Asterisk 1.2.18, Copyright (C) 1999 - 2006 Digium, Inc. and others. Created by Mark Spencer Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'show license' for details. ========================================================================= Connected to Asterisk 1.2.18 currently running on PCF100 (pid = 10744) PCF100*CLI> <-- SIP read from 10.0.2.150:5060: REGISTER sip:altigen@10.0.2.50 SIP/2.0 CSeq: 877 REGISTER To: From: ;tag=203_1181350854 Call-ID: 14795_1181350854@10.0.2.150 Contact: sip:altigen@10.0.2.150 Via: SIP/2.0/UDP 10.0.2.150:5060;rport;branch=z9hG4bK16867_1181350854 Authorization: Digest username="altigen",realm="10.0.2.50",nonce="267e04cb",response="605d8ee94c52ca2cc1fac1daeda4af92",uri="sip:altigen@10.0.2.50",algorithm=MD5 Expires: 300 User-Agent: AltiGen IP Phone 0.1 Content-Length: 0 --- (11 headers 0 lines) --- PCF100*CLI> Using latest REGISTER request as basis request Sending to 10.0.2.150 : 5060 (NAT) PCF100*CLI> Transmitting (no NAT) to 10.0.2.150:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.0.2.150:5060;branch=z9hG4bK16867_1181350854;received=10.0.2.150;rport=5060 From: ;tag=203_1181350854 To: Call-ID: 14795_1181350854@10.0.2.150 CSeq: 877 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- PCF100*CLI> Transmitting (no NAT) to 10.0.2.150:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.0.2.150:5060;branch=z9hG4bK16867_1181350854;received=10.0.2.150;rport=5060 From: ;tag=203_1181350854 To: ;tag=as40e445b3 Call-ID: 14795_1181350854@10.0.2.150 CSeq: 877 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY WWW-Authenticate: Digest algorithm=MD5, realm="10.0.2.50", nonce="30be8d91" Content-Length: 0 --- Scheduling destruction of call '14795_1181350854@10.0.2.150' in 15000 ms PCF100*CLI> <-- SIP read from 10.0.2.150:5060: REGISTER sip:altigen@10.0.2.50 SIP/2.0 CSeq: 878 REGISTER To: From: ;tag=203_1181350854 Call-ID: 14795_1181350854@10.0.2.150 Contact: sip:altigen@10.0.2.150 Via: SIP/2.0/UDP 10.0.2.150:5060;rport;branch=z9hG4bK16868_1181350854 Authorization: Digest username="altigen",realm="10.0.2.50",nonce="30be8d91",response="3f5ab5b9d9616599286e423b112d14d2",uri="sip:altigen@10.0.2.50",algorithm=MD5 Expires: 300 User-Agent: AltiGen IP Phone 0.1 Content-Length: 0 --- (11 headers 0 lines) --- Using latest REGISTER request as basis request PCF100*CLI> Sending to 10.0.2.150 : 5060 (NAT) PCF100*CLI> Transmitting (no NAT) to 10.0.2.150:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.0.2.150:5060;branch=z9hG4bK16868_1181350854;received=10.0.2.150;rport=5060 From: ;tag=203_1181350854 To: Call-ID: 14795_1181350854@10.0.2.150 CSeq: 878 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- PCF100*CLI> Transmitting (no NAT) to 10.0.2.150:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.2.150:5060;branch=z9hG4bK16868_1181350854;received=10.0.2.150;rport=5060 From: ;tag=203_1181350854 To: ;tag=as40e445b3 Call-ID: 14795_1181350854@10.0.2.150 CSeq: 878 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Expires: 300 Contact: ;expires=300 Date: Thu, 14 Jun 2007 17:17:56 GMT Content-Length: 0 --- PCF100*CLI> Scheduling destruction of call '14795_1181350854@10.0.2.150' in 15000 ms PCF100*CLI> Destroying call '14795_1181350854@10.0.2.150' PCF100*CLI> 