[Sep 29 11:13:22] Connected to Asterisk SVN-trunk-r220792 currently running on asterisk (pid = 31930) Verbosity is at least 50 asterisk*CLI> core set debug 50 Core debug was 0 and is now 50 asterisk*CLI> sip set history on SIP History Recording Enabled (use 'sip show history') asterisk*CLI> sip set debug on SIP Debugging enabled [2009-09-29 11:13:48] <--- SIP read from UDP:192.168.0.125:59332 ---> INVITE sip:8000@asterisk.crosscomm.net SIP/2.0 Via: SIP/2.0/UDP 192.168.0.125:59332;rport;branch=z9hG4bKPjH1V8TB0.1zprkdRHa9KSQYrk3uoQiuWB Max-Forwards: 70 From: "Brendan Martens" ;tag=roCQcGZ.GiIePW50BiCmn-8-vTPxzTSg To: Contact: "Brendan Martens" Call-ID: AHmctdx6jK1geKxAA3vzn0ZmzcuLZ0Ew CSeq: 15246 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, norefersub User-Agent: Telephone 0.14.3 Content-Type: application/sdp Content-Length: 462 v=0 o=- 3463226028 3463226028 IN IP4 192.168.0.125 s=pjmedia c=IN IP4 192.168.0.125 t=0 0 a=X-nat:0 m=audio 4000 RTP/AVP 103 102 104 117 3 0 8 9 101 a=rtcp:4001 IN IP4 192.168.0.125 a=rtpmap:103 speex/16000 a=rtpmap:102 speex/8000 a=rtpmap:104 speex/32000 a=rtpmap:117 iLBC/8000 a=fmtp:117 mode=30 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=sendrecv a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> [2009-09-29 11:13:48] --- (13 headers 20 lines) --- [2009-09-29 11:13:48] Sending to 192.168.0.125 : 59332 (no NAT) [2009-09-29 11:13:48] Using INVITE request as basis request - AHmctdx6jK1geKxAA3vzn0ZmzcuLZ0Ew [2009-09-29 11:13:48] Found peer 'brendanmartens' for 'brendanmartens' from 192.168.0.125:59332 [2009-09-29 11:13:48] <--- Reliably Transmitting (no NAT) to 192.168.0.125:59332 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.0.125:59332;rport;branch=z9hG4bKPjH1V8TB0.1zprkdRHa9KSQYrk3uoQiuWB;received=192.168.0.125 From: "Brendan Martens" ;tag=roCQcGZ.GiIePW50BiCmn-8-vTPxzTSg To: ;tag=as61e66dd2 Call-ID: AHmctdx6jK1geKxAA3vzn0ZmzcuLZ0Ew CSeq: 15246 INVITE Server: Asterisk PBX SVN-trunk-r220792 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk.crosscomm.net", nonce="2dec0fe8" Content-Length: 0 <------------> [2009-09-29 11:13:48] Scheduling destruction of SIP dialog 'AHmctdx6jK1geKxAA3vzn0ZmzcuLZ0Ew' in 6400 ms (Method: INVITE) [2009-09-29 11:13:48] <--- SIP read from UDP:192.168.0.125:59332 ---> ACK sip:8000@asterisk.crosscomm.net SIP/2.0 Via: SIP/2.0/UDP 192.168.0.125:59332;rport;branch=z9hG4bKPjH1V8TB0.1zprkdRHa9KSQYrk3uoQiuWB Max-Forwards: 70 From: "Brendan Martens" ;tag=roCQcGZ.GiIePW50BiCmn-8-vTPxzTSg To: ;tag=as61e66dd2 Call-ID: AHmctdx6jK1geKxAA3vzn0ZmzcuLZ0Ew CSeq: 15246 ACK Content-Length: 0 <-------------> [2009-09-29 11:13:48] --- (8 headers 0 lines) --- [2009-09-29 11:13:48] <--- SIP read from UDP:192.168.0.125:59332 ---> INVITE sip:8000@asterisk.crosscomm.net SIP/2.0 Via: SIP/2.0/UDP 192.168.0.125:59332;rport;branch=z9hG4bKPjStMHlA0EwKi1HGWhCy98HAo1--w4Y6dV Max-Forwards: 70 From: "Brendan Martens" ;tag=roCQcGZ.GiIePW50BiCmn-8-vTPxzTSg To: Contact: "Brendan Martens" Call-ID: AHmctdx6jK1geKxAA3vzn0ZmzcuLZ0Ew CSeq: 15247 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, norefersub User-Agent: Telephone 0.14.3 Authorization: Digest username="brendanmartens", realm="asterisk.crosscomm.net", nonce="2dec0fe8", uri="sip:8000@asterisk.crosscomm.net", response="53cd0ff74f7d990b192a3b01fb092e13", algorithm=MD5 Content-Type: application/sdp Content-Length: 462 v=0 o=- 