root@k-tanco:~> asterisk -R Asterisk 1.6.0-rc6, Copyright (C) 1999 - 2008 Digium, Inc. and others. Created by Mark Spencer Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core 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 'core show license' for details. ========================================================================= Connected to Asterisk 1.6.0-rc6 currently running on k-tanco (pid = 27014) Verbosity is at least 10 Core debug is at least 10 k-tanco*CLI> sip set debug on SIP Debugging enabled [Sep 20 13:33:29] <--- SIP read from UDP://RFC-1918 IP:35044 ---> SUBSCRIBE sip:asterisk@RFC-1918 IP SIP/2.0 Via: SIP/2.0/UDP RFC-1918 IP:35044;branch=z9hG4bK-d8754z-6c42ae2bb753c337-1---d8754z-;rport Max-Forwards: 70 Contact: To: "Test";tag=as7e1bdf6e From: "Test";tag=1d72747f Call-ID: ZTk0ODIxN2VjYmM0MGY5M2UwMWY0YTQ2ZDQ2ZjNlMzM. CSeq: 22 SUBSCRIBE Expires: 300 User-Agent: X-Lite release 1100l stamp 47546 Authorization: Digest username="6005",realm="asterisk",nonce="00546d80",uri="sip:asterisk@RFC-1918 IP",response="e01c0eae3010e6a3165b11c480404208",algorithm=MD5 Event: message-summary Content-Length: 0 <-------------> [Sep 20 13:33:29] --- (13 headers 0 lines) --- [Sep 20 13:33:29] Found peer '6005' for '6005' from RFC-1918 IP:35044 [Sep 20 13:33:29] Looking for 6005 in client_int_firma (domain gofferje.homelinux.org) [Sep 20 13:33:29] Scheduling destruction of SIP dialog 'ZTk0ODIxN2VjYmM0MGY5M2UwMWY0YTQ2ZDQ2ZjNlMzM.' in 310000 ms (Method: SUBSCRIBE) [Sep 20 13:33:29] <--- Transmitting (no NAT) to RFC-1918 IP:35044 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP RFC-1918 IP:35044;branch=z9hG4bK-d8754z-6c42ae2bb753c337-1---d8754z-;received=RFC-1918 IP;rport=35044 From: "Test";tag=1d72747f To: "Test";tag=as7e1bdf6e Call-ID: ZTk0ODIxN2VjYmM0MGY5M2UwMWY0YTQ2ZDQ2ZjNlMzM. CSeq: 22 SUBSCRIBE User-Agent: Asterisk PBX 1.6.0-rc6 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Expires: 300 Contact: ;expires=300 Content-Length: 0 <------------> [Sep 20 13:33:29] Reliably Transmitting (no NAT) to RFC-1918 IP:35044: NOTIFY sip:6005@RFC-1918 IP:35044 SIP/2.0 Via: SIP/2.0/UDP RFC-1918 IP:5060;branch=z9hG4bK3b6b0633;rport Max-Forwards: 70 From: "asterisk" ;tag=as7e1bdf6e To: ;tag=1d72747f Contact: Call-ID: ZTk0ODIxN2VjYmM0MGY5M2UwMWY0YTQ2ZDQ2ZjNlMzM. CSeq: 121 NOTIFY User-Agent: Asterisk PBX 1.6.0-rc6 Event: message-summary Content-Type: application/simple-message-summary Subscription-State: active Content-Length: 98 Messages-Waiting: no Message-Account: sip:8500@gofferje.homelinux.org Voice-Message: 0/0 (0/0) --- [Sep 20 13:33:29] <--- SIP read from UDP://RFC-1918 IP:35044 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP RFC-1918 IP:5060;branch=z9hG4bK3b6b0633;rport=5060 Contact: To: ;tag=1d72747f From: "asterisk";tag=as7e1bdf6e Call-ID: ZTk0ODIxN2VjYmM0MGY5M2UwMWY0YTQ2ZDQ2ZjNlMzM. CSeq: 121 NOTIFY User-Agent: X-Lite release 1100l stamp 47546 Content-Length: 0 <-------------> [Sep 20 13:33:29] --- (9 headers 0 lines) --- [Sep 20 13:33:34] Reliably Transmitting (no NAT) to RFC-1918 IP:5060: OPTIONS sip:sgofferj@RFC-1918 IP SIP/2.0 Via: SIP/2.0/UDP RFC-1918 IP:5060;branch=z9hG4bK79440ad5;rport Max-Forwards: 70 From: "asterisk" ;tag=as04b7ff2e To: Contact: Call-ID: 2964f0f410c1bee73ff597ed26f3d8dd@RFC-1918 IP CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.0-rc6 Date: Sat, 