*CLI> sip debug peer 2302 SIP Debugging Enabled for IP: 192.168.100.128:5060 *CLI> 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:192.168.100.128 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.200:5060;branch=z9hG4bK38fd206f From: "asterisk" ;tag=as38a0e281 To: Contact: Call-ID: 1a98ba375a68da2464bfed5c6023fee9@192.168.100.200 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Wed, 29 Sep 2004 01:27:39 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 192.168.100.128:5060 Sip read: SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.100.200:5060;branch=z9hG4bK38fd206f From: "asterisk" ;tag=as38a0e281 To: ;tag=792515087 Contact: Call-ID: 1a98ba375a68da2464bfed5c6023fee9@192.168.100.200 Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,NOTIFY CSeq: 102 OPTIONS Server: X-Lite release 1103a Content-Length: 0 10 headers, 0 lines Destroying call '1a98ba375a68da2464bfed5c6023fee9@192.168.100.200' Sip read: INVITE sip:750@192.168.100.200 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.7:5060;rport;branch=z9hG4bKB8BE83275ADD46FFA68A90578F1BF208 From: 2302 ;tag=2229194259 To: Contact: Call-ID: E446D22C-95A3-44AA-BDEC-10EC1728A8A2@192.168.2.7 CSeq: 7354 INVITE Max-Forwards: 70 Content-Type: application/sdp User-Agent: X-Lite release 1103a Content-Length: 293 v=0 o=2302 125444890 125445461 IN IP4 192.168.2.7 s=X-Lite c=IN IP4 192.168.2.7 t=0 0 m=audio 8000 RTP/AVP 3 0 8 98 97 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:3 gsm/8000 a=rtpmap:98 iLBC/8000 a=rtpmap:97 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 11 headers, 13 lines Using latest request as basis request Reliably Transmitting (NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.2.7:5060;branch=z9hG4bKB8BE83275ADD46FFA68A90578F1BF208;received=192.168.100.128;rport=5060 From: 2302 ;tag=2229194259 To: ;tag=as27059f53 Call-ID: E446D22C-95A3-44AA-BDEC-10EC1728A8A2@192.168.2.7 CSeq: 7354 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Proxy-Authenticate: Digest realm="asterisk", nonce="2bd187ac" Content-Length: 0 to 192.168.100.128:5060 Scheduling destruction of call 'E446D22C-95A3-44AA-BDEC-10EC1728A8A2@192.168.2.7' in 15000 ms Found user '2302' Sip read: ACK sip:750@192.168.100.200 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.7:5060;rport;branch=z9hG4bKB8BE83275ADD46FFA68A90578F1BF208 From: 2302 ;tag=2229194259 To: ;tag=as27059f53 Contact: Call-ID: E446D22C-95A3-44AA-BDEC-10EC1728A8A2@192.168.2.7 CSeq: 7354 ACK Max-Forwards: 70 Content-Length: 0 9 headers, 0 lines Sip read: INVITE sip:750@192.168.100.200 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.7:5060;rport;branch=z9hG4bK6118392B6CC044E58DE245F5E1E88C2D From: 2302 ;tag=2229194259 To: Contact: Call-ID: E446D22C-95A3-44AA-BDEC-10EC1728A8A2@192.168.2.7 CSeq: 7355 INVITE Proxy-Authorization: Digest username="2302",realm="asterisk",nonce="2bd187ac",response="9e88a2655de56a96d69e857db1c83712",uri="sip:750@192.168.100.200" Max-Forwards: 70 Content-Type: application/sdp User-Agent: X-Lite release 1103a Content-Length: 293 v=0 o=2302 125444890 125445461 IN IP4 192.168.2.7 s=X-Lite c=IN IP4 192.168.2.7 t=0 0 m=audio 8000 RTP/AVP 3 0 8 98 97 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:3 gsm/8000 a=rtpmap:98 iLBC/8000 a=rtpmap:97 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 12 headers, 13 lines Using latest request as basis request Sending to 192.168.2.7 : 5060 (NAT) Found RTP audio format 3 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 98 Found RTP audio format 97 Found RTP audio format 101 Peer audio RTP is at port 192.168.2.7:8000 Found description format pcmu Found description format pcma Found description format gsm Found description format iLBC Found description format speex Found description format telephone-event Capabilities: us - 0x6(GSM|ULAW), peer - audio=0x60e(GSM|ULAW|ALAW|SPEEX|ILBC)/video=0x0(EMPTY), combined - 0x6(GSM|ULAW) Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723) Found user '2302' Looking for 750 in from-sip list_route: hop: Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.2.7:5060;branch=z9hG4bK6118392B6CC044E58DE245F5E1E88C2D;received=192.168.100.128;rport=5060 From: 2302 ;tag=2229194259 To: ;tag=as7fe173ae Call-ID: E446D22C-95A3-44AA-BDEC-10EC1728A8A2@192.168.2.7 CSeq: 7355 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 192.168.100.128:5060 We're at 192.168.100.200 port 14388 Answering with preferred capability 0x2(GSM) Answering with preferred capability 0x4(ULAW) Answering with non-codec capability 0x1(G723) Reliably Transmitting (NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.7:5060;branch=z9hG4bK6118392B6CC044E58DE245F5E1E88C2D;received=192.168.100.128;rport=5060 From: 2302 ;tag=2229194259 To: ;tag=as7fe173ae Call-ID: E446D22C-95A3-44AA-BDEC-10EC1728A8A2@192.168.2.7 CSeq: 7355 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 245 v=0 o=root 27383 27383 IN IP4 192.168.100.200 s=session c=IN IP4 192.168.100.200 t=0 0 m=audio 14388 RTP/AVP 3 0 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - to 192.168.100.128:5060 Sip read: ACK sip:750@192.168.100.200 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.7:5060;rport;branch=z9hG4bK527A00098AE64F0AA0980DB27C2535E7 From: 2302 ;tag=2229194259 To: ;tag=as7fe173ae Contact: Call-ID: E446D22C-95A3-44AA-BDEC-10EC1728A8A2@192.168.2.7 CSeq: 7355 ACK Max-Forwards: 70 Content-Length: 0 9 headers, 0 lines Sep 28 21:27:56 NOTICE[213005]: rtp.c:489 ast_rtp_read: Unknown RTP codec 72 received Sep 28 21:28:01 NOTICE[213005]: rtp.c:489 ast_rtp_read: Unknown RTP codec 72 received Sip read: BYE sip:750@192.168.100.200 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.7:5060;rport;branch=z9hG4bK6A6F58B7ECFF47EEBD3B7D5953D6F22C From: 2302 ;tag=2229194259 To: ;tag=as7fe173ae Contact: Call-ID: E446D22C-95A3-44AA-BDEC-10EC1728A8A2@192.168.2.7 CSeq: 7356 BYE Max-Forwards: 70 User-Agent: X-Lite release 1103a Content-Length: 0 10 headers, 0 lines Sending to 192.168.2.7 : 5060 (NAT) Transmitting (NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.7:5060;branch=z9hG4bK6A6F58B7ECFF47EEBD3B7D5953D6F22C;received=192.168.100.128;rport=5060 From: 2302 ;tag=2229194259 To: ;tag=as7fe173ae Call-ID: E446D22C-95A3-44AA-BDEC-10EC1728A8A2@192.168.2.7 CSeq: 7356 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 192.168.100.128:5060 Sip read: 0 headers, 0 lines Destroying call 'E446D22C-95A3-44AA-BDEC-10EC1728A8A2@192.168.2.7'