12 headers, 0 lines PCF100*CLI> Reliably Transmitting (no NAT) to 10.0.2.150:5060: OPTIONS sip:altigen@10.0.2.150 SIP/2.0 Via: SIP/2.0/UDP 10.0.2.50:5060;branch=z9hG4bK3cd36e4b;rport From: "asterisk" ;tag=as6bd2a8af To: Contact: Call-ID: 672373670736032754851c266164e8a6@10.0.2.50 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 14 Jun 2007 17:18:27 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- PCF100*CLI> <-- SIP read from 10.0.2.150:5060: SIP/2.0 200 OK To: From: "asterisk" ;tag=as6bd2a8af CSeq: 102 OPTIONS Call-ID: 672373670736032754851c266164e8a6@10.0.2.50 Content-Length: 0 PCF100*CLI> --- (6 headers 0 lines) --- PCF100*CLI> Destroying call '672373670736032754851c266164e8a6@10.0.2.50' PCF100*CLI> We're at 10.0.2.50 port 11872 PCF100*CLI> Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 10 lines PCF100*CLI> Reliably Transmitting (no NAT) to 10.0.2.150:5060: INVITE sip:6122353016@10.0.2.150 SIP/2.0 Via: SIP/2.0/UDP 10.0.2.50:5060;branch=z9hG4bK11163c53;rport From: "6516460935" ;tag=as4b66de61 To: Contact: Call-ID: 791a836641c3693d68a2ae9262d575d9@10.0.2.50 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 14 Jun 2007 17:18:36 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 210 v=0 o=root 10744 10744 IN IP4 10.0.2.50 s=session c=IN IP4 10.0.2.50 t=0 0 m=audio 11872 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- PCF100*CLI> <-- SIP read from 10.0.2.150:5060: SIP/2.0 100 Trying To: ;tag=205_1181350854 From: "6516460935" ;tag=as4b66de61 CSeq: 102 INVITE Call-ID: 791a836641c3693d68a2ae9262d575d9@10.0.2.50 Via: SIP/2.0/UDP 10.0.2.50:5060;branch=z9hG4bK11163c53;rport User-Agent: AltiGen IP Phone 0.1 Content-Length: 0 PCF100*CLI> --- (8 headers 0 lines) --- PCF100*CLI> <-- SIP read from 10.0.2.150:5060: SIP/2.0 180 Ringing To: ;tag=205_1181350854 From: "6516460935" ;tag=as4b66de61 CSeq: 102 INVITE Call-ID: 791a836641c3693d68a2ae9262d575d9@10.0.2.50 Via: SIP/2.0/UDP 10.0.2.50:5060;branch=z9hG4bK11163c53;rport User-Agent: AltiGen IP Phone 0.1 Content-Length: 0 PCF100*CLI> --- (8 headers 0 lines) --- PCF100*CLI> <-- SIP read from 10.0.2.150:5060: SIP/2.0 200 OK To: ;tag=205_1181350854 From: "6516460935" ;tag=as4b66de61 Contact: sip:altigen@10.0.2.150 CSeq: 102 INVITE Call-ID: 791a836641c3693d68a2ae9262d575d9@10.0.2.50 Via: SIP/2.0/UDP 10.0.2.50:5060;branch=z9hG4bK11163c53;rport User-Agent: AltiGen IP Phone 0.1 ASupportedCodec: 4 18 0 Content-Type: application/sdp Content-Length: 179 v=0 o=MxSIP 0 0 IN IP4 10.0.2.150 s=SIP Call c=IN IP4 10.0.2.150 t=0 0 m=audio 49208 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 --- (11 headers 9 lines) --- PCF100*CLI> Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 10.0.2.150:49208 Found description format PCMU PCF100*CLI> Found description format telephone-event Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) PCF100*CLI> Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) PCF100*CLI> list_route: hop: PCF100*CLI> set_destination: Parsing for address/port to send to PCF100*CLI> set_destination: set destination to 10.0.2.150, port 5060 PCF100*CLI> Transmitting (no NAT) to 10.0.2.150:5060: ACK sip:altigen@10.0.2.150 SIP/2.0 Via: SIP/2.0/UDP 10.0.2.50:5060;branch=z9hG4bK493faed4;rport