asterisk = 10.0.0.1 firewall = 10.0.0.2 <-- SIP read from 10.0.0.1:24311: --- (0 headers 0 lines) Nat keepalive --- asterisk*CLI> sip debug peer 358968910308 SIP Debugging Enabled for IP: 10.0.0.1:24311 asterisk*CLI> <-- SIP read from 10.0.0.1:24311: --- (0 headers 0 lines) Nat keepalive --- asterisk*CLI> <-- SIP read from 10.0.0.1:24311: --- (0 headers 0 lines) Nat keepalive --- 11 headers, 0 lines Reliably Transmitting (no NAT) to 10.0.0.1:24311: OPTIONS sip:358968910308@10.0.0.1:24311 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bK04bbca72 From: "asterisk" ;tag=as4903b7e3 To: Contact: Call-ID: 084ae87e0037879369444e5f7b412db1@10.0.0.2 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Thu, 05 Oct 2006 09:35:13 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 --- <-- SIP read from 10.0.0.1:24311: --- (0 headers 0 lines) Nat keepalive --- We're at 10.0.0.2 port 12794 Answering/Requesting with root capability 0x4 (ulaw) Answering with preferred capability 0x8 (alaw) Answering with preferred capability 0x100 (g729) Answering with preferred capability 0x1 (g723) Answering with preferred capability 0x2 (gsm) 12 headers, 12 lines Reliably Transmitting (no NAT) to 10.0.0.1:24311: INVITE sip:358968910308@10.0.0.1:24311 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bK11efe807;rport From: "asreceived" ;tag=as61e25645 To: Contact: Call-ID: 63da4e2d0435d3ab4c1cfe57673b1667@10.0.0.2 CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Thu, 05 Oct 2006 09:35:13 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 255 v=0 o=root 15913 15913 IN IP4 10.0.0.2 s=session c=IN IP4 10.0.0.2 t=0 0 m=audio 12794 RTP/AVP 0 8 18 4 3 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000 a=rtpmap:3 GSM/8000 a=silenceSupp:off - - - - --- Retransmitting #1 (no NAT) to 10.0.0.1:24311: OPTIONS sip:358968910308@10.0.0.1:24311 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bK04bbca72 From: "asterisk" ;tag=as4903b7e3 To: Contact: Call-ID: 084ae87e0037879369444e5f7b412db1@10.0.0.2 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Thu, 05 Oct 2006 09:35:13 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 --- Retransmitting #1 (no NAT) to 10.0.0.1:24311: INVITE sip:358968910308@10.0.0.1:24311 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bK11efe807;rport From: "asreceived" ;tag=as61e25645 To: Contact: Call-ID: 63da4e2d0435d3ab4c1cfe57673b1667@10.0.0.2 CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Thu, 05 Oct 2006 09:35:13 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 255 v=0 o=root 15913 15913 IN IP4 10.0.0.2 s=session c=IN IP4 10.0.0.2 t=0 0 m=audio 12794 RTP/AVP 0 8 18 4 3 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000 a=rtpmap:3 GSM/8000 a=silenceSupp:off - - - - --- Destroying call '084ae87e0037879369444e5f7b412db1@10.0.0.2' Reliably Transmitting (no NAT) to 10.0.0.1:24311: CANCEL sip:358968910308@10.0.0.1:24311 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bK11efe807;rport From: "asreceived" ;tag=as61e25645 To: Contact: Call-ID: 63da4e2d0435d3ab4c1cfe57673b1667@10.0.0.2 CSeq: 102 CANCEL User-Agent: Asterisk PBX Content-Length: 0 --- Scheduling destruction of call '63da4e2d0435d3ab4c1cfe57673b1667@10.0.0.2' in 15000 ms Retransmitting #2 (no NAT) to 10.0.0.1:24311: INVITE sip:358968910308@10.0.0.1:24311 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bK11efe807;rport