[Jun 2 16:14:42] <--- SIP read from UDP:192.168.5.237:5060 ---> INVITE sip:158@192.168.5.249 SIP/2.0 Via: SIP/2.0/UDP 192.168.5.237:5060;branch=z9hG4bK-3e243e7c From: 103 ;tag=f5ec85a8451bf99co0 To: Remote-Party-ID: 103 ;screen=yes;party=calling Call-ID: 70fb8f98-668ef86c@192.168.5.237 CSeq: 101 INVITE Max-Forwards: 70 Contact: 103 Expires: 240 User-Agent: Linksys/SPA2102-5.2.12 Content-Length: 444 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura, replaces Content-Type: application/sdp v=0 o=- 381093 381093 IN IP4 192.168.5.237 s=- c=IN IP4 192.168.5.237 t=0 0 m=audio 16480 RTP/AVP 8 0 18 2 4 96 97 98 100 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729a/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv <-------------> [Jun 2 16:14:42] --- (15 headers 20 lines) --- [Jun 2 16:14:42] == Using UDPTL TOS bits 10 [Jun 2 16:14:42] == Using UDPTL CoS mark 5 [Jun 2 16:14:42] Sending to 192.168.5.237:5060 (no NAT) [Jun 2 16:14:42] Using INVITE request as basis request - 70fb8f98-668ef86c@192.168.5.237 [Jun 2 16:14:42] Found peer '103' for '103' from 192.168.5.237:5060 [Jun 2 16:14:42] <--- Reliably Transmitting (NAT) to 192.168.5.237:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.5.237:5060;branch=z9hG4bK-3e243e7c;received=192.168.5.237;rport=5060 From: 103 ;tag=f5ec85a8451bf99co0 To: ;tag=as2cd4fb5f Call-ID: 70fb8f98-668ef86c@192.168.5.237 CSeq: 101 INVITE Server: Asterisk PBX 1.8.4.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="mail", nonce="5bfd6462" Content-Length: 0 <------------> [Jun 2 16:14:42] Scheduling destruction of SIP dialog '70fb8f98-668ef86c@192.168.5.237' in 6400 ms (Method: INVITE) [Jun 2 16:14:42] <--- SIP read from UDP:192.168.5.237:5060 ---> ACK sip:158@192.168.5.249 SIP/2.0 Via: SIP/2.0/UDP 192.168.5.237:5060;branch=z9hG4bK-3e243e7c From: 103 ;tag=f5ec85a8451bf99co0 To: ;tag=as2cd4fb5f Call-ID: 70fb8f98-668ef86c@192.168.5.237 CSeq: 101 ACK Max-Forwards: 70 Contact: 103 User-Agent: Linksys/SPA2102-5.2.12 Content-Length: 0 <-------------> [Jun 2 16:14:42] --- (10 headers 0 lines) --- [Jun 2 16:14:42] <--- SIP read from UDP:192.168.5.237:5060 ---> INVITE sip:158@192.168.5.249 SIP/2.0 Via: SIP/2.0/UDP 192.168.5.237:5060;branch=z9hG4bK-2adbd220 From: 103 ;tag=f5ec85a8451bf99co0 To: Remote-Party-ID: 103 ;screen=yes;party=calling Call-ID: 70fb8f98-668ef86c@192.168.5.237 CSeq: 102 INVITE Max-Forwards: 70 Authorization: Digest username="103",realm="mail",nonce="5bfd6462",uri="sip:158@192.168.5.249",algorithm=MD5,response="66046d0057cc44a3f9878f502d641469" Contact: 103 Expires: 240 User-Agent: Linksys/SPA2102-5.2.12 Content-Length: 444 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura, replaces Content-Type: application/sdp v=0 o=- 381093 381093 IN IP4 192.168.5.237 s=- c=IN IP4 192.168.5.237 t=0 0 m=audio 16480 RTP/AVP 8 0 18 2 4 96 97 98 100 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729a/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv <-------------> [Jun 2 16:14:42] --- (16 headers 20 lines) --- [Jun 2 16:14:42] Sending to 192.168.5.237:5060 (NAT) [Jun 2 16:14:42] Using INVITE request as basis request - 