From: "6516460935" ;tag=as4b66de61 To: ;tag=205_1181350854 Contact: Call-ID: 791a836641c3693d68a2ae9262d575d9@10.0.2.50 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- PCF100*CLI> <-- SIP read from 10.0.2.150:5060: REGISTER sip:altigen@10.0.2.50 SIP/2.0 CSeq: 879 REGISTER To: From: ;tag=203_1181350854 Call-ID: 14795_1181350854@10.0.2.150 Contact: sip:altigen@10.0.2.150 Via: SIP/2.0/UDP 10.0.2.150:5060;rport;branch=z9hG4bK16874_1181350854 Authorization: Digest username="altigen",realm="10.0.2.50",nonce="30be8d91",response="3f5ab5b9d9616599286e423b112d14d2",uri="sip:altigen@10.0.2.50",algorithm=MD5 Expires: 300 User-Agent: AltiGen IP Phone 0.1 Content-Length: 0 --- (11 headers 0 lines) --- PCF100*CLI> Using latest REGISTER request as basis request PCF100*CLI> Sending to 10.0.2.150 : 5060 (NAT) PCF100*CLI> Transmitting (no NAT) to 10.0.2.150:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.0.2.150:5060;branch=z9hG4bK16874_1181350854;received=10.0.2.150;rport=5060 From: ;tag=203_1181350854 To: Call-ID: 14795_1181350854@10.0.2.150 CSeq: 879 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- PCF100*CLI> Transmitting (no NAT) to 10.0.2.150:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.0.2.150:5060;branch=z9hG4bK16874_1181350854;received=10.0.2.150;rport=5060 From: ;tag=203_1181350854 To: ;tag=as1d55d2c4 Call-ID: 14795_1181350854@10.0.2.150 CSeq: 879 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY WWW-Authenticate: Digest algorithm=MD5, realm="10.0.2.50", nonce="652da767" Content-Length: 0 --- PCF100*CLI> Scheduling destruction of call '14795_1181350854@10.0.2.150' in 15000 ms PCF100*CLI> <-- SIP read from 10.0.2.150:5060: REGISTER sip:altigen@10.0.2.50 SIP/2.0 CSeq: 880 REGISTER To: From: ;tag=203_1181350854 Call-ID: 14795_1181350854@10.0.2.150 Contact: sip:altigen@10.0.2.150 Via: SIP/2.0/UDP 10.0.2.150:5060;rport;branch=z9hG4bK16875_1181350854 Authorization: Digest username="altigen",realm="10.0.2.50",nonce="652da767",response="1be076a706aafff505d775c0b35f951e",uri="sip:altigen@10.0.2.50",algorithm=MD5 Expires: 300 User-Agent: AltiGen IP Phone 0.1 Content-Length: 0 --- (11 headers 0 lines) --- PCF100*CLI> Using latest REGISTER request as basis request Sending to 10.0.2.150 : 5060 (NAT) PCF100*CLI> Transmitting (no NAT) to 10.0.2.150:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.0.2.150:5060;branch=z9hG4bK16875_1181350854;received=10.0.2.150;rport=5060 From: ;tag=203_1181350854 To: Call-ID: 14795_1181350854@10.0.2.150 CSeq: 880 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- PCF100*CLI> Transmitting (no NAT) to 10.0.2.150:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.2.150:5060;branch=z9hG4bK16875_1181350854;received=10.0.2.150;rport=5060 From: ;tag=203_1181350854 To: ;tag=as1d55d2c4 Call-ID: 14795_1181350854@10.0.2.150 CSeq: 880 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Expires: 300 Contact: ;expires=300 Date: Thu, 14 Jun 2007 17:18:56 GMT Content-Length: 0 --- PCF100*CLI> Scheduling destruction of call '14795_1181350854@10.0.2.150' in 15000 ms PCF100*CLI> Destroying call '14795_1181350854@10.0.2.150' PCF100*CLI> 12 headers, 0 lines PCF100*CLI> Reliably Transmitting (no NAT) to 10.0.2.150:5060: OPTIONS sip:altigen@10.0.2.150 