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 [Sep 20 13:36:16] -- Remote UNIX connection k-tanco*CLI> sip set debug on SIP Debugging enabled [Sep 20 13:36:32] <--- SIP read from UDP://RFC-1918 IP:5060 ---> INVITE sip:6005@gofferje.homelinux.org;user=phone SIP/2.0 Route: Via: SIP/2.0/UDP RFC-1918 IP:5060;branch=z9hG4bKqkgea69a3phc788n35esrjb;rport From: ;tag=0q8c82cud1hc7vhl35es To: Contact: Supported: 100rel,sec-agree CSeq: 784 INVITE Call-ID: UQIaWcjaoIeaDNBtgST1aZyjz2kbzO Allow: INVITE,ACK,BYE,CANCEL,REFER,NOTIFY,OPTIONS,PRACK Expires: 120 Privacy: none User-Agent: Nokia RM-244 200.34.36 P-Preferred-Identity: sip:sgofferj@gofferje.homelinux.org Max-Forwards: 70 Content-Type: application/sdp Accept: application/sdp Content-Length: 445 v=0 o=Nokia-SIPUA 63390173789561875 63390173789561875 IN IP4 RFC-1918 IP s=- c=IN IP4 RFC-1918 IP t=0 0 m=audio 49152 RTP/AVP 96 0 8 97 18 98 13 a=sendrecv a=ptime:20 a=maxptime:200 a=fmtp:96 mode-change-neighbor=1 a=fmtp:18 annexb=no a=fmtp:98 0-15 a=rtpmap:96 AMR/8000/1 a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:97 iLBC/8000/1 a=rtpmap:18 G729/8000/1 a=rtpmap:98 telephone-event/8000/1 a=rtpmap:13 CN/8000/1 <-------------> [Sep 20 13:36:32] --- (18 headers 19 lines) --- [Sep 20 13:36:32] == Using SIP RTP CoS mark 5 [Sep 20 13:36:32] == Using SIP VRTP CoS mark 6 [Sep 20 13:36:32] Sending to RFC-1918 IP : 5060 (NAT) [Sep 20 13:36:32] Using INVITE request as basis request - UQIaWcjaoIeaDNBtgST1aZyjz2kbzO [Sep 20 13:36:32] Found user 'sgofferj' for 'sgofferj' [Sep 20 13:36:32] <--- Reliably Transmitting (no NAT) to RFC-1918 IP:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP RFC-1918 IP:5060;branch=z9hG4bKqkgea69a3phc788n35esrjb;received=RFC-1918 IP;rport=5060 From: ;tag=0q8c82cud1hc7vhl35es To: ;tag=as1f27518f Call-ID: UQIaWcjaoIeaDNBtgST1aZyjz2kbzO CSeq: 784 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="1071a18c" Content-Length: 0 <------------> [Sep 20 13:36:32] Scheduling destruction of SIP dialog 'UQIaWcjaoIeaDNBtgST1aZyjz2kbzO' in 32000 ms (Method: INVITE) [Sep 20 13:36:32] <--- SIP read from UDP://RFC-1918 IP:5060 ---> ACK sip:6005@gofferje.homelinux.org;user=phone SIP/2.0 Via: SIP/2.0/UDP RFC-1918 IP:5060;branch=z9hG4bKqkgea69a3phc788n35esrjb;rport Route: From: ;tag=0q8c82cud1hc7vhl35es To: ;tag=as1f27518f Call-ID: UQIaWcjaoIeaDNBtgST1aZyjz2kbzO CSeq: 784 ACK Supported: sec-agree Max-Forwards: 70 Content-Length: 0 <-------------> [Sep 20 13:36:32] --- (10 headers 0 lines) --- [Sep 20 13:36:32] <--- SIP read from UDP://RFC-1918 IP:5060 ---> INVITE sip:6005@gofferje.homelinux.org;user=phone SIP/2.0 Route: Via: SIP/2.0/UDP RFC-1918 IP:5060;branch=z9hG4bK00sg2oc7v7rk40bk77945sr;rport From: ;tag=0q8c82cud1hc7vhl35es To: Contact: Supported: 100rel,sec-agree CSeq: 785 INVITE Call-ID: UQIaWcjaoIeaDNBtgST1aZyjz2kbzO Allow: INVITE,ACK,BYE,CANCEL,REFER,NOTIFY,OPTIONS,PRACK Expires: 120 Privacy: none User-Agent: Nokia RM-244 200.34.36 P-Preferred-Identity: sip:sgofferj@gofferje.homelinux.org Max-Forwards: 70 Authorization: Digest realm="asterisk",nonce="1071a18c",algorithm=MD5,username="sgofferj",uri="sip:6005@gofferje.homelinux.org;user=phone",response="392438db3e998da9f9ee3a1ab0777b04" Content-Type: application/sdp Accept: application/sdp Content-Length: 445 v=0 o=Nokia-SIPUA 63390173789561875 63390173789561875 IN IP4 RFC-1918 IP s=- c=IN IP4 RFC-1918 IP t=0 0 m=audio 49152 RTP/AVP 96 0 8 97 18 98 13 a=sendrecv a=ptime:20 a=maxptime:200 a=fmtp:96 mode-change-neighbor=1 a=fmtp:18 annexb=no a=fmtp:98 0-15 a=rtpmap:96 AMR/8000/1 a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:97 iLBC/8000/1 a=rtpmap:18 G729/8000/1 a=rtpmap:98 telephone-event/8000/1 a=rtpmap:13 CN/8000/1 <-------------> [Sep 20 13:36:32] --- (19 headers 19 lines) --- [Sep 20 13:36:32] Sending to RFC-1918 IP : 5060 (NAT) [Sep 20 13:36:32] Using INVITE request as basis request - UQIaWcjaoIeaDNBtgST1aZyjz2kbzO [Sep 20 13:36:32] Found user 'sgofferj' for 'sgofferj' [Sep 20 13:36:32] Found RTP audio format 96 [Sep 20 13:36:32] Found RTP audio format 0 [Sep 20 13:36:32] Found RTP audio format 8 [Sep 20 13:36:32] Found RTP audio format 97 [Sep 20 13:36:32] Found RTP audio format 18 [Sep 20 13:36:32] Found RTP audio format 98 [Sep 20 13:36:32] Found RTP audio format 13 [Sep 20 13:36:32] Peer audio RTP is at port RFC-1918 IP:49152 [Sep 20 13:36:32] Got unsupported a:fmtp in SDP offer [Sep 20 13:36:32] Got unsupported a:fmtp in SDP offer [Sep 20 13:36:32] Got unsupported a:fmtp in SDP offer [Sep 20 13:36:32] Found unknown media description format AMR for ID 96 [Sep 20 13:36:32] Found audio description format PCMU for ID 0 [Sep 20 13:36:32] Found audio description format PCMA for ID 8 [Sep 20 13:36:32] Found audio description format iLBC for ID 97 [Sep 20 13:36:32] Found audio description format G729 for ID 18 [Sep 20 13:36:32] Found audio description format telephone-event for ID 98 [Sep 20 13:36:32] Found audio description format CN for ID 13 [Sep 20 13:36:32] Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x50c (ulaw|alaw|g729|ilbc)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) [Sep 20 13:36:32] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x3 (telephone-event|CN), combined - 0x1 (telephone-event) [Sep 20 13:36:32] Peer audio RTP is at port RFC-1918 IP:49152 [Sep 20 13:36:32] Looking for 6005 in client_int_firma (domain gofferje.homelinux.org) [Sep 20 13:36:32] list_route: hop: [Sep 20 13:36:32] <--- Transmitting (no NAT) to RFC-1918 IP:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP RFC-1918 IP:5060;branch=z9hG4bK00sg2oc7v7rk40bk77945sr;received=RFC-1918 IP;rport=5060 From: ;tag=0q8c82cud1hc7vhl35es To: Call-ID: UQIaWcjaoIeaDNBtgST1aZyjz2kbzO CSeq: 785 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 13:36:32] -- Executing [6005@client_int_firma:1] Macro("SIP/sgofferj-08267e98", "internalcall,6005") in new stack [Sep 20 13:36:32] -- Executing [s@macro-internalcall:1] GotoIf("SIP/sgofferj-08267e98", "?:3") in