20 Sep 2008 10:33:34 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Length: 0 --- [Sep 20 13:33:34] <--- SIP read from UDP://RFC-1918 IP:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP RFC-1918 IP:5060;branch=z9hG4bK79440ad5;rport=5060;received=RFC-1918 IP To: ;tag=3vd8p8e6fdhc7krio296 From: "asterisk" ;tag=as04b7ff2e Call-ID: 2964f0f410c1bee73ff597ed26f3d8dd@RFC-1918 IP CSeq: 102 OPTIONS Content-Length: 0 <-------------> [Sep 20 13:33:34] --- (7 headers 0 lines) --- [Sep 20 13:33:34] Really destroying SIP dialog '2964f0f410c1bee73ff597ed26f3d8dd@RFC-1918 IP' Method: OPTIONS [Sep 20 13:33:35] <--- SIP read from UDP://RFC-1918 IP:35044 ---> <-------------> [Sep 20 13:33:40] <--- SIP read from UDP://RFC-1918 IP:35044 ---> INVITE sip:6100@gofferje.homelinux.org SIP/2.0 Via: SIP/2.0/UDP RFC-1918 IP:35044;branch=z9hG4bK-d8754z-f86f940bf923673f-1---d8754z-;rport Max-Forwards: 70 Contact: To: "6100" From: "Test";tag=bd1ce66e Call-ID: MmMyMWM5Zjg2MDYzZWEzNTdjZjcwY2ZjOTUyNDAyYjA. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: X-Lite release 1100l stamp 47546 Content-Length: 233 v=0 o=- 3 2 IN IP4 RFC-1918 IP s=CounterPath X-Lite 3.0 c=IN IP4 RFC-1918 IP t=0 0 m=audio 46750 RTP/AVP 3 101 a=alt:1 1 : QO9B0Xrj H3qu66km RFC-1918 IP 46750 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv <-------------> [Sep 20 13:33:40] --- (12 headers 10 lines) --- [Sep 20 13:33:40] == Using SIP RTP CoS mark 5 [Sep 20 13:33:40] == Using SIP VRTP CoS mark 6 [Sep 20 13:33:40] Sending to RFC-1918 IP : 35044 (NAT) [Sep 20 13:33:40] Using INVITE request as basis request - MmMyMWM5Zjg2MDYzZWEzNTdjZjcwY2ZjOTUyNDAyYjA. [Sep 20 13:33:40] Found user '6005' for '6005' [Sep 20 13:33:40] <--- Reliably Transmitting (no NAT) to RFC-1918 IP:35044 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP RFC-1918 IP:35044;branch=z9hG4bK-d8754z-f86f940bf923673f-1---d8754z-;received=RFC-1918 IP;rport=35044 From: "Test";tag=bd1ce66e To: "6100";tag=as71054758 Call-ID: MmMyMWM5Zjg2MDYzZWEzNTdjZjcwY2ZjOTUyNDAyYjA. CSeq: 1 INVITE User-Agent: Asterisk PBX 1.6.0-rc6 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="78bba6ec" Content-Length: 0 <------------> [Sep 20 13:33:40] Scheduling destruction of SIP dialog 'MmMyMWM5Zjg2MDYzZWEzNTdjZjcwY2ZjOTUyNDAyYjA.' in 32000 ms (Method: INVITE) [Sep 20 13:33:40] <--- SIP read from UDP://RFC-1918 IP:35044 ---> ACK sip:6100@gofferje.homelinux.org SIP/2.0 Via: SIP/2.0/UDP RFC-1918 IP:35044;branch=z9hG4bK-d8754z-f86f940bf923673f-1---d8754z-;rport To: "6100";tag=as71054758 From: "Test";tag=bd1ce66e Call-ID: MmMyMWM5Zjg2MDYzZWEzNTdjZjcwY2ZjOTUyNDAyYjA. CSeq: 1 ACK Content-Length: 0 <-------------> [Sep 20 13:33:40] --- (7 headers 0 lines) --- [Sep 20 13:33:40] <--- SIP read from UDP://RFC-1918 IP:35044 ---> INVITE sip:6100@gofferje.homelinux.org SIP/2.0 Via: SIP/2.0/UDP RFC-1918 IP:35044;branch=z9hG4bK-d8754z-6b473e3aca2ff739-1---d8754z-;rport Max-Forwards: 70 Contact: To: "6100" From: "Test";tag=bd1ce66e Call-ID: MmMyMWM5Zjg2MDYzZWEzNTdjZjcwY2ZjOTUyNDAyYjA. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: X-Lite release 1100l stamp 47546 Authorization: Digest username="6005",realm="asterisk",nonce="78bba6ec",uri="sip:6100@gofferje.homelinux.org",response="dedf4381d254cfeb581026edb348c19d",algorithm=MD5 Content-Length: 233 v=0 o=- 3 2 IN IP4 RFC-1918 IP s=CounterPath X-Lite 3.0 c=IN IP4 RFC-1918 IP t=0 0 m=audio 46750 RTP/AVP 3 101 a=alt:1 1 : QO9B0Xrj H3qu66km RFC-1918 IP 46750 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv <-------------> [Sep 20 13:33:40] --- (13 headers 10 lines) --- [Sep 20 13:33:40] Sending to RFC-1918 IP : 35044 (NAT) [Sep 20 13:33:40] Using INVITE request as basis request - MmMyMWM5Zjg2MDYzZWEzNTdjZjcwY2ZjOTUyNDAyYjA. [Sep 20 13:33:40] Found user '6005' for '6005' [Sep 20 13:33:40] Found RTP audio format 3 [Sep 20 13:33:40] Found RTP audio format 101 [Sep 20 13:33:40] Peer audio RTP is at port RFC-1918 IP:46750 [Sep 20 13:33:40] Got unsupported a:fmtp in SDP offer [Sep 20 13:33:40] Found audio description format telephone-event for ID 101 [Sep 20 13:33:40] Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x2 (gsm)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x2 (gsm) [Sep 20 13:33:40] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Sep 20 13:33:40] Peer audio RTP is at port RFC-1918 IP:46750 [Sep 20 13:33:40] Looking for 6100 in client_int_firma (domain gofferje.homelinux.org) [Sep 20 13:33:40] list_route: hop: [Sep 20 13:33:40] <--- Transmitting (no NAT) to RFC-1918 IP:35044 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP RFC-1918 IP:35044;branch=z9hG4bK-d8754z-6b473e3aca2ff739-1---d8754z-;received=RFC-1918 IP;rport=35044 From: "Test";tag=bd1ce66e To: "6100" Call-ID: MmMyMWM5Zjg2MDYzZWEzNTdjZjcwY2ZjOTUyNDAyYjA. CSeq: 2 INVITE User-Agent: Asterisk PBX 1.6.0-rc6 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: Content-Length: 0 <------------> [Sep 20 13:33:40] -- Executing [6100@client_int_firma:1] Macro("SIP/6005-08267e98", "personcall,6100,1000") in new stack [Sep 20 13:33:40] -- Executing [s@macro-personcall:1] GotoIf("SIP/6005-08267e98", "?:3") in new stack [Sep 20 13:33:40] -- Goto (macro-personcall,s,3) [Sep 20 13:33:40] -- Executing [s@macro-personcall:3] Dial("SIP/6005-08267e98", "SIP/sgofferj,30") in new stack [Sep 20 13:33:40] == Using SIP RTP CoS mark 5 [Sep 20 13:33:40] == Using SIP VRTP CoS mark 6 [Sep 20 13:33:40] Audio is at RFC-1918 IP port 12428 [Sep 20 13:33:40] Adding codec 0x2 (gsm) to SDP [Sep 20 13:33:40] Adding codec 0x4 (ulaw) to SDP [Sep 20 13:33:40] Adding codec 0x8 (alaw) to SDP [Sep 20 13:33:40] Adding non-codec 0x1 (telephone-event) to SDP [Sep 20 13:33:40] Reliably Transmitting (no NAT) to RFC-1918 IP:5060: INVITE sip:sgofferj@RFC-1918 IP SIP/2.0 Via: SIP/2.0/UDP RFC-1918 IP:5060;branch=z9hG4bK188d451b;rport Max-Forwards: 70 From: "6005" ;tag=as7848f402 To: Contact: Call-ID: 1ae96a4f338a933964f3a9317833cf25@gofferje.homelinux.org CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.0-rc6 Date: Sat, 20 Sep 2008 10:33:40 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Type: application/sdp Content-Length: 314 v=0 o=root 444750411 444750411 IN IP4 RFC-1918 IP s=Asterisk PBX 1.6.0-rc6 c=IN IP4 RFC-1918 IP t=0 0 m=audio 12428 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 - - - - a=ptime:20 a=sendrecv --- [Sep 20 13:33:40] -- Called sgofferj [Sep 20 13:33:40] <--- SIP read from UDP://RFC-1918 IP:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP RFC-1918 IP:5060;branch=z9hG4bK188d451b;rport=5060;received=RFC-1918 IP To: From: "6005" ;tag=as7848f402 Call-ID: 1ae96a4f338a933964f3a9317833cf25@gofferje.homelinux.org CSeq: 102 INVITE Content-Length: 0 <-------------> [Sep 