3463226028 3463226028 IN IP4 192.168.0.125 s=pjmedia c=IN IP4 192.168.0.125 t=0 0 a=X-nat:0 m=audio 4000 RTP/AVP 103 102 104 117 3 0 8 9 101 a=rtcp:4001 IN IP4 192.168.0.125 a=rtpmap:103 speex/16000 a=rtpmap:102 speex/8000 a=rtpmap:104 speex/32000 a=rtpmap:117 iLBC/8000 a=fmtp:117 mode=30 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=sendrecv a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> [2009-09-29 11:13:48] --- (14 headers 20 lines) --- [2009-09-29 11:13:48] Sending to 192.168.0.125 : 59332 (no NAT) [2009-09-29 11:13:48] Using INVITE request as basis request - AHmctdx6jK1geKxAA3vzn0ZmzcuLZ0Ew [2009-09-29 11:13:48] Found peer 'brendanmartens' for 'brendanmartens' from 192.168.0.125:59332 [2009-09-29 11:13:48] Found RTP audio format 103 [2009-09-29 11:13:48] Found RTP audio format 102 [2009-09-29 11:13:48] Found RTP audio format 104 [2009-09-29 11:13:48] Found RTP audio format 117 [2009-09-29 11:13:48] Found RTP audio format 3 [2009-09-29 11:13:48] Found RTP audio format 0 [2009-09-29 11:13:48] Found RTP audio format 8 [2009-09-29 11:13:48] Found RTP audio format 9 [2009-09-29 11:13:48] Found RTP audio format 101 [2009-09-29 11:13:48] Peer audio RTP is at port 192.168.0.125:4000 [2009-09-29 11:13:48] Got unsupported a:rtcp in SDP offer [2009-09-29 11:13:48] Found audio description format speex for ID 103 [2009-09-29 11:13:48] Found audio description format speex for ID 102 [2009-09-29 11:13:48] Found audio description format speex for ID 104 [2009-09-29 11:13:48] Found audio description format iLBC for ID 117 [2009-09-29 11:13:48] Found audio description format GSM for ID 3 [2009-09-29 11:13:48] Found audio description format PCMU for ID 0 [2009-09-29 11:13:48] Found audio description format PCMA for ID 8 [2009-09-29 11:13:48] Found audio description format G722 for ID 9 [2009-09-29 11:13:48] Found audio description format telephone-event for ID 101 [2009-09-29 11:13:48] Capabilities: us - 0x1fdf (g723|gsm|ulaw|alaw|g726|slin|lpc10|g729|speex|ilbc|g726aal2|g722), peer - audio=0x50160e (gsm|ulaw|alaw|speex|ilbc|g722|h263p|mpeg4)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x160e (gsm|ulaw|alaw|speex|ilbc|g722) [2009-09-29 11:13:48] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [2009-09-29 11:13:48] Peer audio RTP is at port 192.168.0.125:4000 [2009-09-29 11:13:48] Looking for 8000 in softphones (domain asterisk.crosscomm.net) [2009-09-29 11:13:48] list_route: hop: [2009-09-29 11:13:48] <--- Transmitting (no NAT) to 192.168.0.125:59332 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.125:59332;rport;branch=z9hG4bKPjStMHlA0EwKi1HGWhCy98HAo1--w4Y6dV;received=192.168.0.125 From: "Brendan Martens" ;tag=roCQcGZ.GiIePW50BiCmn-8-vTPxzTSg To: Call-ID: AHmctdx6jK1geKxAA3vzn0ZmzcuLZ0Ew CSeq: 15247 INVITE Server: Asterisk PBX SVN-trunk-r220792 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Length: 0 <------------> [2009-09-29 11:13:48] -- Executing [8000@softphones:1] ConfBridge("SIP/brendanmartens-1036d3c8", "8000") in new stack [2009-09-29 11:13:48] -- Playing 'conf-onlyperson.slin' (language 'en') [2009-09-29 11:13:52] Reliably Transmitting (no NAT) to 192.168.0.30:5060: OPTIONS sip:line3@192.168.0.30:5060 SIP/2.0 Via: SIP/2.0/UDP 64.105.202.244:5060;branch=z9hG4bK045f763f Max-Forwards: 70 From: "asterisk" ;tag=as718a246b To: Contact: Call-ID: 6d67c23d29e4af1336172325746e6497@64.105.202.244 CSeq: 102 