*CLI> core set verbose 4 Verbosity was 0 and is now 4 *CLI> core set debug 4 Core debug was 0 and is now 4 *CLI> sip set debug on SIP Debugging enabled *CLI> <--- SIP read from UDP:10.244.43.101:5060 ---> INVITE sip:8998@10.244.43.41:5060 SIP/2.0 Via: SIP/2.0/UDP 10.244.43.101:5060;branch=z9hG4bK1bc16ee3f Max-Forwards: 70 Content-Length: 562 To: 8998 From: 8881 ;tag=0a1455bce64f38c Call-ID: 2bf116d8f128af7633880a8abaac1d87@10.244.43.101 CSeq: 1865296331 INVITE Supported: timer Allow-Events: talk,hold,conference Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO Content-Type: application/sdp Contact: 8881 Supported: replaces User-Agent: Aastra 9133i/1.4.3.1001 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45 v=0 o=MxSIP 0 309576335 IN IP4 10.244.43.101 s=SIP Call c=IN IP4 10.244.43.101 t=0 0 m=audio 3000 RTP/AVP 0 18 96 102 107 104 105 106 97 98 2 99 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:96 BV16/8000 a=rtpmap:102 BV32/16000 a=rtpmap:107 L16/16000 a=rtpmap:104 PCMU/16000 a=rtpmap:105 PCMA/16000 a=rtpmap:106 L16/8000 a=rtpmap:97 G726-16/8000 a=rtpmap:98 G726-24/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:99 G726-40/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=silenceSupp:on - - - - <-------------> --- (15 headers 23 lines) --- Sending to 10.244.43.101:5060 (no NAT) Using INVITE request as basis request - 2bf116d8f128af7633880a8abaac1d87@10.244.43.101 Found peer '8881' for '8881' from 10.244.43.101:5060 <--- Reliably Transmitting (NAT) to 10.244.43.101:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.244.43.101:5060;branch=z9hG4bK1bc16ee3f;received=10.244.43.101;rport=5060 From: 8881 ;tag=0a1455bce64f38c To: 8998 ;tag=as6e762578 Call-ID: 2bf116d8f128af7633880a8abaac1d87@10.244.43.101 CSeq: 1865296331 INVITE Server: Asterisk PBX 1.8.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1f2abefd" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '2bf116d8f128af7633880a8abaac1d87@10.244.43.101' in 6400 ms (Method: INVITE) <--- SIP read from UDP:10.244.43.101:5060 ---> ACK sip:8998@10.244.43.41:5060 SIP/2.0 Via: SIP/2.0/UDP 10.244.43.101:5060;branch=z9hG4bK1bc16ee3f Max-Forwards: 70 Content-Length: 0 To: 8998 ;tag=as6e762578 From: 8881 ;tag=0a1455bce64f38c Call-ID: 2bf116d8f128af7633880a8abaac1d87@10.244.43.101 CSeq: 1865296331 ACK User-Agent: Aastra 9133i/1.4.3.1001 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45 <-------------> --- (9 headers 0 lines) --- <--- SIP read from UDP:10.244.43.101:5060 ---> INVITE sip:8998@10.244.43.41:5060 SIP/2.0 Via: SIP/2.0/UDP 10.244.43.101:5060;branch=z9hG4bK89146fa81 Max-Forwards: 70 Content-Length: 562 To: 8998 From: 8881 ;tag=0a1455bce64f38c Call-ID: 2bf116d8f128af7633880a8abaac1d87@10.244.43.101 CSeq: 1865296332 INVITE Supported: timer Allow-Events: talk,hold,conference Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO Contact: 8881 Content-Type: application/sdp Supported: replaces Authorization:Digest response="36c908a8d0df3a2645315987167e2d9f",username="8881",realm="asterisk",nonce="1f2abefd",algorithm=MD5,uri="sip:8998@10.244.43.41:5060" User-Agent: Aastra 9133i/1.4.3.1001 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45 v=0 o=MxSIP 0 309576335 IN IP4 10.244.43.101 s=SIP Call c=IN IP4 10.244.43.101 t=0 0 m=audio 3000 RTP/AVP 0 18 96 102 107 104 105 106 97 98 2 99 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:96 BV16/8000 a=rtpmap:102 