20 13:33:40] --- (7 headers 0 lines) --- [Sep 20 13:33:41] <--- SIP read from UDP://RFC-1918 IP:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP RFC-1918 IP:5060;branch=z9hG4bK188d451b;rport=5060;received=RFC-1918 IP From: "6005" ;tag=as7848f402 To: ;tag=772oho6nv15nt6k7tu8n43b2of05phpc Contact: Call-ID: 1ae96a4f338a933964f3a9317833cf25@gofferje.homelinux.org CSeq: 102 INVITE Content-Length: 0 <-------------> [Sep 20 13:33:41] --- (8 headers 0 lines) --- [Sep 20 13:33:41] -- SIP/sgofferj-08308760 is ringing [Sep 20 13:33:41] <--- Transmitting (no NAT) to RFC-1918 IP:35044 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP RFC-1918 IP:35044;branch=z9hG4bK-d8754z-6b473e3aca2ff739-1---d8754z-;received=RFC-1918 IP;rport=35044 From: "Test";tag=bd1ce66e To: "6100";tag=as2925a42c Call-ID: MmMyMWM5Zjg2MDYzZWEzNTdjZjcwY2ZjOTUyNDAyYjA. CSeq: 2 INVITE User-Agent: Asterisk PBX 1.6.0-rc6 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: Content-Length: 0 <------------> [Sep 20 13:33:47] <--- SIP read from UDP://RFC-1918 IP:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP RFC-1918 IP:5060;branch=z9hG4bK188d451b;rport=5060;received=RFC-1918 IP To: ;tag=772oho6nv15nt6k7tu8n43b2of05phpc Contact: From: "6005" ;tag=as7848f402 Call-ID: 1ae96a4f338a933964f3a9317833cf25@gofferje.homelinux.org CSeq: 102 INVITE Allow: INVITE,ACK,BYE,CANCEL,REFER,NOTIFY,OPTIONS,PRACK Content-Type: application/sdp Accept: application/sdp Content-Length: 263 v=0 o=Nokia-SIPUA 444750412 444750412 IN IP4 RFC-1918 IP s=Asterisk PBX 1.6.0-rc6 c=IN IP4 RFC-1918 IP t=0 0 m=audio 49152 RTP/AVP 0 101 a=sendrecv a=ptime:20 a=maxptime:200 a=fmtp:101 0-15 a=rtpmap:0 PCMU/8000/1 a=rtpmap:101 telephone-event/8000/1 <-------------> [Sep 20 13:33:47] --- (11 headers 12 lines) --- [Sep 20 13:33:47] Found RTP audio format 0 [Sep 20 13:33:47] Found RTP audio format 101 [Sep 20 13:33:47] Peer audio RTP is at port RFC-1918 IP:49152 [Sep 20 13:33:47] Got unsupported a:fmtp in SDP offer [Sep 20 13:33:47] Found audio description format PCMU for ID 0 [Sep 20 13:33:47] Found audio description format telephone-event for ID 101 [Sep 20 13:33:47] Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) [Sep 20 13:33:47] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Sep 20 13:33:47] Peer audio RTP is at port RFC-1918 IP:49152 [Sep 20 13:33:47] list_route: hop: [Sep 20 13:33:47] set_destination: Parsing for address/port to send to [Sep 20 13:33:47] set_destination: set destination to RFC-1918 IP, port 5060 [Sep 20 13:33:47] Transmitting (no NAT) to RFC-1918 IP:5060: ACK sip:sgofferj@RFC-1918 IP SIP/2.0 Via: SIP/2.0/UDP RFC-1918 IP:5060;branch=z9hG4bK2439d8c2;rport Max-Forwards: 70 From: "6005" ;tag=as7848f402 To: ;tag=772oho6nv15nt6k7tu8n43b2of05phpc Contact: Call-ID: 1ae96a4f338a933964f3a9317833cf25@gofferje.homelinux.org CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.0-rc6 Content-Length: 0 --- [Sep 20 13:33:47] -- SIP/sgofferj-08308760 answered SIP/6005-08267e98 [Sep 20 13:33:47] Audio is at RFC-1918 IP port 19378 [Sep 20 13:33:47] Adding codec 0x2 (gsm) to SDP [Sep 20 13:33:47] Adding non-codec 0x1 (telephone-event) to SDP [Sep 20 13:33:47] <--- Reliably Transmitting (no NAT) to RFC-1918 IP:35044 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP RFC-1918 IP:35044;branch=z9hG4bK-d8754z-6b473e3aca2ff739-1---d8754z-;received=RFC-1918 