70fb8f98-668ef86c@192.168.5.237 [Jun 2 16:14:42] Found peer '103' for '103' from 192.168.5.237:5060 [Jun 2 16:14:42] == Using SIP RTP TOS bits 10 [Jun 2 16:14:42] == Using SIP RTP CoS mark 5 [Jun 2 16:14:42] Found RTP audio format 8 [Jun 2 16:14:42] Found RTP audio format 0 [Jun 2 16:14:42] Found RTP audio format 18 [Jun 2 16:14:42] Found RTP audio format 2 [Jun 2 16:14:42] Found RTP audio format 4 [Jun 2 16:14:42] Found RTP audio format 96 [Jun 2 16:14:42] Found RTP audio format 97 [Jun 2 16:14:42] Found RTP audio format 98 [Jun 2 16:14:42] Found RTP audio format 100 [Jun 2 16:14:42] Found RTP audio format 101 [Jun 2 16:14:42] Found audio description format PCMA for ID 8 [Jun 2 16:14:42] Found audio description format PCMU for ID 0 [Jun 2 16:14:42] Found audio description format G729a for ID 18 [Jun 2 16:14:42] Found audio description format G726-32 for ID 2 [Jun 2 16:14:42] Found audio description format G723 for ID 4 [Jun 2 16:14:42] Found audio description format G726-40 for ID 96 [Jun 2 16:14:42] Found audio description format G726-24 for ID 97 [Jun 2 16:14:42] Found audio description format G726-16 for ID 98 [Jun 2 16:14:42] Found audio description format NSE for ID 100 [Jun 2 16:14:42] Found audio description format telephone-event for ID 101 [Jun 2 16:14:42] Capabilities: us - 0x90d (g723|ulaw|alaw|g726|g729), peer - audio=0x100d0d (g723|ulaw|alaw|g726|g729|ilbc|h263p)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x90d (g723|ulaw|alaw|g726|g729) [Jun 2 16:14:42] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Jun 2 16:14:42] Peer audio RTP is at port 192.168.5.237:16480 [Jun 2 16:14:42] Looking for 158 in office1 (domain 192.168.5.249) [Jun 2 16:14:42] list_route: hop: [Jun 2 16:14:42] <--- Transmitting (NAT) to 192.168.5.237:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.5.237:5060;branch=z9hG4bK-2adbd220;received=192.168.5.237;rport=5060 From: 103 ;tag=f5ec85a8451bf99co0 To: Call-ID: 70fb8f98-668ef86c@192.168.5.237 CSeq: 102 INVITE Server: Asterisk PBX 1.8.4.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> [Jun 2 16:14:42] -- Executing [158@office1:1] Goto("SIP/103-00000063", "fax,s,1") in new stack [Jun 2 16:14:42] -- Goto (fax,s,1) [Jun 2 16:14:42] -- Executing [s@fax:1] Answer("SIP/103-00000063", "") in new stack [Jun 2 16:14:42] Audio is at 5060 [Jun 2 16:14:42] Adding codec 0x8 (alaw) to SDP [Jun 2 16:14:42] Adding codec 0x4 (ulaw) to SDP [Jun 2 16:14:42] Adding codec 0x100 (g729) to SDP [Jun 2 16:14:42] Adding codec 0x800 (g726) to SDP [Jun 2 16:14:42] Adding codec 0x1 (g723) to SDP [Jun 2 16:14:42] Adding non-codec 0x1 (telephone-event) to SDP [Jun 2 16:14:42] <--- Reliably Transmitting (NAT) to 192.168.5.237:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.5.237:5060;branch=z9hG4bK-2adbd220;received=192.168.5.237;rport=5060 From: 103 ;tag=f5ec85a8451bf99co0 To: ;tag=as334cec94 Call-ID: 70fb8f98-668ef86c@192.168.5.237 CSeq: 102 INVITE Server: Asterisk PBX 1.8.4.