new stack [Sep 20 13:36:32] -- Goto (macro-internalcall,s,3) [Sep 20 13:36:32] -- Executing [s@macro-internalcall:3] Dial("SIP/sgofferj-08267e98", "SIP/6005/ringer=inside,30") in new stack [Sep 20 13:36:32] == Using SIP RTP CoS mark 5 [Sep 20 13:36:32] == Using SIP VRTP CoS mark 6 [Sep 20 13:36:32] Audio is at RFC-1918 IP port 15782 [Sep 20 13:36:32] Adding codec 0x4 (ulaw) to SDP [Sep 20 13:36:32] Adding codec 0x8 (alaw) to SDP [Sep 20 13:36:32] Adding codec 0x2 (gsm) to SDP [Sep 20 13:36:32] Adding non-codec 0x1 (telephone-event) to SDP [Sep 20 13:36:32] Reliably Transmitting (no NAT) to RFC-1918 IP:35044: INVITE sip:ringer=inside@RFC-1918 IP:35044 SIP/2.0 Via: SIP/2.0/UDP RFC-1918 IP:5060;branch=z9hG4bK2419caf5;rport Max-Forwards: 70 From: "Stefan Mobile" ;tag=as1217b2c1 To: Contact: Call-ID: 755a2db507576e1e6bb686ce1597098f@RFC-1918 IP CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.0-rc6 Date: Sat, 20 Sep 2008 10:36:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Type: application/sdp Content-Length: 316 v=0 o=root 1666059062 1666059062 IN IP4 RFC-1918 IP s=Asterisk PBX 1.6.0-rc6 c=IN IP4 RFC-1918 IP t=0 0 m=audio 15782 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 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 13:36:32] -- Called 6005/ringer=inside [Sep 20 13:36:32] <--- SIP read from UDP://RFC-1918 IP:35044 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP RFC-1918 IP:5060;branch=z9hG4bK2419caf5;rport=5060 Contact: To: ;tag=f943f318 From: "Stefan Mobile";tag=as1217b2c1 Call-ID: 755a2db507576e1e6bb686ce1597098f@RFC-1918 IP CSeq: 102 INVITE User-Agent: X-Lite release 1100l stamp 47546 Content-Length: 0 <-------------> [Sep 20 13:36:32] --- (9 headers 0 lines) --- [Sep 20 13:36:32] -- SIP/6005-08308760 is ringing [Sep 20 13:36:32] <--- Transmitting (no NAT) to RFC-1918 IP:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP RFC-1918 IP:5060;branch=z9hG4bK00sg2oc7v7rk40bk77945sr;received=RFC-1918 IP;rport=5060 From: ;tag=0q8c82cud1hc7vhl35es To: ;tag=as1e386941 Call-ID: UQIaWcjaoIeaDNBtgST1aZyjz2kbzO CSeq: 785 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 13:36:34] 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=z9hG4bK3e7b7546;rport Max-Forwards: 70 From: "asterisk" ;tag=as2a4e56bc To: Contact: Call-ID: 450c9090321f51427055f08c79f6bf3a@RFC-1918 IP CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.0-rc6 Date: Sat, 20 Sep 2008 10:36:34 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Length: 0 --- [Sep 20 13:36:34] <--- SIP read from UDP://RFC-1918 IP:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP RFC-1918 IP:5060;branch=z9hG4bK3e7b7546;rport=5060;received=RFC-1918 IP To: ;tag=uf0llonmglhc7vji07qp From: "asterisk" ;tag=as2a4e56bc Call-ID: 450c9090321f51427055f08c79f6bf3a@RFC-1918 IP CSeq: 102 OPTIONS Content-Length: 0 <-------------> [Sep 20 13:36:34] --- (7 headers 0 lines) --- [Sep 20 13:36:34] Really destroying SIP dialog '450c9090321f51427055f08c79f6bf3a@RFC-1918 IP' Method: OPTIONS [Sep 20 13:36:35] <--- SIP read from UDP://RFC-1918 IP:35044 ---> <-------------> [Sep 20 13:36:37] <--- SIP read from UDP://RFC-1918 IP:35044 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP RFC-1918 IP:5060;branch=z9hG4bK2419caf5;rport=5060 Contact: To: ;tag=f943f318 From: "Stefan Mobile";tag=as1217b2c1 Call-ID: 755a2db507576e1e6bb686ce1597098f@RFC-1918 IP CSeq: 102 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: 183 v=0 o=- 9 2 IN IP4 RFC-1918 IP s=CounterPath X-Lite 3.0 c=IN IP4 RFC-1918 IP t=0 0 m=audio 10910 RTP/AVP 3 101 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv <-------------> [Sep 20 13:36:37] --- (11 headers 9 lines) --- [Sep 20 13:36:37] Found RTP audio format 3 [Sep 20 13:36:37] Found RTP audio format 101 [Sep 20 13:36:37] Peer audio RTP is at port RFC-1918 IP:10910 [Sep 20 13:36:37] Got unsupported a:fmtp in SDP offer [Sep 20 13:36:37] Found audio description format telephone-event for ID 101 [Sep 20 13:36:37] Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x2 (gsm)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x2 (gsm) [Sep 20 13:36:37] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Sep 20 13:36:37] Peer audio RTP is at port RFC-1918 IP:10910 [Sep 20 13:36:37] list_route: hop: [Sep 20 13:36:37] set_destination: Parsing for address/port to send to [Sep 20 13:36:37] set_destination: set destination to RFC-1918 IP, port 35044 [Sep 20 13:36:37] Transmitting (no NAT) to RFC-1918 IP:35044: ACK sip:ringer=inside@RFC-1918 IP:35044 SIP/2.0 Via: SIP/2.0/UDP RFC-1918 IP:5060;branch=z9hG4bK5eb3e3ba;rport Max-Forwards: 70 From: "Stefan Mobile" ;tag=as1217b2c1 To: ;tag=f943f318 Contact: Call-ID: 755a2db507576e1e6bb686ce1597098f@RFC-1918 IP CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.0-rc6 Content-Length: 0 --- [Sep 20 13:36:37] -- SIP/6005-08308760 answered SIP/sgofferj-08267e98 [Sep 20 13:36:37] Audio is at RFC-1918 IP port 18968 [Sep 20 13:36:37] Adding codec 0x4 (ulaw) to SDP [Sep 20 13:36:37] Adding codec 0x8 (alaw) to SDP [Sep 20 13:36:37] Adding non-codec 0x1 (telephone-event) to SDP [Sep 20 13:36:37] <--- Reliably Transmitting (no NAT) to RFC-1918 IP:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP RFC-1918 IP:5060;branch=z9hG4bK00sg2oc7v7rk40bk77945sr;received=RFC-1918 IP;rport=5060 From: ;tag=0q8c82cud1hc7vhl35es To: ;tag=as1e386941 Call-ID: UQIaWcjaoIeaDNBtgST1aZyjz2kbzO CSeq: 785 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: 288 v=0 o=root 262554901 262554901 IN IP4 RFC-1918 IP s=Asterisk PBX 1.6.0-rc6 c=IN IP4 RFC-1918 IP t=0 0 m=audio 18968 RTP/AVP 0 8 98 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:98 telephone-event/8000 a=fmtp:98 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Sep 20 13:36:38] <--- 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=z9hG4bKucau000b9d1j80bkffpdnl3;rport To: ;tag=as1e386941 From: ;tag=0q8c82cud1hc7vhl35es Call-ID: UQIaWcjaoIeaDNBtgST1aZyjz2kbzO CSeq: 785 