IP;rport=35044 From: "Test";tag=bd1ce66e To: "6100";tag=as2925a42c Call-ID: MmMyMWM5Zjg2MDYzZWEzNTdjZjcwY2ZjOTUyNDAyYjA. CSeq: 2 INVITE User-Agent: Asterisk PBX 1.6.0-rc6 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 266 v=0 o=root 249653479 249653479 IN IP4 RFC-1918 IP s=Asterisk PBX 1.6.0-rc6 c=IN IP4 RFC-1918 IP t=0 0 m=audio 19378 RTP/AVP 3 101 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Sep 20 13:33:47] <--- SIP read from UDP://RFC-1918 IP:35044 ---> ACK sip:6100@RFC-1918 IP SIP/2.0 Via: SIP/2.0/UDP RFC-1918 IP:35044;branch=z9hG4bK-d8754z-eb0db2072147837b-1---d8754z-;rport Max-Forwards: 70 Contact: To: "6100";tag=as2925a42c From: "Test";tag=bd1ce66e Call-ID: MmMyMWM5Zjg2MDYzZWEzNTdjZjcwY2ZjOTUyNDAyYjA. CSeq: 2 ACK User-Agent: X-Lite release 1100l stamp 47546 Authorization: Digest username="6005",realm="asterisk",nonce="78bba6ec",uri="sip:6100@gofferje.homelinux.org",response="dedf4381d254cfeb581026edb348c19d",algorithm=MD5 Content-Length: 0 <-------------> [Sep 20 13:33:47] --- (11 headers 0 lines) --- [Sep 20 13:33:52] <--- SIP read from UDP://RFC-1918 IP:5060 ---> INVITE sip:6005@RFC-1918 IP SIP/2.0 Via: SIP/2.0/UDP RFC-1918 IP:5060;branch=z9hG4bKd39er679jdhc6333s4ndg7o;rport To: "6005" ;tag=as7848f402 From: ;tag=772oho6nv15nt6k7tu8n43b2of05phpc Contact: Call-ID: 1ae96a4f338a933964f3a9317833cf25@gofferje.homelinux.org CSeq: 1 INVITE Allow: INVITE,ACK,BYE,CANCEL,REFER,NOTIFY,OPTIONS,PRACK Expires: 10 User-Agent: Nokia RM-244 200.34.36 Max-Forwards: 70 Content-Type: application/sdp Accept: application/sdp Content-Length: 263 v=0 o=Nokia-SIPUA 444750412 444750413 IN IP4 RFC-1918 IP s=Asterisk PBX 1.6.0-rc6 c=IN IP4 RFC-1918 IP t=0 0 m=audio 49152 RTP/AVP 0 101 a=sendonly a=ptime:20 a=maxptime:200 a=fmtp:101 0-15 a=rtpmap:0 PCMU/8000/1 a=rtpmap:101 telephone-event/8000/1 <-------------> [Sep 20 13:33:52] --- (14 headers 12 lines) --- [Sep 20 13:33:52] Sending to RFC-1918 IP : 5060 (NAT) [Sep 20 13:33:52] Found RTP audio format 0 [Sep 20 13:33:52] Found RTP audio format 101 [Sep 20 13:33:52] Peer audio RTP is at port RFC-1918 IP:49152 [Sep 20 13:33:52] Got unsupported a:fmtp in SDP offer [Sep 20 13:33:52] Found audio description format PCMU for ID 0 [Sep 20 13:33:52] Found audio description format telephone-event for ID 101 [Sep 20 13:33:52] Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) [Sep 20 13:33:52] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Sep 20 13:33:52] Peer audio RTP is at port RFC-1918 IP:49152 [Sep 20 13:33:52] <--- Transmitting (NAT) to RFC-1918 IP:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP RFC-1918 IP:5060;branch=z9hG4bKd39er679jdhc6333s4ndg7o;received=RFC-1918 IP;rport=5060 From: ;tag=772oho6nv15nt6k7tu8n43b2of05phpc To: "6005" ;tag=as7848f402 Call-ID: 1ae96a4f338a933964f3a9317833cf25@gofferje.homelinux.org CSeq: 1 INVITE User-Agent: Asterisk PBX 1.6.0-rc6 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: Content-Length: 0 <------------> [Sep 20 13:33:52] Audio is at RFC-1918 IP port 12428 [Sep 20 13:33:52] Adding codec 0x4 (ulaw) to SDP [Sep 20 13:33:52] Adding non-codec 0x1 (telephone-event) to SDP [Sep 20 13:33:52] <--- Reliably Transmitting (NAT) to RFC-1918 IP:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP RFC-1918 IP:5060;branch=z9hG4bKd39er679jdhc6333s4ndg7o;received=RFC-1918 IP;rport=5060 From: ;tag=772oho6nv15nt6k7tu8n43b2of05phpc To: "6005" ;tag=as7848f402 Call-ID: 1ae96a4f338a933964f3a9317833cf25@gofferje.homelinux.org CSeq: 1 INVITE User-Agent: Asterisk PBX 1.6.0-rc6 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 267 v=0 o=root 444750411 444750412 IN IP4 RFC-1918 IP s=Asterisk PBX 1.6.0-rc6 c=IN IP4 RFC-1918 IP t=0 0 m=audio 12428 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=recvonly <------------> [Sep 20 13:33:52] -- Started music on hold, class 'default', on channel 'SIP/6005-08267e98' [Sep 20 13:33:53] Retransmitting #1 (NAT) to RFC-1918 IP:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP RFC-1918 IP:5060;branch=z9hG4bKd39er679jdhc6333s4ndg7o;received=RFC-1918 IP;rport=5060 From: ;tag=772oho6nv15nt6k7tu8n43b2of05phpc To: "6005" ;tag=as7848f402 Call-ID: 1ae96a4f338a933964f3a9317833cf25@gofferje.homelinux.org CSeq: 1 INVITE User-Agent: Asterisk PBX 1.6.0-rc6 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 267 v=0 o=root 444750411 444750412 IN IP4 RFC-1918 IP s=Asterisk PBX 1.6.0-rc6 c=IN IP4 RFC-1918 IP t=0 0 m=audio 12428 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=recvonly --- [Sep 20 13:33:53] <--- SIP read from UDP://RFC-1918 IP:5060 ---> ACK sip:6005@RFC-1918 IP SIP/2.0 Via: SIP/2.0/UDP RFC-1918 IP:5060;branch=z9hG4bKrgv2nutmviq7dap0pi45rrb;rport To: "6005" ;tag=as7848f402 From: ;tag=772oho6nv15nt6k7tu8n43b2of05phpc Call-ID: 1ae96a4f338a933964f3a9317833cf25@gofferje.homelinux.org CSeq: 1 ACK Supported: sec-agree Max-Forwards: 70 Content-Length: 0 <-------------> [Sep 20 13:33:53] --- (9 headers 0 lines) --- [Sep 20 13:33:53] <--- SIP read from UDP://RFC-1918 IP:5060 ---> ACK sip:6005@RFC-1918 IP SIP/2.0 Via: SIP/2.0/UDP RFC-1918 IP:5060;branch=z9hG4bKrgv2nutmviq7dap0pi45rrb;rport To: "6005" ;tag=as7848f402 From: ;tag=772oho6nv15nt6k7tu8n43b2of05phpc Call-ID: 1ae96a4f338a933964f3a9317833cf25@gofferje.homelinux.org CSeq: 1 ACK Supported: sec-agree Max-Forwards: 70 Content-Length: 0 <-------------> [Sep 20 13:33:53] --- (9 headers 0 lines) --- [Sep 20 13:34:01] <--- SIP read from UDP://RFC-1918 IP:5060 ---> INVITE sip:6005@RFC-1918 IP SIP/2.0 Via: SIP/2.0/UDP RFC-1918 IP:5060;branch=z9hG4bKmcfqve41ephc6d99m5g9caq;rport To: "6005" ;tag=as7848f402 From: ;tag=772oho6nv15nt6k7tu8n43b2of05phpc Contact: Call-ID: 1ae96a4f338a933964f3a9317833cf25@gofferje.homelinux.org CSeq: 2 INVITE Allow: INVITE,ACK,BYE,CANCEL,REFER,NOTIFY,OPTIONS,PRACK Expires: 10 User-Agent: Nokia RM-244 200.34.36 Max-Forwards: 70 Content-Type: application/sdp Accept: application/sdp Content-Length: 263 v=0 o=Nokia-SIPUA 444750412 444750414 IN IP4 RFC-1918 IP s=Asterisk PBX 1.6.0-rc6 c=IN IP4 RFC-1918 IP t=0 0 m=audio 49152 RTP/AVP 0 101 a=sendrecv a=ptime:20 a=maxptime:200 a=fmtp:101 0-15 a=rtpmap:0 PCMU/8000/1 a=rtpmap:101 telephone-event/8000/1 <-------------> [Sep 20 13:34:01] --- (14 headers 12 lines) --- [Sep 20 13:34:01] Sending to RFC-1918 IP : 5060 (NAT) [Sep 20 13:34:01] Found RTP audio format 0 [Sep 20 13:34:01] Found RTP audio format 101 [Sep 20 13:34:01] Peer audio RTP is at port RFC-1918 IP:49152 [Sep 20 13:34:01] Got unsupported a:fmtp in SDP offer [Sep 20 13:34:01] Found audio description format PCMU for ID 0 [Sep 20 13:34:01] Found audio description format telephone-event for ID 101 [Sep 20 13:34:01] Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) [Sep 20 