Mar 19 13:29:39 VERBOSE[26940]: Asterisk Ready. Asterisk Event Logger restarted Mar 19 13:29:42 DEBUG[26940]: Auto destroying call '4f88506313bfe2013f8aa86d4e45f519@10.1.1.48' Mar 19 13:29:42 VERBOSE[26940]: Destroying call '4f88506313bfe2013f8aa86d4e45f519@10.1.1.48' Mar 19 13:29:42 DEBUG[26940]: Auto destroying call '6aad86830dc374e16b1b6de119e0e010@10.1.1.48' Mar 19 13:29:42 VERBOSE[26940]: Destroying call '6aad86830dc374e16b1b6de119e0e010@10.1.1.48' Mar 19 13:29:42 DEBUG[26940]: Auto destroying call '1eccc4e46068a9a70e0a27d6762e2cba@10.1.1.48' Mar 19 13:29:42 VERBOSE[26940]: Destroying call '1eccc4e46068a9a70e0a27d6762e2cba@10.1.1.48' Mar 19 13:29:45 VERBOSE[26940]: Sip read: INVITE sip:3000@10.1.1.48 SIP/2.0 Via: SIP/2.0/UDP 10.1.1.100:5060;rport;branch=z9hG4bK4D3D778B3A1542D795A50D6F24E40D90 From: Wilson Chan ;tag=972238047 To: Contact: Call-ID: 92882539-1348-4622-8EEE-1B51026734BE@10.1.1.100 CSeq: 41336 INVITE Max-Forwards: 70 Content-Type: application/sdp User-Agent: X-Lite release 1103m Content-Length: 293 v=0 o=wilson 182967293 182967293 IN IP4 10.1.1.100 s=X-Lite c=IN IP4 10.1.1.100 t=0 0 m=audio 8000 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 Mar 19 13:29:45 VERBOSE[26940]: 11 headers, 13 lines Mar 19 13:29:45 DEBUG[26940]: Allocating new SIP call for 92882539-1348-4622-8EEE-1B51026734BE@10.1.1.100 Mar 19 13:29:45 VERBOSE[26940]: Using latest request as basis request Mar 19 13:29:45 VERBOSE[26940]: Sending to 10.1.1.100 : 5060 (non-NAT) Mar 19 13:29:45 DEBUG[26940]: Setting NAT on RTP to 0 Mar 19 13:29:45 VERBOSE[26940]: Reliably Transmitting (no NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.1.1.100:5060;branch=z9hG4bK4D3D778B3A1542D795A50D6F24E40D90 From: Wilson Chan ;tag=972238047 To: ;tag=as3e441ed8 Call-ID: 92882539-1348-4622-8EEE-1B51026734BE@10.1.1.100 CSeq: 41336 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Proxy-Authenticate: Digest realm="asterisk", nonce="4d496f3c" Content-Length: 0 to 10.1.1.100:5060 Mar 19 13:29:45 VERBOSE[26940]: Scheduling destruction of call '92882539-1348-4622-8EEE-1B51026734BE@10.1.1.100' in 15000 ms Mar 19 13:29:45 VERBOSE[26940]: Found user 'wilson' Mar 19 13:29:45 VERBOSE[26940]: Sip read: ACK sip:3000@10.1.1.48 SIP/2.0 Via: SIP/2.0/UDP 10.1.1.100:5060;rport;branch=z9hG4bK4D3D778B3A1542D795A50D6F24E40D90 From: Wilson Chan ;tag=972238047 To: ;tag=as3e441ed8 Contact: Call-ID: 92882539-1348-4622-8EEE-1B51026734BE@10.1.1.100 CSeq: 41336 ACK Max-Forwards: 70 Content-Length: 0 Mar 19 13:29:45 VERBOSE[26940]: 9 headers, 0 lines Mar 19 13:29:45 DEBUG[26940]: Stopping retransmission on '92882539-1348-4622-8EEE-1B51026734BE@10.1.1.100' of Response 41336: Found Mar 19 13:29:45 VERBOSE[26940]: Sip read: INVITE sip:3000@10.1.1.48 SIP/2.0 Via: SIP/2.0/UDP 10.1.1.100:5060;rport;branch=z9hG4bK5F9CD27E2048483F9B7A62D198E42420 From: Wilson Chan ;tag=972238047 To: Contact: Call-ID: 92882539-1348-4622-8EEE-1B51026734BE@10.1.1.100 CSeq: 41337 INVITE Proxy-Authorization: Digest username="wilson",realm="asterisk",nonce="4d496f3c",response="97459b277da29c443a70c3d10b98b160",uri="sip:3000@10.1.1.48" Max-Forwards: 70 Content-Type: application/sdp User-Agent: X-Lite release 1103m Content-Length: 293 v=0 o=wilson 182967293 182967293 IN IP4 10.1.1.100 s=X-Lite