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 380 v=0 o=root 1240040461 1240040461 IN IP4 192.168.5.249 s=Asterisk PBX 1.8.4.1 c=IN IP4 192.168.5.249 t=0 0 m=audio 19098 RTP/AVP 8 0 18 2 4 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=fmtp:4 annexa=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <------------> [Jun 2 16:14:42] <--- SIP read from UDP:192.168.5.237:5060 ---> ACK sip:158@192.168.5.249:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.5.237:5060;branch=z9hG4bK-26845788 From: 103 ;tag=f5ec85a8451bf99co0 To: ;tag=as334cec94 Call-ID: 70fb8f98-668ef86c@192.168.5.237 CSeq: 102 ACK Max-Forwards: 70 Authorization: Digest username="103",realm="mail",nonce="5bfd6462",uri="sip:158@192.168.5.249",algorithm=MD5,response="66046d0057cc44a3f9878f502d641469" Contact: 103 User-Agent: Linksys/SPA2102-5.2.12 Content-Length: 0 <-------------> [Jun 2 16:14:42] --- (11 headers 0 lines) --- [Jun 2 16:14:42] -- Executing [s@fax:2] Set("SIP/103-00000063", "FAXFILE=/mnt/fax_kw12/fax_2011-06-02_16-14-42.tif") in new stack [Jun 2 16:14:42] -- Executing [s@fax:3] ReceiveFAX("SIP/103-00000063", "/mnt/fax_kw12/fax_2011-06-02_16-14-42.tif") in new stack [Jun 2 16:14:42] -- Channel 'SIP/103-00000063' receiving FAX '/mnt/fax_kw12/fax_2011-06-02_16-14-42.tif' [Jun 2 16:14:45] set_destination: Parsing for address/port to send to [Jun 2 16:14:45] set_destination: set destination to 192.168.5.237:5060 [Jun 2 16:14:45] Reliably Transmitting (NAT) to 192.168.5.237:5060: INVITE sip:103@192.168.5.237:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.5.249:5060;branch=z9hG4bK2b03b0bb;rport Max-Forwards: 70 From: ;tag=as334cec94 To: 103 ;tag=f5ec85a8451bf99co0 Contact: Call-ID: 70fb8f98-668ef86c@192.168.5.237 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.4.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 266 v=0 o=root 1240040461 1240040462 IN IP4 192.168.5.249 s=Asterisk PBX 1.8.4.1 c=IN IP4 192.168.5.249 t=0 0 m=image 4155 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxDatagram:849 a=T38FaxUdpEC:t38UDPFEC --- [Jun 2 16:14:45] <--- SIP read from UDP:192.168.5.237:5060 ---> SIP/2.0 200 OK To: 103 ;tag=f5ec85a8451bf99co0 From: ;tag=as334cec94 Call-ID: 70fb8f98-668ef86c@192.168.5.237 CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.5.249:5060;branch=z9hG4bK2b03b0bb Contact: 103 Server: Linksys/SPA2102-5.2.12 Content-Length: 267 Content-Type: application/sdp v=0 o=- 381405 381405 IN IP4 192.168.5.237 s=- c=IN IP4 192.168.5.237 t=0 0 m=image 16480 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:200 a=T38FaxMaxDatagram:200 a=T38FaxUdpEC:t38UDPRedundancy <-------------> [Jun 2 16:14:45] --- (10 headers 12 lines) --- [Jun 2 16:14:45] Got T.38 offer in SDP in dialog 70fb8f98-668ef86c@192.168.5.237 [Jun 2 16:14:46] Capabilities: us - 0x90d (g723|ulaw|alaw|g726|g729), peer - audio=0x0 (nothing)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x0 (nothing) [Jun 2 16:14:46] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing) [Jun 2 16:14:46] Got T.38 Re-invite without audio. Keeping RTP active during T.38 session. [Jun 2 16:14:46] set_destination: Parsing for address/port to send to [Jun 2 16:14:46] set_destination: set destination to 192.168.5.237:5060 [Jun 2 16:14:46] Transmitting (NAT) to 192.168.5.237:5060: ACK