Authorization: Digest username="88456", realm="asterisk", algorithm=MD5, uri="sip:01236777@sip.provider.com", nonce="336baa54", response="609e715b2341ba 7d52f8bcd0684e8974", opaque="" Date: Wed, 15 Feb 2006 08:41:41 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 217 v=0 o=root 21128 21129 IN IP4 192.168.2.202 s=session c=IN IP4 192.168.2.202 t=0 0 m=audio 10002 RTP/AVP 3 101 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- borys*CLI> <-- SIP read from 83.149.106.71:5060: SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 192.168.2.202:5060;branch=z9hG4bK6c5c0b3d;rport=5060;received=6 2.233.163.154 From: "paytest" ;tag=as77bbb80a To: Call-ID: 5090e816200363b77d6f2b7845e150de@sip.provider.com CSeq: 103 INVITE Content-Length: 0 Warning: 392 sip.provider.com:5060 "Noisy feedback tells: pid=6005 req_src_ip=62 .233.163.154 req_src_port=5060 in_uri=sip:01236777@sip.provider.com out_uri=sip :01236777@83.149.106.7:5060 via_cnt==1" --- (8 headers 0 lines)--- borys*CLI> <-- SIP read from 83.149.106.71:5060: --- (0 headers 0 lines) Nat keepalive --- borys*CLI> <-- SIP read from 83.149.106.71:5060: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.2.202:5060;received=62.233.163.154;branch=z9hG4bK6c5c0b 3d From: "paytest" ;tag=as77bbb80a To: ;tag=as4bb81779 Call-ID: 5090e816200363b77d6f2b7845e150de@sip.provider.com CSeq: 103 INVITE User-Agent: foo Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Type: application/sdp Content-Length: 363 v=0 o=root 15888 15888 IN IP4 83.149.106.7 s=session c=IN IP4 83.149.106.71 t=0 0 m=audio 57210 RTP/AVP 0 8 18 3 110 97 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:3 GSM/8000 a=rtpmap:110 speex/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=nortpproxy:yes --- (11 headers 16 lines)--- Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 3 Found RTP audio format 110 Found RTP audio format 97 Found RTP audio format 101 Peer audio RTP is at port 83.149.106.71:57210 Found description format PCMU Found description format PCMA Found description format G729 Found description format GSM Found description format speex Found description format iLBC Found description format telephone-event Capabilities: us - 0x2 (gsm), peer - audio=0x70e (gsm|ulaw|alaw|g729|speex|ilbc) /video=0x0 (nothing), combined - 0x2 (gsm) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event) , combined - 0x1 (telephone-event) borys*CLI> <-- SIP read from 83.149.106.71:5060: --- (0 headers 0 lines) Nat keepalive --- borys*CLI> <-- SIP read from 83.149.106.71:5060: --- (0 headers 0 lines) Nat keepalive --- borys*CLI> <-- SIP read from 83.149.106.71:5060: --- (0 headers 0 lines) Nat keepalive --- borys*CLI> <-- SIP read from 83.149.106.71:5060: --- (0 headers 0 lines) Nat keepalive --- Feb 15 09:42:24 NOTICE[32042]: chan_sip.c:5238 sip_reregister: -- Re-registration for 88456@sip.provider.com REGISTER 13 headers, 0 lines Reliably Transmitting (no NAT) to 83.149.106.71:5060: REGISTER sip:sip.provider.com SIP/2.0 Via: SIP/2.0/UDP 192.168.2.202:5060;branch=z9hG4bK142da11f;rport From: ;tag=as4244bb45 To: Call-ID: 63cf632a543857f04b80430818071bd0@192.168.2.202 CSeq: 973 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Authorization: Digest username="88456", realm="sip.provider.com", algorithm=MD5, uri="sip:sip.provider.com", nonce="43f2e972f3063c3cca42dee961baf8bc14361638", response="347cfce08282472672bf859d5f4bec26", opaque="" Expires: 120 Contact: Event: registration Content-Length: 0 --- borys*CLI> <-- SIP read from 83.149.106.71:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.202:5060;branch=z9hG4bK142da11f;rport=5060;received=62.233.163.154 From: ;tag=as4244bb45 To: ;tag=2b7d19ef0c39d8851f535a4b5dde05d7.1888 Call-ID: 63cf632a543857f04b80430818071bd0@192.168.2.202 CSeq: 973 REGISTER Contact: ;expires=180 Content-Length: 0 Warning: 392 sip.provider.com:5060 "Noisy feedback tells: pid=6006 req_src_ip=62.233.163.154 req_src_port=5060 in_uri=sip:sip.provider.com out_uri=sip:sip.provider.com via_cnt==1" --- (9 headers 0 lines)--- Scheduling destruction of call '63cf632a543857f04b80430818071bd0@192.168.2.202' in 32000 ms Feb 15 09:42:25 NOTICE[32042]: chan_sip.c:9660 handle_response_register: Outbound Registration: Expiry for sip.provider.com is 120 sec (Scheduling reregistration in 105 s) Reliably Transmitting (no NAT) to 83.149.106.71:5060: CANCEL sip:01236777@sip.provider.com SIP/2.0 Via: SIP/2.0/UDP 192.168.2.202:5060;branch=z9hG4bK6c5c0b3d;rport From: "paytest" ;tag=as77bbb80a To: Contact: Call-ID: 5090e816200363b77d6f2b7845e150de@sip.provider.com CSeq: 103 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Proxy-Authorization: Digest username="88456", realm="asterisk", algorithm=MD5, uri="sip:01236777@sip.provider.com", nonce="336baa54", response="db4fa68667acb7acef1de52beaf37e5c", opaque="" Content-Length: 0 --- Scheduling destruction of call '5090e816200363b77d6f2b7845e150de@sip.provider.com' in 15000 ms borys*CLI> <-- SIP read from 83.149.106.71:5060: SIP/2.0 200 canceling Via: SIP/2.0/UDP 192.168.2.202:5060;branch=z9hG4bK6c5c0b3d;rport=5060;received=62.233.163.154 From: "paytest" ;tag=as77bbb80a To: ;tag=eddeb0674413c672e972bf69acd45b83-dd83 Call-ID: 5090e816200363b77d6f2b7845e150de@sip.provider.com CSeq: 103 CANCEL Content-Length: 0 Warning: 392 sip.provider.com:5060 "Noisy feedback tells: pid=6005 req_src_ip=62.233.163.154 req_src_port=5060 in_uri=sip:01236777@sip.provider.com out_uri=sip:01236777@83.149.106.7:5060 via_cnt==1" --- (8 headers 0 lines)--- borys*CLI> <-- SIP read from 83.149.106.71:5060: SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 192.168.2.202:5060;received=62.233.163.154;branch=z9hG4bK6c5c0b3d From: "paytest" ;tag=as77bbb80a To: ;tag=as4bb81779 Call-ID: 5090e816200363b77d6f2b7845e150de@sip.provider.com CSeq: 103 INVITE User-Agent: foo Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Length: 0 --- (10 headers 0 lines)--- Transmitting (no NAT) to 83.149.106.71:5060: ACK sip:01236777@sip.provider.com SIP/2.0 Via: SIP/2.0/UDP 192.168.2.202:5060;branch=z9hG4bK6c5c0b3d;rport From: "paytest" ;tag=as77bbb80a To: ;tag=as4bb81779 Contact: Call-ID: 5090e816200363b77d6f2b7845e150de@sip.provider.com CSeq: 103 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Destroying call '5090e816200363b77d6f2b7845e150de@sip.provider.com' borys*CLI> <-- SIP read from 83.149.106.71:5060: --- (0 headers 0 lines) Nat keepalive --- borys*CLI>