c=IN IP4 10.1.1.100 t=0 0 m=audio 8000 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 Mar 19 13:29:45 VERBOSE[26940]: 12 headers, 13 lines Mar 19 13:29:45 VERBOSE[26940]: Using latest request as basis request Mar 19 13:29:45 VERBOSE[26940]: Sending to 10.1.1.100 : 5060 (non-NAT) Mar 19 13:29:45 DEBUG[26940]: Setting NAT on RTP to 0 Mar 19 13:29:45 VERBOSE[26940]: Found user 'wilson' Mar 19 13:29:45 VERBOSE[26940]: Found RTP audio format 0 Mar 19 13:29:45 VERBOSE[26940]: Found RTP audio format 8 Mar 19 13:29:45 VERBOSE[26940]: Found RTP audio format 3 Mar 19 13:29:45 VERBOSE[26940]: Found RTP audio format 98 Mar 19 13:29:45 VERBOSE[26940]: Found RTP audio format 97 Mar 19 13:29:45 VERBOSE[26940]: Found RTP audio format 101 Mar 19 13:29:45 VERBOSE[26940]: Peer audio RTP is at port 10.1.1.100:8000 Mar 19 13:29:45 DEBUG[26940]: Peer audio RTP is at port 10.1.1.100:8000 Mar 19 13:29:45 VERBOSE[26940]: Found description format pcmu Mar 19 13:29:45 VERBOSE[26940]: Found description format pcma Mar 19 13:29:45 VERBOSE[26940]: Found description format gsm Mar 19 13:29:45 VERBOSE[26940]: Found description format iLBC Mar 19 13:29:45 VERBOSE[26940]: Found description format speex Mar 19 13:29:45 VERBOSE[26940]: Found description format telephone-event Mar 19 13:29:45 VERBOSE[26940]: Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x60e (gsm|ulaw|alaw|speex|ilbc)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) Mar 19 13:29:45 VERBOSE[26940]: Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) Mar 19 13:29:45 DEBUG[26940]: Check for res for wilson Mar 19 13:29:45 VERBOSE[26940]: Looking for 3000 in from-sip Mar 19 13:29:45 DEBUG[26940]: build_route: Contact hop: Mar 19 13:29:45 VERBOSE[26940]: list_route: hop: Mar 19 13:29:45 VERBOSE[26940]: Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.1.1.100:5060;branch=z9hG4bK5F9CD27E2048483F9B7A62D198E42420 From: Wilson Chan ;tag=972238047 To: ;tag=as39c6158c Call-ID: 92882539-1348-4622-8EEE-1B51026734BE@10.1.1.100 CSeq: 41337 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 10.1.1.100:5060 Mar 19 13:29:45 DEBUG[26940]: Changing state for SIP/wilson Mar 19 13:29:45 DEBUG[26940]: Launching 'Dial' Mar 19 13:29:45 VERBOSE[26940]: -- Executing Dial("SIP/wilson-b0cc", "SIP/wilson-polycom|20|tT") in new stack Mar 19 13:29:45 DEBUG[26940]: SIMPLE DIAL (NO URL) Mar 19 13:29:45 DEBUG[26940]: Allocating new SIP call for (null) Mar 19 13:29:45 DEBUG[26940]: Setting NAT on RTP to 0 Mar 19 13:29:45 DEBUG[26940]: Not copying variable STACK-from-sip-3000-1. Mar 19 13:29:45 DEBUG[26940]: Not copying variable SIPCALLID. Mar 19 13:29:45 DEBUG[26940]: Not copying variable SIPUSERAGENT. Mar 19 13:29:45 DEBUG[26940]: Not copying variable SIPDOMAIN. Mar 19 13:29:45 DEBUG[26940]: Not copying variable SIPURI. Mar 19 13:29:45 DEBUG[26940]: Outgoing Call for wilson-polycom Mar 19 13:29:45 VERBOSE[26940]: We're at 10.1.1.48 port 17740 Mar 19 13:29:45 VERBOSE[26940]: Answering/Requesting with root capability 0x2 (gsm) Mar 19 13:29:45 VERBOSE[26940]: Answering with capability 0x4 (ulaw) Mar 19 13:29:45 VERBOSE[26940]: Answering with capability 0x8 (alaw) Mar 19 13:29:45 VERBOSE[26940]: Answering with non-codec capability 0x1 (telephone-event) Mar 19 13:29:45 VERBOSE[26940]: 