SIP/2.0 Via: SIP/2.0/UDP 10.0.2.50:5060;branch=z9hG4bK6d9e5228;rport From: "asterisk" ;tag=as483c0b23 To: Contact: Call-ID: 45ba7fcd7781d2242990059626601361@10.0.2.50 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 14 Jun 2007 17:19:27 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- PCF100*CLI> <-- SIP read from 10.0.2.150:5060: SIP/2.0 200 OK To: From: "asterisk" ;tag=as483c0b23 CSeq: 102 OPTIONS Call-ID: 45ba7fcd7781d2242990059626601361@10.0.2.50 Content-Length: 0 PCF100*CLI> --- (6 headers 0 lines) --- PCF100*CLI> Destroying call '45ba7fcd7781d2242990059626601361@10.0.2.50' PCF100*CLI> <-- SIP read from 10.0.2.150:5060: INVITE sip:6516460935@10.0.2.50 SIP/2.0 CSeq: 2 INVITE To: "6516460935" ;tag=as4b66de61 From: ;tag=205_1181350854 Call-ID: 791a836641c3693d68a2ae9262d575d9@10.0.2.50 Via: SIP/2.0/UDP 10.0.2.150:5060;rport;branch=z9hG4bK16876_1181350854 SessionID: DINACallType: User-Agent: AltiGen IP Phone 0.1 Contact: sip:altigen@10.0.2.150 ASupportedCodec: 4 18 0 Content-Type: application/sdp Content-Length: 190 v=0 o=MxSIP 0 1 IN IP4 0.0.0.0 s=SIP Call c=IN IP4 0.0.0.0 t=0 0 m=audio 10062 RTP/AVP 0 101 a=xptime:20 10 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 --- (13 headers 10 lines) --- PCF100*CLI> Using INVITE request as basis request - 791a836641c3693d68a2ae9262d575d9@10.0.2.50 Sending to 10.0.2.150 : 5060 (NAT) Found RTP audio format 0 PCF100*CLI> Found RTP audio format 101 PCF100*CLI> Peer audio RTP is at port 0.0.0.0:10062 Found description format PCMU PCF100*CLI> Found description format telephone-event Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) PCF100*CLI> Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) PCF100*CLI> We're at 10.0.2.50 port 11872 PCF100*CLI> Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP PCF100*CLI> Reliably Transmitting (NAT) to 10.0.2.150:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.2.150:5060;branch=z9hG4bK16876_1181350854;received=10.0.2.150;rport=5060 From: ;tag=205_1181350854 To: "6516460935" ;tag=as4b66de61 Call-ID: 791a836641c3693d68a2ae9262d575d9@10.0.2.50 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 210 v=0 o=root 10744 10745 IN IP4 10.0.2.50 s=session c=IN IP4 10.0.2.50 t=0 0 m=audio 11872 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- PCF100*CLI> <-- SIP read from 10.0.2.150:5060: ACK sip:6516460935@10.0.2.50 SIP/2.0 CSeq: 2 ACK To: "6516460935" ;tag=as4b66de61 From: ;tag=205_1181350854 Call-ID: 791a836641c3693d68a2ae9262d575d9@10.0.2.50 Via: SIP/2.0/UDP 10.0.2.150:5060;rport;branch=z9hG4bK16877_1181350854 User-Agent: AltiGen IP Phone 0.1 Content-Length: 0 --- (8 headers 0 lines) --- PCF100*CLI> <-- SIP read from 10.0.2.150:5060: BYE sip:6516460935@10.0.2.50 SIP/2.0 CSeq: 3 BYE To: "6516460935" ;tag=as4b66de61 From: ;tag=205_1181350854 Call-ID: 791a836641c3693d68a2ae9262d575d9@10.0.2.50 Via: SIP/2.0/UDP 10.0.2.150:5060;rport;branch=z9hG4bK16878_1181350854 Contact: sip:altigen@10.0.2.150 User-Agent: AltiGen IP Phone 0.1 Content-Length: 0 PCF100*CLI> --- (9 headers 0 lines) --- Sending to 10.0.2.150 : 5060 (NAT) PCF100*CLI> Transmitting (NAT) to 10.0.2.150:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.2.150:5060;branch=z9hG4bK16878_1181350854;received=10.0.2.150;rport=5060 