OPTIONS User-Agent: Asterisk PBX SVN-trunk-r220792 Date: Tue, 29 Sep 2009 15:13:52 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- [2009-09-29 11:13:52] <--- SIP read from UDP:192.168.0.30:5060 ---> SIP/2.0 486 Busy Here To: ;tag=f76be8be46253182i0 From: "asterisk" ;tag=as718a246b Call-ID: 6d67c23d29e4af1336172325746e6497@64.105.202.244 CSeq: 102 OPTIONS Via: SIP/2.0/UDP 64.105.202.244:5060;branch=z9hG4bK045f763f Server: Linksys/PAP2T-5.1.6(LS) Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura, replaces <-------------> [2009-09-29 11:13:52] --- (10 headers 0 lines) --- [2009-09-29 11:13:52] Really destroying SIP dialog '6d67c23d29e4af1336172325746e6497@64.105.202.244' Method: OPTIONS [2009-09-29 11:13:52] Reliably Transmitting (no NAT) to 192.168.0.125:59332: OPTIONS sip:brendanmartens@192.168.0.125:59332 SIP/2.0 Via: SIP/2.0/UDP 64.105.202.244:5060;branch=z9hG4bK48bb247f Max-Forwards: 70 From: "asterisk" ;tag=as298b9103 To: Contact: Call-ID: 6ba5164c51cdc8cd2d1996a97cb91159@64.105.202.244 CSeq: 102 OPTIONS User-Agent: Asterisk PBX SVN-trunk-r220792 Date: Tue, 29 Sep 2009 15:13:52 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- [2009-09-29 11:13:52] <--- SIP read from UDP:192.168.0.125:59332 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 64.105.202.244:5060;received=64.105.202.244;branch=z9hG4bK48bb247f Call-ID: 6ba5164c51cdc8cd2d1996a97cb91159@64.105.202.244 From: "asterisk" ;tag=as298b9103 To: CSeq: 102 OPTIONS Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Accept: application/sdp, application/pidf+xml, application/xpidf+xml, message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain Supported: replaces, 100rel, norefersub Allow-Events: presence, refer User-Agent: Telephone 0.14.3 Content-Type: application/sdp Content-Length: 451 v=0 o=- 3463226032 3463226032 IN IP4 192.168.0.125 s=pjmedia c=IN IP4 192.168.0.125 t=0 0 m=audio 4000 RTP/AVP 103 102 104 117 3 0 8 9 101 a=rtcp:4001 IN IP4 192.168.0.125 a=rtpmap:103 speex/16000 a=rtpmap:102 speex/8000 a=rtpmap:104 speex/32000 a=rtpmap:117 iLBC/8000 a=fmtp:117 mode=30 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=sendrecv a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> [2009-09-29 11:13:52] --- (13 headers 19 lines) --- [2009-09-29 11:13:52] Really destroying SIP dialog '6ba5164c51cdc8cd2d1996a97cb91159@64.105.202.244' Method: OPTIONS [2009-09-29 11:13:52] Reliably Transmitting (no NAT) to 192.168.0.30:5061: OPTIONS sip:line4@192.168.0.30:5061 SIP/2.0 Via: SIP/2.0/UDP 64.105.202.244:5060;branch=z9hG4bK49ff7c80 Max-Forwards: 70 From: "asterisk" ;tag=as71d6dd20 To: Contact: Call-ID: 3b307ce14abd58f01250b4dc2655b083@64.105.202.244 CSeq: 102 OPTIONS User-Agent: Asterisk PBX SVN-trunk-r220792 Date: Tue, 29 Sep 2009 15:13:52 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- [2009-09-29 11:13:52] <--- SIP read from UDP:192.168.0.30:5061 ---> SIP/2.0 200 OK To: ;tag=3de4cede920481e2i1 From: "asterisk" ;tag=as71d6dd20 Call-ID: 3b307ce14abd58f01250b4dc2655b083@64.105.202.244 CSeq: 102 OPTIONS Via: SIP/2.0/UDP 64.105.202.244:5060;branch=z9hG4bK49ff7c80 Server: Linksys/PAP2T-5.1.6(LS) Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura, replaces <-------------> [2009-09-29 11:13:52] --- (10 headers 0 lines) --- [2009-09-29 11:13:52] Really destroying SIP dialog '3b307ce14abd58f01250b4dc2655b083@64.105.202.244' Method: OPTIONS [2009-09-29 