12 headers, 12 lines Mar 19 13:29:45 VERBOSE[26940]: Reliably Transmitting: INVITE sip:wilson-polycom@10.1.1.200:5060 SIP/2.0 Via: SIP/2.0/UDP 10.1.1.48:5060;branch=z9hG4bK3ba007e1 From: "Wilson's Computer" ;tag=as5ab95e27 To: Contact: Call-ID: 39271f7a43b7b11c73b3bce743152e35@10.1.1.48 CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Sat, 19 Mar 2005 21:29:45 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 257 v=0 o=root 26940 26940 IN IP4 10.1.1.48 s=session c=IN IP4 10.1.1.48 t=0 0 m=audio 17740 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - (no NAT) to 10.1.1.200:5060 Mar 19 13:29:45 VERBOSE[26940]: -- Called wilson-polycom Mar 19 13:29:45 DEBUG[26940]: Set channel SIP/wilson-polycom-02d7 to read format gsm Mar 19 13:29:45 DEBUG[26940]: Set channel SIP/wilson-b0cc to write format gsm Mar 19 13:29:45 DEBUG[26940]: Set channel SIP/wilson-polycom-02d7 to write format ulaw Mar 19 13:29:45 DEBUG[26940]: Set channel SIP/wilson-b0cc to read format ulaw Mar 19 13:29:45 VERBOSE[26940]: Sip read: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.1.1.48:5060;branch=z9hG4bK3ba007e1 From: "Wilson's Computer" ;tag=as5ab95e27 To: ;tag=7FCC270E-46CCA9CB CSeq: 102 INVITE Call-ID: 39271f7a43b7b11c73b3bce743152e35@10.1.1.48 Contact: User-Agent: PolycomSoundPointIP-SPIP_600-UA/1.4.1 Content-Length: 0 Mar 19 13:29:45 VERBOSE[26940]: 9 headers, 0 lines Mar 19 13:29:45 DEBUG[26940]: (Provisional) Stopping retransmission (but retaining packet) on '39271f7a43b7b11c73b3bce743152e35@10.1.1.48' Request 102: Found Mar 19 13:29:45 VERBOSE[26940]: Sip read: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.1.1.48:5060;branch=z9hG4bK3ba007e1 From: "Wilson's Computer" ;tag=as5ab95e27 To: ;tag=7FCC270E-46CCA9CB CSeq: 102 INVITE Call-ID: 39271f7a43b7b11c73b3bce743152e35@10.1.1.48 Contact: User-Agent: PolycomSoundPointIP-SPIP_600-UA/1.4.1 Allow-Events: talk,hold,conference Content-Length: 0 Mar 19 13:29:45 VERBOSE[26940]: 10 headers, 0 lines Mar 19 13:29:45 DEBUG[26940]: (Provisional) Stopping retransmission (but retaining packet) on '39271f7a43b7b11c73b3bce743152e35@10.1.1.48' Request 102: Found Mar 19 13:29:45 DEBUG[26940]: Changing state for SIP/wilson Mar 19 13:29:45 VERBOSE[26940]: -- SIP/wilson-polycom-02d7 is ringing Mar 19 13:29:45 VERBOSE[26940]: Transmitting (no NAT): SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.1.1.100:5060;branch=z9hG4bK5F9CD27E2048483F9B7A62D198E42420 From: Wilson Chan ;tag=972238047 To: ;tag=as39c6158c Call-ID: 92882539-1348-4622-8EEE-1B51026734BE@10.1.1.100 CSeq: 41337 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 10.1.1.100:5060 Mar 19 13:29:47 VERBOSE[26940]: Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.1.48:5060;branch=z9hG4bK3ba007e1 From: "Wilson's Computer" ;tag=as5ab95e27 To: ;tag=7FCC270E-46CCA9CB CSeq: 102 INVITE Call-ID: 39271f7a43b7b11c73b3bce743152e35@10.1.1.48 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_600-UA/1.4.1 Content-Type: application/sdp Content-Length: 183 v=0 o=- 1111267783 1111267783 IN IP4 10.1.1.200 s=Polycom IP Phone c=IN IP4 10.1.1.200 t=0 0 m=audio 2250 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 Mar 19 13:29:47 VERBOSE[26940]: 11 headers, 8 lines Mar 19 13:29:47 DEBUG[26940]: Acked pending invite 102 Mar 