From: ;tag=205_1181350854 To: "6516460935" ;tag=as4b66de61 Call-ID: 791a836641c3693d68a2ae9262d575d9@10.0.2.50 CSeq: 3 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing --- PCF100*CLI> Destroying call '791a836641c3693d68a2ae9262d575d9@10.0.2.50' PCF100*CLI> <-- SIP read from 10.0.2.150:5060: REGISTER sip:altigen@10.0.2.50 SIP/2.0 CSeq: 881 REGISTER To: From: ;tag=203_1181350854 Call-ID: 14795_1181350854@10.0.2.150 Contact: sip:altigen@10.0.2.150 Via: SIP/2.0/UDP 10.0.2.150:5060;rport;branch=z9hG4bK16879_1181350854 Authorization: Digest username="altigen",realm="10.0.2.50",nonce="652da767",response="1be076a706aafff505d775c0b35f951e",uri="sip:altigen@10.0.2.50",algorithm=MD5 Expires: 300 User-Agent: AltiGen IP Phone 0.1 Content-Length: 0 --- (11 headers 0 lines) --- PCF100*CLI> Using latest REGISTER request as basis request PCF100*CLI> Sending to 10.0.2.150 : 5060 (NAT) PCF100*CLI> Transmitting (no NAT) to 10.0.2.150:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.0.2.150:5060;branch=z9hG4bK16879_1181350854;received=10.0.2.150;rport=5060 From: ;tag=203_1181350854 To: Call-ID: 14795_1181350854@10.0.2.150 CSeq: 881 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- PCF100*CLI> Transmitting (no NAT) to 10.0.2.150:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.0.2.150:5060;branch=z9hG4bK16879_1181350854;received=10.0.2.150;rport=5060 From: ;tag=203_1181350854 To: ;tag=as5e44436d Call-ID: 14795_1181350854@10.0.2.150 CSeq: 881 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY WWW-Authenticate: Digest algorithm=MD5, realm="10.0.2.50", nonce="2a66dded" Content-Length: 0 --- PCF100*CLI> Scheduling destruction of call '14795_1181350854@10.0.2.150' in 15000 ms PCF100*CLI> <-- SIP read from 10.0.2.150:5060: REGISTER sip:altigen@10.0.2.50 SIP/2.0 CSeq: 882 REGISTER To: From: ;tag=203_1181350854 Call-ID: 14795_1181350854@10.0.2.150 Contact: sip:altigen@10.0.2.150 Via: SIP/2.0/UDP 10.0.2.150:5060;rport;branch=z9hG4bK16880_1181350854 Authorization: Digest username="altigen",realm="10.0.2.50",nonce="2a66dded",response="9fdecff232d5de3bd073dfac10f786b4",uri="sip:altigen@10.0.2.50",algorithm=MD5 Expires: 300 User-Agent: AltiGen IP Phone 0.1 Content-Length: 0 --- (11 headers 0 lines) --- PCF100*CLI> Using latest REGISTER request as basis request Sending to 10.0.2.150 : 5060 (NAT) PCF100*CLI> Transmitting (no NAT) to 10.0.2.150:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.0.2.150:5060;branch=z9hG4bK16880_1181350854;received=10.0.2.150;rport=5060 From: ;tag=203_1181350854 To: Call-ID: 14795_1181350854@10.0.2.150 CSeq: 882 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- PCF100*CLI> Transmitting (no NAT) to 10.0.2.150:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.2.150:5060;branch=z9hG4bK16880_1181350854;received=10.0.2.150;rport=5060 From: ;tag=203_1181350854 To: ;tag=as5e44436d Call-ID: 14795_1181350854@10.0.2.150 CSeq: 882 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Expires: 300 Contact: ;expires=300 Date: Thu, 14 Jun 2007 17:19:56 GMT Content-Length: 0 --- PCF100*CLI> Scheduling destruction of call '14795_1181350854@10.0.2.150' in 15000 ms PCF100*CLI> Destroying call '14795_1181350854@10.0.2.150' PCF100*CLI> quoit