ACK Supported: sec-agree Max-Forwards: 70 Authorization: Digest realm="asterisk",nonce="1071a18c",algorithm=MD5,username="sgofferj",uri="sip:6005@gofferje.homelinux.org;user=phone",response="392438db3e998da9f9ee3a1ab0777b04" Content-Length: 0 <-------------> [Sep 20 13:36:38] --- (10 headers 0 lines) --- [Sep 20 13:36:43] <--- 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=z9hG4bKumh9p0jfk9hc7ii4jmd5bbl;rport To: ;tag=as1e386941 From: ;tag=0q8c82cud1hc7vhl35es Contact: Call-ID: UQIaWcjaoIeaDNBtgST1aZyjz2kbzO CSeq: 786 INVITE Allow: INVITE,ACK,BYE,CANCEL,REFER,NOTIFY,OPTIONS,PRACK Expires: 10 Privacy: none User-Agent: Nokia RM-244 200.34.36 Max-Forwards: 70 Content-Type: application/sdp Accept: application/sdp Content-Length: 255 v=0 o=Nokia-SIPUA 63390173789561875 63390173789561876 IN IP4 RFC-1918 IP s=- c=IN IP4 RFC-1918 IP t=0 0 m=audio 49152 RTP/AVP 0 98 a=sendonly a=ptime:20 a=maxptime:200 a=fmtp:98 0-15 a=rtpmap:0 PCMU/8000/1 a=rtpmap:98 telephone-event/8000/1 <-------------> [Sep 20 13:36:43] --- (15 headers 12 lines) --- [Sep 20 13:36:43] Sending to RFC-1918 IP : 5060 (NAT) [Sep 20 13:36:43] <--- Transmitting (NAT) to RFC-1918 IP:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP RFC-1918 IP:5060;branch=z9hG4bKumh9p0jfk9hc7ii4jmd5bbl;received=RFC-1918 IP;rport=5060 From: ;tag=0q8c82cud1hc7vhl35es To: ;tag=as1e386941 Call-ID: UQIaWcjaoIeaDNBtgST1aZyjz2kbzO CSeq: 786 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 13:36:43] Audio is at RFC-1918 IP port 18968 [Sep 20 13:36:43] Adding codec 0x4 (ulaw) to SDP [Sep 20 13:36:43] Adding codec 0x8 (alaw) to SDP [Sep 20 13:36:43] Adding non-codec 0x1 (telephone-event) to SDP [Sep 20 13:36:43] <--- Reliably Transmitting (NAT) to RFC-1918 IP:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP RFC-1918 IP:5060;branch=z9hG4bKumh9p0jfk9hc7ii4jmd5bbl;received=RFC-1918 IP;rport=5060 From: ;tag=0q8c82cud1hc7vhl35es To: ;tag=as1e386941 Call-ID: UQIaWcjaoIeaDNBtgST1aZyjz2kbzO CSeq: 786 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: 288 v=0 o=root 262554901 262554901 IN IP4 RFC-1918 IP s=Asterisk PBX 1.6.0-rc6 c=IN IP4 RFC-1918 IP t=0 0 m=audio 18968 RTP/AVP 0 8 98 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:98 telephone-event/8000 a=fmtp:98 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Sep 20 13:36:43] <--- 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=z9hG4bKrths78luhifqvap0nm9b6eb;rport To: ;tag=as1e386941 From: ;tag=0q8c82cud1hc7vhl35es Call-ID: UQIaWcjaoIeaDNBtgST1aZyjz2kbzO CSeq: 786 ACK Supported: sec-agree Max-Forwards: 70 Content-Length: 0 <-------------> [Sep 20 13:36:43] --- (9 headers 0 lines) --- [Sep 20 13:36:43] <--- 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=z9hG4bKdc8tt8g79thc6o5v8flj8n9;rport To: ;tag=as1e386941 From: ;tag=0q8c82cud1hc7vhl35es Contact: Supported: 100rel Call-ID: UQIaWcjaoIeaDNBtgST1aZyjz2kbzO CSeq: 787 INVITE Allow: INVITE,ACK,BYE,CANCEL,REFER,NOTIFY,OPTIONS,PRACK Expires: 120 Max-Forwards: 70 