sip:103@192.168.5.237:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.5.249:5060;branch=z9hG4bK059e67fa;rport Max-Forwards: 70 From: ;tag=as334cec94 To: 103 ;tag=f5ec85a8451bf99co0 Contact: Call-ID: 70fb8f98-668ef86c@192.168.5.237 CSeq: 102 ACK User-Agent: Asterisk PBX 1.8.4.1 Content-Length: 0 --- [Jun 2 16:14:53] Reliably Transmitting (NAT) to 192.168.5.237:5060: OPTIONS sip:103@192.168.5.237:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.5.249:5060;branch=z9hG4bK7c9c6d99;rport Max-Forwards: 70 From: "asterisk" ;tag=as1c9981c6 To: Contact: Call-ID: 0e4b62b847063ceb60ecc2452e3c3ecf@192.168.5.249:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.4.1 Date: Thu, 02 Jun 2011 09:14:53 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Jun 2 16:14:53] <--- SIP read from UDP:192.168.5.237:5060 ---> SIP/2.0 200 OK To: ;tag=9304557d408e15a1i0 From: "asterisk" ;tag=as1c9981c6 Call-ID: 0e4b62b847063ceb60ecc2452e3c3ecf@192.168.5.249:5060 CSeq: 102 OPTIONS Via: SIP/2.0/UDP 192.168.5.249:5060;branch=z9hG4bK7c9c6d99 Server: Linksys/SPA2102-5.2.12 Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura, replaces <-------------> [Jun 2 16:14:53] --- (10 headers 0 lines) --- [Jun 2 16:14:53] Really destroying SIP dialog '0e4b62b847063ceb60ecc2452e3c3ecf@192.168.5.249:5060' Method: OPTIONS [Jun 2 16:14:53] NOTICE[19931]: chan_sip.c:12419 sip_reregister: -- Re-registration for 3832091931@195.189.239.32 [Jun 2 16:14:53] > doing dnsmgr_lookup for '195.189.239.32' [Jun 2 16:14:53] > doing dnsmgr_lookup for '195.189.239.32' [Jun 2 16:14:53] NOTICE[19931]: chan_sip.c:19815 handle_response_register: Outbound Registration: Expiry for 195.189.239.32 is 120 sec (Scheduling reregistration in 105 s) pbx*CLI> pbx*CLI> pbx*CLI> pbx*CLI> pbx*CLI> pbx*CLI> pbx*CLI> [Jun 2 16:15:25] set_destination: Parsing for address/port to send to [Jun 2 16:15:25] set_destination: set destination to 192.168.5.237:5060 [Jun 2 16:15:25] Audio is at 5060 [Jun 2 16:15:25] Adding codec 0x8 (alaw) to SDP [Jun 2 16:15:25] Adding codec 0x4 (ulaw) to SDP [Jun 2 16:15:25] Adding codec 0x100 (g729) to SDP [Jun 2 16:15:25] Adding codec 0x800 (g726) to SDP [Jun 2 16:15:25] Adding codec 0x1 (g723) to SDP [Jun 2 16:15:25] Adding non-codec 0x1 (telephone-event) to SDP [Jun 2 16:15:25] Reliably Transmitting (NAT) to 192.168.5.237:5060: INVITE sip:103@192.168.5.237:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.5.249:5060;branch=z9hG4bK0221c7da;rport Max-Forwards: 70 From: ;tag=as334cec94 To: 103 ;tag=f5ec85a8451bf99co0 Contact: Call-ID: 70fb8f98-668ef86c@192.168.5.237 CSeq: 103 INVITE User-Agent: Asterisk PBX 1.8.4.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 380 v=0 o=root 1240040461 1240040463 IN IP4 192.168.5.249 s=Asterisk PBX 1.8.4.1 c=IN IP4 192.168.5.249 t=0 0 m=audio 19098 RTP/AVP 8 0 18 2 4 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=fmtp:4 annexa=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Jun 2 16:15:25] <--- SIP read from UDP:192.168.5.237:5060 ---> SIP/2.0 200 OK To: 103 ;tag=f5ec85a8451bf99co0 From: ;tag=as334cec94 Call-ID: 70fb8f98-668ef86c@192.168.5.237 CSeq: 103 INVITE Via: SIP/2.0/UDP 192.168.5.249:5060;branch=z9hG4bK0221c7da Contact: 103 Server: Linksys/SPA2102-5.2.12 Content-Length: 255 Content-Type: application/sdp v=0 o=- 385325 385325 IN IP4 192.168.5.237 s=- c=IN IP4 192.168.5.237 t=0 0 m=audio 16480 RTP/AVP 8 100 101 a=rtpmap:8 PCMA/8000 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv <-------------> [Jun 2 16:15:25] --- (10 headers 13 lines) --- [Jun 2 16:15:25] Found RTP audio format 8 [Jun 2 16:15:25] Found RTP audio format 100 [Jun 2 16:15:25] Found RTP audio format 101 [Jun 2 16:15:25] Found audio description format PCMA for ID 8 [Jun 2 16:15:25] Found audio description format NSE for ID 100 [Jun 2 16:15:25] Found audio description format telephone-event for ID 101 [Jun 2 16:15:25] Capabilities: us - 0x90d (g723|ulaw|alaw|g726|g729), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) [Jun 2 16:15:25] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Jun 2 16:15:25] Peer audio RTP is at port 192.168.5.237:16480 [Jun 2 16:15:25] set_destination: Parsing for address/port to send to [Jun 2 16:15:25] set_destination: set destination to 192.168.5.237:5060 [Jun 2 16:15:25] Transmitting (NAT) to 192.168.5.237:5060: ACK sip:103@192.168.5.237:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.5.249:5060;branch=z9hG4bK072adc8b;rport Max-Forwards: 70 From: ;tag=as334cec94 To: 103 ;tag=f5ec85a8451bf99co0 Contact: Call-ID: 70fb8f98-668ef86c@192.168.5.237 CSeq: 103 ACK User-Agent: Asterisk PBX 1.8.4.1 Content-Length: 0 --- [Jun 2 16:15:25] -- Auto fallthrough, channel 'SIP/103-00000063' status is 'UNKNOWN' [Jun 2 16:15:25] -- Executing [h@fax:1] System("SIP/103-00000063", "chmod 777 /mnt/fax_kw12/fax_2011-06-02_16-14-42.tif") in new stack [Jun 2 16:15:25] -- Executing [h@fax:2] NoOp("SIP/103-00000063", "SUCCESS") in new stack [Jun 2 16:15:25] -- Executing [h@fax:3] NoOp("SIP/103-00000063", "") in new stack [Jun 2 16:15:25] Scheduling destruction of SIP dialog '70fb8f98-668ef86c@192.168.5.237' in 6400 ms (Method: ACK) [Jun 2 16:15:25] set_destination: Parsing for address/port to send to [Jun 2 16:15:25] set_destination: set destination to 192.168.5.237:5060 [Jun 2 16:15:25] Reliably Transmitting (NAT) to 192.168.5.237:5060: BYE sip:103@192.168.5.237:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.5.249:5060;branch=z9hG4bK267a0702;rport Max-Forwards: 70 From: ;tag=as334cec94 To: 103 ;tag=f5ec85a8451bf99co0 Call-ID: 70fb8f98-668ef86c@192.168.5.237 CSeq: 104 BYE User-Agent: Asterisk PBX 1.8.4.1 Proxy-Authorization: Digest username="103", realm="mail", algorithm=MD5, uri="192.168.5.249", nonce="", response="0a86260e8deb20fa8b193b050fcf6aaf" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [Jun 2 16:15:25] <--- SIP read from UDP:192.168.5.237:5060 ---> SIP/2.0 200 OK To: 103 ;tag=f5ec85a8451bf99co0 From: ;tag=as334cec94 Call-ID: 70fb8f98-668ef86c@192.168.5.237 CSeq: 104 BYE Via: SIP/2.0/UDP 192.168.5.249:5060;branch=z9hG4bK267a0702 Server: Linksys/SPA2102-5.2.12 Content-Length: 0 <-------------> [Jun 2 16:15:25] --- (8 headers 0 lines) --- [Jun 2 16:15:25] Really destroying SIP dialog '70fb8f98-668ef86c@192.168.5.237' Method: ACK