13:34:01] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Sep 20 13:34:01] Peer audio RTP is at port RFC-1918 IP:49152 [Sep 20 13:34:01] <--- Transmitting (NAT) to RFC-1918 IP:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP RFC-1918 IP:5060;branch=z9hG4bKmcfqve41ephc6d99m5g9caq;received=RFC-1918 IP;rport=5060 From: ;tag=772oho6nv15nt6k7tu8n43b2of05phpc To: "6005" ;tag=as7848f402 Call-ID: 1ae96a4f338a933964f3a9317833cf25@gofferje.homelinux.org CSeq: 2 INVITE User-Agent: Asterisk PBX 1.6.0-rc6 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: Content-Length: 0 <------------> [Sep 20 13:34:01] Audio is at RFC-1918 IP port 12428 [Sep 20 13:34:01] Adding codec 0x4 (ulaw) to SDP [Sep 20 13:34:01] Adding non-codec 0x1 (telephone-event) to SDP [Sep 20 13:34:01] <--- Reliably Transmitting (NAT) to RFC-1918 IP:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP RFC-1918 IP:5060;branch=z9hG4bKmcfqve41ephc6d99m5g9caq;received=RFC-1918 IP;rport=5060 From: ;tag=772oho6nv15nt6k7tu8n43b2of05phpc To: "6005" ;tag=as7848f402 Call-ID: 1ae96a4f338a933964f3a9317833cf25@gofferje.homelinux.org CSeq: 2 INVITE User-Agent: Asterisk PBX 1.6.0-rc6 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 267 v=0 o=root 444750411 444750413 IN IP4 RFC-1918 IP s=Asterisk PBX 1.6.0-rc6 c=IN IP4 RFC-1918 IP t=0 0 m=audio 12428 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Sep 20 13:34:01] -- Stopped music on hold on SIP/6005-08267e98 [Sep 20 13:34:01] Retransmitting #1 (NAT) to RFC-1918 IP:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP RFC-1918 IP:5060;branch=z9hG4bKmcfqve41ephc6d99m5g9caq;received=RFC-1918 IP;rport=5060 From: ;tag=772oho6nv15nt6k7tu8n43b2of05phpc To: "6005" ;tag=as7848f402 Call-ID: 1ae96a4f338a933964f3a9317833cf25@gofferje.homelinux.org CSeq: 2 INVITE User-Agent: Asterisk PBX 1.6.0-rc6 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 267 v=0 o=root 444750411 444750413 IN IP4 RFC-1918 IP s=Asterisk PBX 1.6.0-rc6 c=IN IP4 RFC-1918 IP t=0 0 m=audio 12428 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Sep 20 13:34:01] <--- SIP read from UDP://RFC-1918 IP:5060 ---> ACK sip:6005@RFC-1918 IP SIP/2.0 Via: SIP/2.0/UDP RFC-1918 IP:5060;branch=z9hG4bKj269o0o64tkerap0hqk3a03;rport To: "6005" ;tag=as7848f402 From: ;tag=772oho6nv15nt6k7tu8n43b2of05phpc Call-ID: 1ae96a4f338a933964f3a9317833cf25@gofferje.homelinux.org CSeq: 2 ACK Supported: sec-agree Max-Forwards: 70 Content-Length: 0 <-------------> [Sep 20 13:34:01] --- (9 headers 0 lines) --- [Sep 20 13:34:01] <--- SIP read from UDP://RFC-1918 IP:5060 ---> ACK sip:6005@RFC-1918 IP SIP/2.0 Via: SIP/2.0/UDP RFC-1918 IP:5060;branch=z9hG4bKj269o0o64tkerap0hqk3a03;rport To: "6005" ;tag=as7848f402 From: ;tag=772oho6nv15nt6k7tu8n43b2of05phpc Call-ID: 1ae96a4f338a933964f3a9317833cf25@gofferje.homelinux.org CSeq: 2 ACK Supported: sec-agree Max-Forwards: 70 Content-Length: 0 <-------------> [Sep 20 13:34:01] --- (9 headers 0 lines) --- [Sep 20 13:34:05] Reliably Transmitting (no NAT) to RFC-1918 IP:35044: OPTIONS sip:6005@RFC-1918 IP:35044;rinstance=4f1e7b5da9867330 SIP/2.0 Via: SIP/2.0/UDP RFC-1918 IP:5060;branch=z9hG4bK0ff02aa7;rport Max-Forwards: 70 From: "asterisk" ;tag=as1f5e7a62 To: Contact: Call-ID: 2a1bc07f186557251db08ab37f562eb8@RFC-1918 IP CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.0-rc6 Date: Sat, 20 Sep 2008 10:34:05 