11:13:53] <--- SIP read from UDP:192.168.0.125:59332 ---> <-------------> [2009-09-29 11:13:54] <--- SIP read from UDP:192.168.0.30:5060 ---> SUBSCRIBE sip:asterisk.crosscomm.net SIP/2.0 Via: SIP/2.0/UDP 192.168.0.30:5060;branch=z9hG4bK-dd1b6ffc From: CrossComm, Inc. ;tag=67edad8c7c7cf558 To: CrossComm, Inc. ;tag=as35894806 Call-ID: d9d6289c-20703588@192.168.0.30 CSeq: 58669 SUBSCRIBE Max-Forwards: 70 Authorization: Digest username="line3",realm="asterisk.crosscomm.net",nonce="6741a5ed",uri="sip:asterisk.crosscomm.net",algorithm=MD5,response="0a9b70dd5e7b6415aaa1b1eaccca053d" Contact: CrossComm, Inc. Accept: application/simple-message-summary Expires: 2147483647 Event: message-summary User-Agent: Linksys/PAP2T-5.1.6(LS) Content-Length: 0 <-------------> [2009-09-29 11:13:54] --- (14 headers 0 lines) --- [2009-09-29 11:13:54] Received resubscription for a dialog we no longer know about. Telling remote side to subscribe again. [2009-09-29 11:13:54] <--- Transmitting (no NAT) to 192.168.0.30:5060 ---> SIP/2.0 481 Subscription does not exist Via: SIP/2.0/UDP 192.168.0.30:5060;branch=z9hG4bK-dd1b6ffc;received=192.168.0.30 From: CrossComm, Inc. ;tag=67edad8c7c7cf558 To: CrossComm, Inc. ;tag=as35894806 Call-ID: d9d6289c-20703588@192.168.0.30 CSeq: 58669 SUBSCRIBE Server: Asterisk PBX SVN-trunk-r220792 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> [2009-09-29 11:13:54] Really destroying SIP dialog 'd9d6289c-20703588@192.168.0.30' Method: SUBSCRIBE [2009-09-29 11:13:55] <--- SIP read from UDP:192.168.0.30:5060 ---> INVITE sip:8000@asterisk.crosscomm.net SIP/2.0 Via: SIP/2.0/UDP 192.168.0.30:5060;branch=z9hG4bK-cb7d6ddc From: CrossComm, Inc. ;tag=53f0e2f24529cf6co0 To: Call-ID: e56c95ea-c7ccf8d4@192.168.0.30 CSeq: 101 INVITE Max-Forwards: 70 Contact: CrossComm, Inc. Expires: 240 User-Agent: Linksys/PAP2T-5.1.6(LS) Content-Length: 397 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura, replaces Content-Type: application/sdp v=0 o=- 8024887 8024887 IN IP4 192.168.0.30 s=- c=IN IP4 192.168.0.30 t=0 0 m=audio 16408 RTP/AVP 0 2 4 8 18 96 97 98 101 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729a/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv <-------------> [2009-09-29 11:13:55] --- (14 headers 18 lines) --- [2009-09-29 11:13:55] Sending to 192.168.0.30 : 5060 (no NAT) [2009-09-29 11:13:55] Using INVITE request as basis request - e56c95ea-c7ccf8d4@192.168.0.30 [2009-09-29 11:13:55] Found peer 'line3' for 'line3' from 192.168.0.30:5060 [2009-09-29 11:13:55] <--- Reliably Transmitting (no NAT) to 192.168.0.30:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.0.30:5060;branch=z9hG4bK-cb7d6ddc;received=192.168.0.30 From: CrossComm, Inc. ;tag=53f0e2f24529cf6co0 To: ;tag=as44c6cd3b Call-ID: e56c95ea-c7ccf8d4@192.168.0.30 CSeq: 101 INVITE Server: Asterisk PBX SVN-trunk-r220792 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk.crosscomm.net", nonce="35fe3581" Content-Length: 0 <------------> [2009-09-29 11:13:55] Scheduling destruction of SIP dialog 'e56c95ea-c7ccf8d4@192.168.0.30' in 6400 ms (Method: INVITE) [2009-09-29 11:13:55] <--- SIP read from UDP:192.168.0.30:5060 ---> ACK sip:8000@asterisk.crosscomm.net SIP/2.0 Via: SIP/2.0/UDP 192.168.0.30:5060;branch=z9hG4bK-cb7d6ddc From: CrossComm, Inc. ;tag=53f0e2f24529cf6co0 To: ;tag=as44c6cd3b Call-ID: e56c95ea-c7ccf8d4@192.168.0.30 CSeq: 101 ACK Max-Forwards: 70 