Content-Type: application/sdp Accept: application/sdp Content-Length: 267 v=0 o=Nokia-SIPUA 63390173789561875 63390173789561877 IN IP4 RFC-1918 IP s=- c=IN IP4 RFC-1918 IP t=0 0 a=sendonly m=audio 49152 RTP/AVP 0 98 a=sendrecv a=ptime:20 a=maxptime:200 a=fmtp:98 0-15 a=rtpmap:0 PCMU/8000/1 a=rtpmap:98 telephone-event/8000/1 <-------------> [Sep 20 13:36:43] --- (14 headers 13 lines) --- [Sep 20 13:36:43] Sending to RFC-1918 IP : 5060 (NAT) [Sep 20 13:36:43] <--- Transmitting (NAT) to RFC-1918 IP:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP RFC-1918 IP:5060;branch=z9hG4bKdc8tt8g79thc6o5v8flj8n9;received=RFC-1918 IP;rport=5060 From: ;tag=0q8c82cud1hc7vhl35es To: ;tag=as1e386941 Call-ID: UQIaWcjaoIeaDNBtgST1aZyjz2kbzO CSeq: 787 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 13:36:43] Audio is at RFC-1918 IP port 18968 [Sep 20 13:36:43] Adding codec 0x4 (ulaw) to SDP [Sep 20 13:36:43] Adding codec 0x8 (alaw) to SDP [Sep 20 13:36:43] Adding non-codec 0x1 (telephone-event) to SDP [Sep 20 13:36:43] <--- Reliably Transmitting (NAT) to RFC-1918 IP:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP RFC-1918 IP:5060;branch=z9hG4bKdc8tt8g79thc6o5v8flj8n9;received=RFC-1918 IP;rport=5060 From: ;tag=0q8c82cud1hc7vhl35es To: ;tag=as1e386941 Call-ID: UQIaWcjaoIeaDNBtgST1aZyjz2kbzO CSeq: 787 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: 288 v=0 o=root 262554901 262554901 IN IP4 RFC-1918 IP s=Asterisk PBX 1.6.0-rc6 c=IN IP4 RFC-1918 IP t=0 0 m=audio 18968 RTP/AVP 0 8 98 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:98 telephone-event/8000 a=fmtp:98 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Sep 20 13:36:43] <--- 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=z9hG4bKocilqqo4qoltjap0f80kt03;rport To: ;tag=as1e386941 From: ;tag=0q8c82cud1hc7vhl35es Call-ID: UQIaWcjaoIeaDNBtgST1aZyjz2kbzO CSeq: 787 ACK Supported: sec-agree Max-Forwards: 70 Content-Length: 0 <-------------> [Sep 20 13:36:43] --- (9 headers 0 lines) --- [Sep 20 13:36:43] <--- 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=z9hG4bKc4d5m4fquphc6vdm5jm7osf;rport To: ;tag=as1e386941 From: ;tag=0q8c82cud1hc7vhl35es Contact: Supported: 100rel Call-ID: UQIaWcjaoIeaDNBtgST1aZyjz2kbzO CSeq: 788 INVITE Allow: INVITE,ACK,BYE,CANCEL,REFER,NOTIFY,OPTIONS,PRACK Expires: 120 Max-Forwards: 70 Content-Type: application/sdp Accept: application/sdp Content-Length: 250 v=0 o=Nokia-SIPUA 63390173789561875 63390173789561878 IN IP4 RFC-1918 IP s=- c=IN IP4 0.0.0.0 t=0 0 m=audio 49152 RTP/AVP 0 98 a=sendrecv a=ptime:20 a=maxptime:200 a=fmtp:98 0-15 a=rtpmap:0 PCMU/8000/1 a=rtpmap:98 telephone-event/8000/1 <-------------> [Sep 20 13:36:43] --- (14 headers 12 lines) --- [Sep 20 13:36:43] Sending to RFC-1918 IP : 5060 (NAT) [Sep 20 13:36:43] <--- Transmitting (NAT) to RFC-1918 IP:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP RFC-1918 IP:5060;branch=z9hG4bKc4d5m4fquphc6vdm5jm7osf;received=RFC-1918 IP;rport=5060 From: ;tag=0q8c82cud1hc7vhl35es