19 13:29:47 DEBUG[26940]: Stopping retransmission on '39271f7a43b7b11c73b3bce743152e35@10.1.1.48' of Request 102: Found Mar 19 13:29:47 VERBOSE[26940]: Found RTP audio format 0 Mar 19 13:29:47 VERBOSE[26940]: Found RTP audio format 101 Mar 19 13:29:47 VERBOSE[26940]: Peer audio RTP is at port 10.1.1.200:2250 Mar 19 13:29:47 DEBUG[26940]: Peer audio RTP is at port 10.1.1.200:2250 Mar 19 13:29:47 VERBOSE[26940]: Found description format PCMU Mar 19 13:29:47 VERBOSE[26940]: Found description format telephone-event Mar 19 13:29:47 VERBOSE[26940]: Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Mar 19 13:29:47 VERBOSE[26940]: Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) Mar 19 13:29:47 DEBUG[26940]: build_route: Contact hop: Mar 19 13:29:47 VERBOSE[26940]: list_route: hop: Mar 19 13:29:47 VERBOSE[26940]: set_destination: Parsing for address/port to send to Mar 19 13:29:47 VERBOSE[26940]: set_destination: set destination to 10.1.1.200, port 5060 Mar 19 13:29:47 VERBOSE[26940]: Transmitting: ACK sip:wilson-polycom@10.1.1.200:5060 SIP/2.0 Via: SIP/2.0/UDP 10.1.1.48:5060;branch=z9hG4bK78f22c1f From: "Wilson's Computer" ;tag=as5ab95e27 To: ;tag=7FCC270E-46CCA9CB Contact: Call-ID: 39271f7a43b7b11c73b3bce743152e35@10.1.1.48 CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 10.1.1.200:5060 Mar 19 13:29:47 VERBOSE[26940]: -- SIP/wilson-polycom-02d7 answered SIP/wilson-b0cc Mar 19 13:29:47 DEBUG[26940]: Set channel SIP/wilson-b0cc to read format ulaw Mar 19 13:29:47 DEBUG[26940]: Set channel SIP/wilson-polycom-02d7 to write format ulaw Mar 19 13:29:47 DEBUG[26940]: Set channel SIP/wilson-b0cc to write format gsm Mar 19 13:29:47 DEBUG[26940]: Set channel SIP/wilson-polycom-02d7 to read format gsm Mar 19 13:29:47 DEBUG[26940]: sip_answer(SIP/wilson-b0cc) Mar 19 13:29:47 VERBOSE[26940]: We're at 10.1.1.48 port 13560 Mar 19 13:29:47 VERBOSE[26940]: Answering with preferred capability 0x2 (gsm) Mar 19 13:29:47 VERBOSE[26940]: Answering with preferred capability 0x4 (ulaw) Mar 19 13:29:47 VERBOSE[26940]: Answering with preferred capability 0x8 (alaw) Mar 19 13:29:47 VERBOSE[26940]: Answering with non-codec capability 0x1 (telephone-event) Mar 19 13:29:47 VERBOSE[26940]: Reliably Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.1.100:5060;branch=z9hG4bK5F9CD27E2048483F9B7A62D198E42420 From: Wilson Chan ;tag=972238047 To: ;tag=as39c6158c Call-ID: 92882539-1348-4622-8EEE-1B51026734BE@10.1.1.100 CSeq: 41337 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 257 v=0 o=root 26940 26940 IN IP4 10.1.1.48 s=session c=IN IP4 10.1.1.48 t=0 0 m=audio 13560 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - to 10.1.1.100:5060 Mar 19 13:29:47 VERBOSE[26940]: -- Attempting native bridge of SIP/wilson-b0cc and SIP/wilson-polycom-02d7 Mar 19 13:29:47 VERBOSE[26940]: Sip read: ACK sip:3000@10.1.1.48 SIP/2.0 Via: SIP/2.0/UDP 10.1.1.100:5060;rport;branch=z9hG4bK2EE6BAA19B6C4C50B62EE2F15AC97201 From: Wilson Chan ;tag=972238047 To: ;tag=as39c6158c Contact: Call-ID: 92882539-1348-4622-8EEE-1B51026734BE@10.1.1.100 CSeq: 41337 ACK Max-Forwards: 70 Content-Length: 0 Mar 19 13:29:47 VERBOSE[26940]: 9 headers, 0 lines Mar 19 13:29:47 DEBUG[26940]: Stopping retransmission on '92882539-1348-4622-8EEE-1B51026734BE@10.1.1.100' of