Contact: CrossComm, Inc. User-Agent: Linksys/PAP2T-5.1.6(LS) Content-Length: 0 <-------------> [2009-09-29 11:13:55] --- (10 headers 0 lines) --- [2009-09-29 11:13:55] <--- SIP read from UDP:192.168.0.30:5060 ---> INVITE sip:8000@asterisk.crosscomm.net SIP/2.0 Via: SIP/2.0/UDP 192.168.0.30:5060;branch=z9hG4bK-d20e5990 From: CrossComm, Inc. ;tag=53f0e2f24529cf6co0 To: Call-ID: e56c95ea-c7ccf8d4@192.168.0.30 CSeq: 102 INVITE Max-Forwards: 70 Authorization: Digest username="line3",realm="asterisk.crosscomm.net",nonce="35fe3581",uri="sip:8000@asterisk.crosscomm.net",algorithm=MD5,response="bdc3be6404cdb427d634cbe3b91b4fc1" Contact: CrossComm, Inc. Expires: 240 User-Agent: Linksys/PAP2T-5.1.6(LS) Content-Length: 397 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura, replaces Content-Type: application/sdp v=0 o=- 8024887 8024887 IN IP4 192.168.0.30 s=- c=IN IP4 192.168.0.30 t=0 0 m=audio 16408 RTP/AVP 0 2 4 8 18 96 97 98 101 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729a/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv <-------------> [2009-09-29 11:13:55] --- (15 headers 18 lines) --- [2009-09-29 11:13:55] Sending to 192.168.0.30 : 5060 (no NAT) [2009-09-29 11:13:55] Using INVITE request as basis request - e56c95ea-c7ccf8d4@192.168.0.30 [2009-09-29 11:13:55] Found peer 'line3' for 'line3' from 192.168.0.30:5060 [2009-09-29 11:13:55] Found RTP audio format 0 [2009-09-29 11:13:55] Found RTP audio format 2 [2009-09-29 11:13:55] Found RTP audio format 4 [2009-09-29 11:13:55] Found RTP audio format 8 [2009-09-29 11:13:55] Found RTP audio format 18 [2009-09-29 11:13:55] Found RTP audio format 96 [2009-09-29 11:13:55] Found RTP audio format 97 [2009-09-29 11:13:55] Found RTP audio format 98 [2009-09-29 11:13:55] Found RTP audio format 101 [2009-09-29 11:13:55] Peer audio RTP is at port 192.168.0.30:16408 [2009-09-29 11:13:55] Found audio description format PCMU for ID 0 [2009-09-29 11:13:55] Found audio description format G726-32 for ID 2 [2009-09-29 11:13:55] Found audio description format G723 for ID 4 [2009-09-29 11:13:55] Found audio description format PCMA for ID 8 [2009-09-29 11:13:55] Found audio description format G729a for ID 18 [2009-09-29 11:13:55] Found audio description format G726-40 for ID 96 [2009-09-29 11:13:55] Found audio description format G726-24 for ID 97 [2009-09-29 11:13:55] Found audio description format G726-16 for ID 98 [2009-09-29 11:13:55] Found audio description format telephone-event for ID 101 [2009-09-29 11:13:55] Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x100d0d (g723|ulaw|alaw|g726|g729|ilbc|h263p)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) [2009-09-29 11:13:55] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [2009-09-29 11:13:55] Peer audio RTP is at port 192.168.0.30:16408 [2009-09-29 11:13:55] Looking for 8000 in softphones (domain asterisk.crosscomm.net) [2009-09-29 11:13:55] list_route: hop: [2009-09-29 11:13:55] <--- Transmitting (no NAT) to 192.168.0.30:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.30:5060;branch=z9hG4bK-d20e5990;received=192.168.0.30 From: CrossComm, Inc. ;tag=53f0e2f24529cf6co0 To: Call-ID: e56c95ea-c7ccf8d4@192.168.0.30 CSeq: 102 INVITE Server: Asterisk PBX SVN-trunk-r220792 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Length: 0 <------------> [2009-09-29 11:13:55] -- Executing [8000@softphones:1] ConfBridge("SIP/line3-10369180", "8000") in