From: "asreceived" ;tag=as61e25645 To: Contact: Call-ID: 63da4e2d0435d3ab4c1cfe57673b1667@10.0.0.2 CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Thu, 05 Oct 2006 09:35:13 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 255 v=0 o=root 15913 15913 IN IP4 10.0.0.2 s=session c=IN IP4 10.0.0.2 t=0 0 m=audio 12794 RTP/AVP 0 8 18 4 3 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000 a=rtpmap:3 GSM/8000 a=silenceSupp:off - - - - --- We're at 10.0.0.2 port 12934 Answering/Requesting with root capability 0x4 (ulaw) Answering with preferred capability 0x8 (alaw) Answering with preferred capability 0x100 (g729) Answering with preferred capability 0x1 (g723) Answering with preferred capability 0x2 (gsm) 12 headers, 12 lines Reliably Transmitting (no NAT) to 10.0.0.1:24311: INVITE sip:358968910308@10.0.0.1:24311 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bK447719b4;rport From: "asreceived" ;tag=as52a969b3 To: Contact: Call-ID: 2f15a05b0d561e835a7adfc8786fbe54@10.0.0.2 CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Thu, 05 Oct 2006 09:35:33 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 255 v=0 o=root 15913 15913 IN IP4 10.0.0.2 s=session c=IN IP4 10.0.0.2 t=0 0 m=audio 12934 RTP/AVP 0 8 18 4 3 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000 a=rtpmap:3 GSM/8000 a=silenceSupp:off - - - - --- asterisk*CLI> <-- SIP read from 10.0.0.1:24311: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bK447719b4;rport From: "asreceived" ;tag=as52a969b3 To: ;tag=3018104732 Contact: Call-ID: 2f15a05b0d561e835a7adfc8786fbe54@10.0.0.2 CSeq: 102 INVITE Server: X-Lite release 1105x Content-Length: 0 --- (9 headers 0 lines)--- <-- SIP read from 10.0.0.1:24311: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bK447719b4;rport From: "asreceived" ;tag=as52a969b3 To: ;tag=3018104732 Contact: Call-ID: 2f15a05b0d561e835a7adfc8786fbe54@10.0.0.2 CSeq: 102 INVITE Server: X-Lite release 1105x Content-Length: 0 --- (9 headers 0 lines)--- asterisk*CLI> <-- SIP read from 10.0.0.1:24311: --- (0 headers 0 lines) Nat keepalive --- Retransmitting #3 (no NAT) to 10.0.0.1:24311: INVITE sip:358968910308@10.0.0.1:24311 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bK11efe807;rport From: "asreceived" ;tag=as61e25645 To: Contact: Call-ID: 63da4e2d0435d3ab4c1cfe57673b1667@10.0.0.2 CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Thu, 05 Oct 2006 09:35:13 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 255 v=0 o=root 15913 15913 IN IP4 10.0.0.2 s=session c=IN IP4 10.0.0.2 t=0 0 m=audio 12794 RTP/AVP 0 8 18 4 3 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000 a=rtpmap:3 GSM/8000 a=silenceSupp:off - - - - --- Retransmitting #1 (no NAT) to 10.0.0.1:24311: CANCEL sip:358968910308@10.0.0.1:24311 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bK11efe807;rport From: "asreceived" ;tag=as61e25645 To: Contact: Call-ID: 63da4e2d0435d3ab4c1cfe57673b1667@10.0.0.2 CSeq: 102 CANCEL User-Agent: Asterisk PBX Content-Length: 0 --- Retransmitting #4 (no NAT) to 10.0.0.1:24311: INVITE sip:358968910308@10.0.0.1:24311 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bK11efe807;rport From: "asreceived" ;tag=as61e25645 To: Contact: Call-ID: 63da4e2d0435d3ab4c1cfe57673b1667@10.0.0.2 CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Thu, 05 Oct 2006 09:35:13 