Response 41337: Found Mar 19 13:29:47 DEBUG[26940]: Got RTCP report of 84 bytes Mar 19 13:29:47 DEBUG[26940]: Ooh, format changed from unknown to ulaw Mar 19 13:29:47 DEBUG[26940]: Got RTCP report of 114 bytes Mar 19 13:29:47 DEBUG[26940]: Ooh, format changed from unknown to gsm Mar 19 13:29:50 VERBOSE[26940]: Sip read: BYE sip:3002@10.1.1.48 SIP/2.0 Via: SIP/2.0/UDP 10.1.1.200:5060;branch=z9hG4bKe8524e01FB6BD1A4 From: ;tag=7FCC270E-46CCA9CB To: "Wilson's Computer" ;tag=as5ab95e27 CSeq: 1 BYE Call-ID: 39271f7a43b7b11c73b3bce743152e35@10.1.1.48 Contact: User-Agent: PolycomSoundPointIP-SPIP_600-UA/1.4.1 Max-Forwards: 70 Content-Length: 0 Mar 19 13:29:50 VERBOSE[26940]: 10 headers, 0 lines Mar 19 13:29:50 VERBOSE[26940]: Sending to 10.1.1.200 : 5060 (non-NAT) Mar 19 13:29:50 VERBOSE[26940]: Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.1.200:5060;branch=z9hG4bKe8524e01FB6BD1A4 From: ;tag=7FCC270E-46CCA9CB To: "Wilson's Computer" ;tag=as5ab95e27 Call-ID: 39271f7a43b7b11c73b3bce743152e35@10.1.1.48 CSeq: 1 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 10.1.1.200:5060 Mar 19 13:29:50 DEBUG[26940]: Didn't get a frame from channel: SIP/wilson-polycom-02d7 Mar 19 13:29:50 DEBUG[26940]: Bridge stops bridging channels SIP/wilson-b0cc and SIP/wilson-polycom-02d7 Mar 19 13:29:50 DEBUG[26940]: Hanging up channel 'SIP/wilson-polycom-02d7' Mar 19 13:29:50 DEBUG[26940]: sip_hangup(SIP/wilson-polycom-02d7) Mar 19 13:29:50 DEBUG[26940]: update_user_counter(wilson-polycom) - decrement outUse counter Mar 19 13:29:50 DEBUG[26940]: Changing state for SIP/wilson Mar 19 13:29:50 DEBUG[26940]: Exiting with DIALSTATUS=ANSWER. Mar 19 13:29:50 DEBUG[26940]: Spawn extension (from-sip,3000,1) exited non-zero on 'SIP/wilson-b0cc' Mar 19 13:29:50 DEBUG[26940]: Hanging up channel 'SIP/wilson-b0cc' Mar 19 13:29:50 DEBUG[26940]: sip_hangup(SIP/wilson-b0cc) Mar 19 13:29:50 DEBUG[26940]: update_user_counter(wilson) - decrement inUse counter Mar 19 13:29:50 VERBOSE[26940]: set_destination: Parsing for address/port to send to Mar 19 13:29:50 VERBOSE[26940]: set_destination: set destination to 10.1.1.100, port 5060 Mar 19 13:29:50 VERBOSE[26940]: Reliably Transmitting: BYE sip:wilson@10.1.1.100:5060 SIP/2.0 Via: SIP/2.0/UDP 10.1.1.48:5060;branch=z9hG4bK6949df6a;rport From: ;tag=as39c6158c To: Wilson Chan ;tag=972238047 Contact: Call-ID: 92882539-1348-4622-8EEE-1B51026734BE@10.1.1.100 CSeq: 102 BYE User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 10.1.1.100:5060 Mar 19 13:29:50 DEBUG[26940]: Changing state for SIP/wilson Mar 19 13:29:50 VERBOSE[26940]: Sip read: SIP/2.0 200 Ok Via: SIP/2.0/UDP 10.1.1.48:5060;branch=z9hG4bK6949df6a;rport From: ;tag=as39c6158c To: Wilson Chan ;tag=972238047 Contact: Call-ID: 92882539-1348-4622-8EEE-1B51026734BE@10.1.1.100 CSeq: 102 BYE Server: X-Lite release 1103m Content-Length: 0 Mar 19 13:29:50 VERBOSE[26940]: 9 headers, 0 lines Mar 19 13:29:50 DEBUG[26940]: Stopping retransmission on '92882539-1348-4622-8EEE-1B51026734BE@10.1.1.100' of Request 102: Found Mar 19 13:29:50 VERBOSE[26940]: Message is BYE Mar 19 13:29:50 VERBOSE[26940]: Destroying call '39271f7a43b7b11c73b3bce743152e35@10.1.1.48' Mar 19 13:29:50 VERBOSE[26940]: Destroying call '92882539-1348-4622-8EEE-1B51026734BE@10.1.1.100'