new stack [2009-09-29 11:14:18] <--- SIP read from UDP:192.168.0.125:59332 ---> REGISTER sip:asterisk.crosscomm.net SIP/2.0 Via: SIP/2.0/UDP 192.168.0.125:59332;rport;branch=z9hG4bKPjK-d5y9P88Ulb3xPfGL9qekWJ8PQzZMRQ Max-Forwards: 70 From: "Brendan Martens" ;tag=VDY5B3zj2u5H8JgqO7Me4OVux-o39Tgv To: "Brendan Martens" Call-ID: WDlyfnhHlUj0.T1GJPGkdSbu4IJFoEhQ CSeq: 23991 REGISTER User-Agent: Telephone 0.14.3 Contact: "Brendan Martens" Expires: 300 Content-Length: 0 <-------------> [2009-09-29 11:14:18] --- (11 headers 0 lines) --- [2009-09-29 11:14:18] Sending to 192.168.0.125 : 59332 (no NAT) [2009-09-29 11:14:18] <--- Transmitting (no NAT) to 192.168.0.125:59332 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.0.125:59332;rport;branch=z9hG4bKPjK-d5y9P88Ulb3xPfGL9qekWJ8PQzZMRQ;received=192.168.0.125 From: "Brendan Martens" ;tag=VDY5B3zj2u5H8JgqO7Me4OVux-o39Tgv To: "Brendan Martens" ;tag=as6ab32e42 Call-ID: WDlyfnhHlUj0.T1GJPGkdSbu4IJFoEhQ CSeq: 23991 REGISTER Server: Asterisk PBX SVN-trunk-r220792 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk.crosscomm.net", nonce="0b6a114e" Content-Length: 0 <------------> [2009-09-29 11:14:18] Scheduling destruction of SIP dialog 'WDlyfnhHlUj0.T1GJPGkdSbu4IJFoEhQ' in 32000 ms (Method: REGISTER) [2009-09-29 11:14:18] <--- SIP read from UDP:192.168.0.125:59332 ---> REGISTER sip:asterisk.crosscomm.net SIP/2.0 Via: SIP/2.0/UDP 192.168.0.125:59332;rport;branch=z9hG4bKPjBZZ48aOdgkZ57bwAUDZEgAze7e6a8zs1 Max-Forwards: 70 From: "Brendan Martens" ;tag=VDY5B3zj2u5H8JgqO7Me4OVux-o39Tgv To: "Brendan Martens" Call-ID: WDlyfnhHlUj0.T1GJPGkdSbu4IJFoEhQ CSeq: 23992 REGISTER User-Agent: Telephone 0.14.3 Contact: "Brendan Martens" Expires: 300 Authorization: Digest username="brendanmartens", realm="asterisk.crosscomm.net", nonce="0b6a114e", uri="sip:asterisk.crosscomm.net", response="58446ccd0de0780f86111880733baa34", algorithm=MD5 Content-Length: 0 <-------------> [2009-09-29 11:14:18] --- (12 headers 0 lines) --- [2009-09-29 11:14:18] Sending to 192.168.0.125 : 59332 (no NAT) [2009-09-29 11:14:18] Reliably Transmitting (no NAT) to 192.168.0.125:59332: OPTIONS sip:brendanmartens@192.168.0.125:59332 SIP/2.0 Via: SIP/2.0/UDP 64.105.202.244:5060;branch=z9hG4bK33c2d553 Max-Forwards: 70 From: "asterisk" ;tag=as08dae874 To: Contact: Call-ID: 6f186b862bc96d1922aa2c462af726b9@64.105.202.244 CSeq: 102 OPTIONS User-Agent: Asterisk PBX SVN-trunk-r220792 Date: Tue, 29 Sep 2009 15:14:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- [2009-09-29 11:14:18] > Saved useragent "Telephone 0.14.3" for peer brendanmartens [2009-09-29 11:14:18] <--- Transmitting (no NAT) to 192.168.0.125:59332 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.125:59332;rport;branch=z9hG4bKPjBZZ48aOdgkZ57bwAUDZEgAze7e6a8zs1;received=192.168.0.125 From: "Brendan Martens" ;tag=VDY5B3zj2u5H8JgqO7Me4OVux-o39Tgv To: "Brendan Martens" ;tag=as6ab32e42 Call-ID: WDlyfnhHlUj0.T1GJPGkdSbu4IJFoEhQ CSeq: 23992 REGISTER Server: Asterisk PBX SVN-trunk-r220792 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Expires: 300 Contact: sip:brendanmartens@192.168.0.125:59332;expires=300 Date: Tue, 29 Sep 2009 15:14:18 GMT Content-Length: 0 <------------> [2009-09-29 11:14:18] Scheduling destruction of SIP dialog 'WDlyfnhHlUj0.T1GJPGkdSbu4IJFoEhQ' in 32000 ms (Method: REGISTER) [2009-09-29 11:14:18] <--- SIP read from UDP:192.168.0.125:59332 