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 255 v=0 o=root 15913 15913 IN IP4 10.0.0.2 s=session c=IN IP4 10.0.0.2 t=0 0 m=audio 12794 RTP/AVP 0 8 18 4 3 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000 a=rtpmap:3 GSM/8000 a=silenceSupp:off - - - - --- Retransmitting #2 (no NAT) to 10.0.0.1:24311: CANCEL sip:358968910308@10.0.0.1:24311 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bK11efe807;rport From: "asreceived" ;tag=as61e25645 To: Contact: Call-ID: 63da4e2d0435d3ab4c1cfe57673b1667@10.0.0.2 CSeq: 102 CANCEL User-Agent: Asterisk PBX Content-Length: 0 --- asterisk*CLI> <-- SIP read from 10.0.0.1:24311: BYE sip:097537565@10.0.0.2 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.1:24311;rport;branch=z9hG4bK761E4AF4F79F48CA854C3F6E96078EE9 From: ;tag=1990678324 To: "asreceived" ;tag=as61e25645 Contact: Call-ID: 63da4e2d0435d3ab4c1cfe57673b1667@10.0.0.2 CSeq: 30485 BYE Max-Forwards: 70 User-Agent: X-Lite release 1105x Content-Length: 0 --- (10 headers 0 lines)--- Sending to 10.0.0.1 : 24311 (non-NAT) Transmitting (no NAT) to 10.0.0.1:24311: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.1:24311;branch=z9hG4bK761E4AF4F79F48CA854C3F6E96078EE9 From: ;tag=1990678324 To: "asreceived" ;tag=as61e25645 Call-ID: 63da4e2d0435d3ab4c1cfe57673b1667@10.0.0.2 CSeq: 30485 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 --- Retransmitting #5 (no NAT) to 10.0.0.1:24311: INVITE sip:358968910308@10.0.0.1:24311 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bK11efe807;rport From: "asreceived" ;tag=as61e25645 To: Contact: Call-ID: 63da4e2d0435d3ab4c1cfe57673b1667@10.0.0.2 CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Thu, 05 Oct 2006 09:35:13 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 255 v=0 o=root 15913 15913 IN IP4 10.0.0.2 s=session c=IN IP4 10.0.0.2 t=0 0 m=audio 12794 RTP/AVP 0 8 18 4 3 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000 a=rtpmap:3 GSM/8000 a=silenceSupp:off - - - - --- Retransmitting #3 (no NAT) to 10.0.0.1:24311: CANCEL sip:358968910308@10.0.0.1:24311 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bK11efe807;rport From: "asreceived" ;tag=as61e25645 To: Contact: Call-ID: 63da4e2d0435d3ab4c1cfe57673b1667@10.0.0.2 CSeq: 102 CANCEL User-Agent: Asterisk PBX Content-Length: 0 --- Oct 5 12:35:37 WARNING[15913]: chan_sip.c:860 retrans_pkt: Maximum retries exceeded on call 63da4e2d0435d3ab4c1cfe57673b1667@10.0.0.2 for seqno 102 (Critical Request) Retransmitting #4 (no NAT) to 10.0.0.1:24311: CANCEL sip:358968910308@10.0.0.1:24311 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bK11efe807;rport From: "asreceived" ;tag=as61e25645 To: Contact: Call-ID: 63da4e2d0435d3ab4c1cfe57673b1667@10.0.0.2 CSeq: 102 CANCEL User-Agent: Asterisk PBX Content-Length: 0 --- Retransmitting #5 (no NAT) to 10.0.0.1:24311: CANCEL sip:358968910308@10.0.0.1:24311 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bK11efe807;rport From: "asreceived" ;tag=as61e25645 To: Contact: Call-ID: 63da4e2d0435d3ab4c1cfe57673b1667@10.0.0.2 CSeq: 102 CANCEL User-Agent: Asterisk PBX Content-Length: 0 --- Oct 5 12:35:39 WARNING[15913]: chan_sip.c:860 retrans_pkt: Maximum retries exceeded on call 63da4e2d0435d3ab4c1cfe57673b1667@10.0.0.2 for seqno 102 (Non-critical Request) Destroying call '63da4e2d0435d3ab4c1cfe57673b1667@10.0.0.2' asterisk*CLI> <-- SIP read from 