To: ;tag=as1e386941 Call-ID: UQIaWcjaoIeaDNBtgST1aZyjz2kbzO CSeq: 788 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 13:36:43] Audio is at RFC-1918 IP port 18968 [Sep 20 13:36:43] Adding codec 0x4 (ulaw) to SDP [Sep 20 13:36:43] Adding codec 0x8 (alaw) to SDP [Sep 20 13:36:43] Adding non-codec 0x1 (telephone-event) to SDP [Sep 20 13:36:43] <--- Reliably Transmitting (NAT) to RFC-1918 IP:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP RFC-1918 IP:5060;branch=z9hG4bKc4d5m4fquphc6vdm5jm7osf;received=RFC-1918 IP;rport=5060 From: ;tag=0q8c82cud1hc7vhl35es To: ;tag=as1e386941 Call-ID: UQIaWcjaoIeaDNBtgST1aZyjz2kbzO CSeq: 788 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: 288 v=0 o=root 262554901 262554901 IN IP4 RFC-1918 IP s=Asterisk PBX 1.6.0-rc6 c=IN IP4 RFC-1918 IP t=0 0 m=audio 18968 RTP/AVP 0 8 98 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:98 telephone-event/8000 a=fmtp:98 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Sep 20 13:36:43] <--- 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=z9hG4bKf3oro2k8cg38hap070gicrb;rport To: ;tag=as1e386941 From: ;tag=0q8c82cud1hc7vhl35es Call-ID: UQIaWcjaoIeaDNBtgST1aZyjz2kbzO CSeq: 788 ACK Supported: sec-agree Max-Forwards: 70 Content-Length: 0 <-------------> [Sep 20 13:36:43] --- (9 headers 0 lines) --- [Sep 20 13:36:54] <--- 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=z9hG4bK54j1icci3dhc7utnrmirnl9;rport To: ;tag=as1e386941 From: ;tag=0q8c82cud1hc7vhl35es Contact: Call-ID: UQIaWcjaoIeaDNBtgST1aZyjz2kbzO CSeq: 789 INVITE Allow: INVITE,ACK,BYE,CANCEL,REFER,NOTIFY,OPTIONS,PRACK Expires: 10 Privacy: none User-Agent: Nokia RM-244 200.34.36 Max-Forwards: 70 Content-Type: application/sdp Accept: application/sdp Content-Length: 255 v=0 o=Nokia-SIPUA 63390173789561875 63390173789561879 IN IP4 RFC-1918 IP s=- c=IN IP4 RFC-1918 IP t=0 0 m=audio 49152 RTP/AVP 0 98 a=sendrecv a=ptime:20 a=maxptime:200 a=fmtp:98 0-15 a=rtpmap:0 PCMU/8000/1 a=rtpmap:98 telephone-event/8000/1 <-------------> [Sep 20 13:36:54] --- (15 headers 12 lines) --- [Sep 20 13:36:54] Sending to RFC-1918 IP : 5060 (NAT) [Sep 20 13:36:54] <--- Transmitting (NAT) to RFC-1918 IP:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP RFC-1918 IP:5060;branch=z9hG4bK54j1icci3dhc7utnrmirnl9;received=RFC-1918 IP;rport=5060 From: ;tag=0q8c82cud1hc7vhl35es To: ;tag=as1e386941 Call-ID: UQIaWcjaoIeaDNBtgST1aZyjz2kbzO CSeq: 789 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 13:36:54] Audio is at RFC-1918 IP port 18968 [Sep 20 13:36:54] Adding codec 0x4 (ulaw) to SDP [Sep 20 13:36:54] Adding codec 0x8 (alaw) to SDP [Sep 20 13:36:54] Adding non-codec 0x1 (telephone-event) to SDP [Sep 20 13:36:54] <--- Reliably Transmitting (NAT) to RFC-1918 IP:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP RFC-1918 IP:5060;branch=z9hG4bK54j1icci3dhc7utnrmirnl9;received=RFC-1918 IP;rport=5060 From: ;tag=0q8c82cud1hc7vhl35es To: ;tag=as1e386941 Call-ID: UQIaWcjaoIeaDNBtgST1aZyjz2kbzO CSeq: 789 