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 64.105.202.244:5060;received=64.105.202.244;branch=z9hG4bK33c2d553 Call-ID: 6f186b862bc96d1922aa2c462af726b9@64.105.202.244 From: "asterisk" ;tag=as08dae874 To: CSeq: 102 OPTIONS Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Accept: application/sdp, application/pidf+xml, application/xpidf+xml, message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain Supported: replaces, 100rel, norefersub Allow-Events: presence, refer User-Agent: Telephone 0.14.3 Content-Type: application/sdp Content-Length: 451 v=0 o=- 3463226058 3463226058 IN IP4 192.168.0.125 s=pjmedia c=IN IP4 192.168.0.125 t=0 0 m=audio 4000 RTP/AVP 103 102 104 117 3 0 8 9 101 a=rtcp:4001 IN IP4 192.168.0.125 a=rtpmap:103 speex/16000 a=rtpmap:102 speex/8000 a=rtpmap:104 speex/32000 a=rtpmap:117 iLBC/8000 a=fmtp:117 mode=30 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=sendrecv a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> [2009-09-29 11:14:18] --- (13 headers 19 lines) --- [2009-09-29 11:14:18] Really destroying SIP dialog '6f186b862bc96d1922aa2c462af726b9@64.105.202.244' Method: OPTIONS asterisk*CLI> [2009-09-29 11:14:24] <--- SIP read from UDP:192.168.0.30:5060 ---> SUBSCRIBE sip:asterisk.crosscomm.net SIP/2.0 Via: SIP/2.0/UDP 192.168.0.30:5060;branch=z9hG4bK-132f4941 From: CrossComm, Inc. ;tag=b8a16595a1a5009c To: CrossComm, Inc. ;tag=as35894806 Call-ID: a1aa8fc9-9f8bbf8@192.168.0.30 CSeq: 42206 SUBSCRIBE Max-Forwards: 70 Authorization: Digest username="line3",realm="asterisk.crosscomm.net",nonce="6741a5ed",uri="sip:asterisk.crosscomm.net",algorithm=MD5,response="0a9b70dd5e7b6415aaa1b1eaccca053d" Contact: CrossComm, Inc. Accept: application/simple-message-summary Expires: 2147483647 Event: message-summary User-Agent: Linksys/PAP2T-5.1.6(LS) Content-Length: 0 <-------------> [2009-09-29 11:14:24] --- (14 headers 0 lines) --- [2009-09-29 11:14:24] Received resubscription for a dialog we no longer know about. Telling remote side to subscribe again. [2009-09-29 11:14:24] <--- Transmitting (no NAT) to 192.168.0.30:5060 ---> SIP/2.0 481 Subscription does not exist Via: SIP/2.0/UDP 192.168.0.30:5060;branch=z9hG4bK-132f4941;received=192.168.0.30 From: CrossComm, Inc. ;tag=b8a16595a1a5009c To: CrossComm, Inc. ;tag=as35894806 Call-ID: a1aa8fc9-9f8bbf8@192.168.0.30 CSeq: 42206 SUBSCRIBE Server: Asterisk PBX SVN-trunk-r220792 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> [2009-09-29 11:14:24] Really destroying SIP dialog 'a1aa8fc9-9f8bbf8@192.168.0.30' Method: SUBSCRIBE [2009-09-29 11:14:25] <--- SIP read from UDP:192.168.0.30:5060 ---> CANCEL sip:8000@asterisk.crosscomm.net SIP/2.0 Via: SIP/2.0/UDP 192.168.0.30:5060;branch=z9hG4bK-d20e5990 From: CrossComm, Inc. ;tag=53f0e2f24529cf6co0 To: Call-ID: e56c95ea-c7ccf8d4@192.168.0.30 CSeq: 102 CANCEL Max-Forwards: 70 Authorization: Digest username="line3",realm="asterisk.crosscomm.net",nonce="35fe3581",uri="sip:8000@asterisk.crosscomm.net",algorithm=MD5,response="75639b0562b8b4cfe93d3f23d564e349" User-Agent: Linksys/PAP2T-5.1.6(LS) Content-Length: 0 <-------------> [2009-09-29 11:14:25] --- (10 headers 0 lines) --- [2009-09-29 11:14:25] Sending to 192.168.0.30 : 5060 (no NAT) [2009-09-29 11:14:25] <--- Reliably Transmitting (no NAT) to 192.168.0.30:5060 ---> SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 