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Length: 0 --- [Sep 20 13:34:05] <--- SIP read from UDP://RFC-1918 IP:35044 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP RFC-1918 IP:5060;branch=z9hG4bK0ff02aa7;rport=5060 Contact: To: ;tag=8634f406 From: "asterisk";tag=as1f5e7a62 Call-ID: 2a1bc07f186557251db08ab37f562eb8@RFC-1918 IP CSeq: 102 OPTIONS Accept: application/sdp Accept-Language: en Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: X-Lite release 1100l stamp 47546 Content-Length: 0 <-------------> [Sep 20 13:34:05] --- (12 headers 0 lines) --- [Sep 20 13:34:05] Really destroying SIP dialog '2a1bc07f186557251db08ab37f562eb8@RFC-1918 IP' Method: OPTIONS [Sep 20 13:34:05] <--- SIP read from UDP://RFC-1918 IP:35044 ---> <-------------> [Sep 20 13:34:07] <--- SIP read from UDP://RFC-1918 IP:5060 ---> BYE sip:6005@RFC-1918 IP SIP/2.0 Via: SIP/2.0/UDP RFC-1918 IP:5060;branch=z9hG4bK4ivmjm0oa5hc707l9tkagb3;rport To: "6005" ;tag=as7848f402 From: ;tag=772oho6nv15nt6k7tu8n43b2of05phpc Call-ID: 1ae96a4f338a933964f3a9317833cf25@gofferje.homelinux.org CSeq: 3 BYE Max-Forwards: 70 Content-Length: 0 <-------------> [Sep 20 13:34:07] --- (8 headers 0 lines) --- [Sep 20 13:34:07] Sending to RFC-1918 IP : 5060 (NAT) [Sep 20 13:34:07] <--- Transmitting (NAT) to RFC-1918 IP:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP RFC-1918 IP:5060;branch=z9hG4bK4ivmjm0oa5hc707l9tkagb3;received=RFC-1918 IP;rport=5060 From: ;tag=772oho6nv15nt6k7tu8n43b2of05phpc To: "6005" ;tag=as7848f402 Call-ID: 1ae96a4f338a933964f3a9317833cf25@gofferje.homelinux.org CSeq: 3 BYE User-Agent: Asterisk PBX 1.6.0-rc6 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: Content-Length: 0 <------------> [Sep 20 13:34:07] == Spawn extension (macro-personcall, s, 3) exited non-zero on 'SIP/6005-08267e98' in macro 'personcall' [Sep 20 13:34:07] == Spawn extension (macro-personcall, s, 3) exited non-zero on 'SIP/6005-08267e98' [Sep 20 13:34:07] Scheduling destruction of SIP dialog 'MmMyMWM5Zjg2MDYzZWEzNTdjZjcwY2ZjOTUyNDAyYjA.' in 32000 ms (Method: ACK) [Sep 20 13:34:07] set_destination: Parsing for address/port to send to [Sep 20 13:34:07] set_destination: set destination to RFC-1918 IP, port 35044 [Sep 20 13:34:07] Reliably Transmitting (no NAT) to RFC-1918 IP:35044: BYE sip:6005@RFC-1918 IP:35044 SIP/2.0 Via: SIP/2.0/UDP RFC-1918 IP:5060;branch=z9hG4bK362d378f;rport Max-Forwards: 70 From: "6100";tag=as2925a42c To: "Test";tag=bd1ce66e Call-ID: MmMyMWM5Zjg2MDYzZWEzNTdjZjcwY2ZjOTUyNDAyYjA. CSeq: 102 BYE User-Agent: Asterisk PBX 1.6.0-rc6 Content-Length: 0 --- [Sep 20 13:34:07] <--- SIP read from UDP://RFC-1918 IP:35044 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP RFC-1918 IP:5060;branch=z9hG4bK362d378f;rport=5060 Contact: To: "Test";tag=bd1ce66e From: "6100";tag=as2925a42c Call-ID: MmMyMWM5Zjg2MDYzZWEzNTdjZjcwY2ZjOTUyNDAyYjA. CSeq: 102 BYE User-Agent: X-Lite release 1100l stamp 47546 Content-Length: 0 <-------------> [Sep 20 13:34:07] --- (9 headers 0 lines) --- [Sep 20 13:34:07] SIP Response message for INCOMING dialog BYE arrived [Sep 20 13:34:07] Really destroying SIP dialog '1ae96a4f338a933964f3a9317833cf25@gofferje.homelinux.org' Method: BYE [Sep 20 13:34:07] Really destroying SIP dialog 'MmMyMWM5Zjg2MDYzZWEzNTdjZjcwY2ZjOTUyNDAyYjA.' Method: ACK k-tanco*CLI> sip set debug off SIP Debugging Disabled