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: 288 v=0 o=root 262554901 262554901 IN IP4 RFC-1918 IP s=Asterisk PBX 1.6.0-rc6 c=IN IP4 RFC-1918 IP t=0 0 m=audio 18968 RTP/AVP 0 8 98 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:98 telephone-event/8000 a=fmtp:98 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Sep 20 13:36:54] <--- 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=z9hG4bKs00pva1tkepftap0vp0ruij;rport To: ;tag=as1e386941 From: ;tag=0q8c82cud1hc7vhl35es Call-ID: UQIaWcjaoIeaDNBtgST1aZyjz2kbzO CSeq: 789 ACK Supported: sec-agree Max-Forwards: 70 Content-Length: 0 <-------------> [Sep 20 13:36:54] --- (9 headers 0 lines) --- [Sep 20 13:36:56] <--- 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=z9hG4bK7u4tuk8b7hhc6fshi8jq4pb;rport To: ;tag=as1e386941 From: ;tag=0q8c82cud1hc7vhl35es Call-ID: UQIaWcjaoIeaDNBtgST1aZyjz2kbzO CSeq: 790 BYE Max-Forwards: 70 Content-Length: 0 <-------------> [Sep 20 13:36:56] --- (8 headers 0 lines) --- [Sep 20 13:36:56] Sending to RFC-1918 IP : 5060 (NAT) [Sep 20 13:36:56] <--- Transmitting (NAT) to RFC-1918 IP:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP RFC-1918 IP:5060;branch=z9hG4bK7u4tuk8b7hhc6fshi8jq4pb;received=RFC-1918 IP;rport=5060 From: ;tag=0q8c82cud1hc7vhl35es To: ;tag=as1e386941 Call-ID: UQIaWcjaoIeaDNBtgST1aZyjz2kbzO CSeq: 790 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 13:36:56] Scheduling destruction of SIP dialog '755a2db507576e1e6bb686ce1597098f@RFC-1918 IP' in 6464 ms (Method: INVITE) [Sep 20 13:36:56] set_destination: Parsing for address/port to send to [Sep 20 13:36:56] set_destination: set destination to RFC-1918 IP, port 35044 [Sep 20 13:36:56] Reliably Transmitting (no NAT) to RFC-1918 IP:35044: BYE sip:ringer=inside@RFC-1918 IP:35044 SIP/2.0 Via: SIP/2.0/UDP RFC-1918 IP:5060;branch=z9hG4bK70392d8f;rport Max-Forwards: 70 From: "Stefan Mobile" ;tag=as1217b2c1 To: ;tag=f943f318 Call-ID: 755a2db507576e1e6bb686ce1597098f@RFC-1918 IP CSeq: 103 BYE User-Agent: Asterisk PBX 1.6.0-rc6 Content-Length: 0 --- [Sep 20 13:36:56] == Spawn extension (macro-internalcall, s, 3) exited non-zero on 'SIP/sgofferj-08267e98' in macro 'internalcall' [Sep 20 13:36:56] == Spawn extension (macro-internalcall, s, 3) exited non-zero on 'SIP/sgofferj-08267e98' [Sep 20 13:36:56] <--- SIP read from UDP://RFC-1918 IP:35044 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP RFC-1918 IP:5060;branch=z9hG4bK70392d8f;rport=5060 Contact: To: ;tag=f943f318 From: "Stefan Mobile";tag=as1217b2c1 Call-ID: 755a2db507576e1e6bb686ce1597098f@RFC-1918 IP CSeq: 103 BYE User-Agent: X-Lite release 1100l stamp 47546 Content-Length: 0 <-------------> [Sep 20 13:36:56] --- (9 headers 0 lines) --- [Sep 20 13:36:56] Really destroying SIP dialog '755a2db507576e1e6bb686ce1597098f@RFC-1918 IP' Method: INVITE [Sep 20 13:36:56] Really destroying SIP dialog 'UQIaWcjaoIeaDNBtgST1aZyjz2kbzO' Method: BYE k-tanco*CLI> sip set debug off SIP Debugging Disabled k-tanco*CLI> quit