192.168.0.30:5060;branch=z9hG4bK-d20e5990;received=192.168.0.30 From: CrossComm, Inc. ;tag=53f0e2f24529cf6co0 To: ;tag=as4808d33d Call-ID: e56c95ea-c7ccf8d4@192.168.0.30 CSeq: 102 INVITE Server: Asterisk PBX SVN-trunk-r220792 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> [2009-09-29 11:14:25] <--- Transmitting (no NAT) to 192.168.0.30:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.30:5060;branch=z9hG4bK-d20e5990;received=192.168.0.30 From: CrossComm, Inc. ;tag=53f0e2f24529cf6co0 To: ;tag=as4808d33d Call-ID: e56c95ea-c7ccf8d4@192.168.0.30 CSeq: 102 CANCEL Server: Asterisk PBX SVN-trunk-r220792 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> [2009-09-29 11:14:25] <--- SIP read from UDP:192.168.0.30:5060 ---> ACK sip:8000@asterisk.crosscomm.net SIP/2.0 Via: SIP/2.0/UDP 192.168.0.30:5060;branch=z9hG4bK-d20e5990 From: CrossComm, Inc. ;tag=53f0e2f24529cf6co0 To: ;tag=as4808d33d Call-ID: e56c95ea-c7ccf8d4@192.168.0.30 CSeq: 102 ACK Max-Forwards: 70 Authorization: Digest username="line3",realm="asterisk.crosscomm.net",nonce="35fe3581",uri="sip:8000@asterisk.crosscomm.net",algorithm=MD5,response="bdc3be6404cdb427d634cbe3b91b4fc1" Contact: CrossComm, Inc. User-Agent: Linksys/PAP2T-5.1.6(LS) Content-Length: 0 <-------------> [2009-09-29 11:14:25] --- (11 headers 0 lines) --- [2009-09-29 11:14:25] Really destroying SIP dialog 'e56c95ea-c7ccf8d4@192.168.0.30' Method: ACK [2009-09-29 11:14:26] <--- SIP read from UDP:192.168.0.125:59332 ---> CANCEL sip:8000@asterisk.crosscomm.net SIP/2.0 Via: SIP/2.0/UDP 192.168.0.125:59332;rport;branch=z9hG4bKPjStMHlA0EwKi1HGWhCy98HAo1--w4Y6dV Max-Forwards: 70 From: "Brendan Martens" ;tag=roCQcGZ.GiIePW50BiCmn-8-vTPxzTSg To: Call-ID: AHmctdx6jK1geKxAA3vzn0ZmzcuLZ0Ew CSeq: 15247 CANCEL User-Agent: Telephone 0.14.3 Content-Length: 0 <-------------> [2009-09-29 11:14:26] --- (9 headers 0 lines) --- [2009-09-29 11:14:26] Sending to 192.168.0.125 : 59332 (no NAT) [2009-09-29 11:14:26] <--- Reliably Transmitting (no NAT) to 192.168.0.125:59332 ---> SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 192.168.0.125:59332;rport;branch=z9hG4bKPjStMHlA0EwKi1HGWhCy98HAo1--w4Y6dV;received=192.168.0.125 From: "Brendan Martens" ;tag=roCQcGZ.GiIePW50BiCmn-8-vTPxzTSg To: ;tag=as625702e7 Call-ID: AHmctdx6jK1geKxAA3vzn0ZmzcuLZ0Ew CSeq: 15247 INVITE Server: Asterisk PBX SVN-trunk-r220792 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> [2009-09-29 11:14:26] <--- Transmitting (no NAT) to 192.168.0.125:59332 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.125:59332;rport;branch=z9hG4bKPjStMHlA0EwKi1HGWhCy98HAo1--w4Y6dV;received=192.168.0.125 From: "Brendan Martens" ;tag=roCQcGZ.GiIePW50BiCmn-8-vTPxzTSg To: ;tag=as625702e7 Call-ID: AHmctdx6jK1geKxAA3vzn0ZmzcuLZ0Ew CSeq: 15247 CANCEL Server: Asterisk PBX SVN-trunk-r220792 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> [2009-09-29 11:14:26] <--- SIP read from UDP:192.168.0.125:59332 ---> ACK sip:8000@asterisk.crosscomm.net SIP/2.0 Via: SIP/2.0/UDP 192.168.0.125:59332;rport;branch=z9hG4bKPjStMHlA0EwKi1HGWhCy98HAo1--w4Y6dV Max-Forwards: 70 From: "Brendan Martens" ;tag=roCQcGZ.GiIePW50BiCmn-8-vTPxzTSg To: ;tag=as625702e7 Call-ID: AHmctdx6jK1geKxAA3vzn0ZmzcuLZ0Ew CSeq: 15247 ACK Content-Length: 0 <-------------> [2009-09-29 11:14:26] --- (8 headers 0 lines) --- [2009-09-29 11:14:26] Really destroying SIP dialog 'AHmctdx6jK1geKxAA3vzn0ZmzcuLZ0Ew' Method: ACK asterisk*CLI>