10.0.0.1:24311: SIP/2.0 200 Ok Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bK447719b4;rport From: "asreceived" ;tag=as52a969b3 To: ;tag=3018104732 Contact: Call-ID: 2f15a05b0d561e835a7adfc8786fbe54@10.0.0.2 CSeq: 102 INVITE Content-Type: application/sdp Server: X-Lite release 1105x Content-Length: 316 v=0 o=358968910308 269908784 269915575 IN IP4 10.0.0.1 s=X-Lite c=IN IP4 10.0.0.1 t=0 0 m=audio 26432 RTP/AVP 0 8 3 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 a=sendrecv --- (10 headers 14 lines)--- Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 3 Found RTP audio format 98 Found RTP audio format 97 Found RTP audio format 101 Peer audio RTP is at port 10.0.0.1:26432 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 - 0x10f (g723|gsm|ulaw|alaw|g729), peer - audio=0x60e (gsm|ulaw|alaw|speex|ilbc)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 10.0.0.1, port 24311 Transmitting (no NAT) to 10.0.0.1:24311: ACK sip:358968910308@10.0.0.1:24311 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bK5449bb8b;rport From: "asreceived" ;tag=as52a969b3 To: ;tag=3018104732 Contact: Call-ID: 2f15a05b0d561e835a7adfc8786fbe54@10.0.0.2 CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 --- 11 headers, 0 lines Reliably Transmitting (no NAT) to 10.0.0.1:24311: OPTIONS sip:358968910308@10.0.0.1:24311 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bK06992209 From: "asterisk" ;tag=as4949058b To: Contact: Call-ID: 29f507a30c0a46993e7c49b44b0223a0@10.0.0.2 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Thu, 05 Oct 2006 09:35:43 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 --- <-- SIP read from 10.0.0.1:24311: SIP/2.0 200 Ok Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bK06992209 From: "asterisk" ;tag=as4949058b To: ;tag=2987254751 Contact: Call-ID: 29f507a30c0a46993e7c49b44b0223a0@10.0.0.2 Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,NOTIFY CSeq: 102 OPTIONS Server: X-Lite release 1105x Content-Length: 0 --- (10 headers 0 lines)--- Destroying call '29f507a30c0a46993e7c49b44b0223a0@10.0.0.2' asterisk*CLI> <-- SIP read from 10.0.0.1:24311: --- (0 headers 0 lines) Nat keepalive --- Oct 5 12:35:46 WARNING[15913]: chan_sip.c:860 retrans_pkt: Maximum retries exceeded on call 002B-007F-4FD55E00-99@81.29.134.169 for seqno 229155361 (Non-critical Response) asterisk*CLI> <-- SIP read from 10.0.0.1:24311: BYE sip:097537565@10.0.0.2 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.1:24311;rport;branch=z9hG4bKC9BDCBA5F2C64D409CDF1D0E43A91C4F From: ;tag=3018104732 To: "asreceived" ;tag=as52a969b3 Contact: Call-ID: 2f15a05b0d561e835a7adfc8786fbe54@10.0.0.2 CSeq: 21636 BYE Max-Forwards: 70 User-Agent: X-Lite release 1105x Content-Length: 0 --- (10 headers 0 lines)--- Sending to 10.0.0.1 : 24311 (non-NAT) Transmitting (no NAT) to 10.0.0.1:24311: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.1:24311;branch=z9hG4bKC9BDCBA5F2C64D409CDF1D0E43A91C4F From: ;tag=3018104732 To: "asreceived" ;tag=as52a969b3 Call-ID: 2f15a05b0d561e835a7adfc8786fbe54@10.0.0.2 CSeq: 21636 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 --- Destroying call '2f15a05b0d561e835a7adfc8786fbe54@10.0.0.2' asterisk*CLI> <-- SIP read from 10.0.0.1:24311: --- (0 headers 0 lines) Nat keepalive --- asterisk*CLI> sip no debug SIP Debugging Disabled asterisk*CLI> exit [root@asterisk asterisk]#