BV32/16000 a=rtpmap:107 L16/16000 a=rtpmap:104 PCMU/16000 a=rtpmap:105 PCMA/16000 a=rtpmap:106 L16/8000 a=rtpmap:97 G726-16/8000 a=rtpmap:98 G726-24/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:99 G726-40/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=silenceSupp:on - - - - <-------------> --- (16 headers 23 lines) --- Sending to 10.244.43.101:5060 (NAT) Using INVITE request as basis request - 2bf116d8f128af7633880a8abaac1d87@10.244.43.101 Found peer '8881' for '8881' from 10.244.43.101:5060 == Using SIP RTP CoS mark 5 Found RTP audio format 0 Found RTP audio format 18 Found RTP audio format 96 Found RTP audio format 102 Found RTP audio format 107 Found RTP audio format 104 Found RTP audio format 105 Found RTP audio format 106 Found RTP audio format 97 Found RTP audio format 98 Found RTP audio format 2 Found RTP audio format 99 Found RTP audio format 8 Found RTP audio format 101 Found audio description format PCMU for ID 0 Found audio description format G729 for ID 18 Found audio description format BV16 for ID 96 Found audio description format BV32 for ID 102 Found audio description format L16 for ID 107 Found audio description format PCMU for ID 104 Found audio description format PCMA for ID 105 Found audio description format L16 for ID 106 Found audio description format G726-16 for ID 97 Found audio description format G726-24 for ID 98 Found audio description format G726-32 for ID 2 Found audio description format G726-40 for ID 99 Found audio description format PCMA for ID 8 Found audio description format telephone-event for ID 101 Capabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0x470ad4c (ulaw|alaw|g726|slin|g729|ilbc|slin16|h263p|h264|mpeg4|red|siren7)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 10.244.43.101:3000 Looking for 8998 in de-interne (domain 10.244.43.41:5060) list_route: hop: <--- Transmitting (NAT) to 10.244.43.101:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.244.43.101:5060;branch=z9hG4bK89146fa81;received=10.244.43.101;rport=5060 From: 8881 ;tag=0a1455bce64f38c To: 8998 Call-ID: 2bf116d8f128af7633880a8abaac1d87@10.244.43.101 CSeq: 1865296332 INVITE Server: Asterisk PBX 1.8.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> -- Executing [8998@de-interne:1] Page("SIP/8881-00000001", "MulticastRTP/basic/225.3.15.13:32000") in new stack -- Called basic/225.3.15.13:32000 -- Playing 'beep.gsm' (language 'en') -- MulticastRTP/0xd55a730 answered -- Created MeetMe conference 1023 for conference '389850381d' Audio is at 5060 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (NAT) to 10.244.43.101:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.244.43.101:5060;branch=z9hG4bK89146fa81;received=10.244.43.101;rport=5060 From: 8881 ;tag=0a1455bce64f38c To: 8998 ;tag=as34dcff8f Call-ID: 2bf116d8f128af7633880a8abaac1d87@10.244.43.101 CSeq: 1865296332 INVITE Server: Asterisk PBX 1.8.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 258 v=0 o=root 1470846794 1470846794 IN IP4 10.244.43.41 s=Asterisk PBX 1.8.0 c=IN IP4 10.244.43.41 t=0 0 m=audio 15740 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <------------> Retransmitting #1 (NAT) to 10.244.43.101:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.244.43.101:5060;branch=z9hG4bK89146fa81;received=10.244.43.101;rport=5060 From: 8881 ;tag=0a1455bce64f38c To: 8998 ;tag=as34dcff8f Call-ID: 2bf116d8f128af7633880a8abaac1d87@10.244.43.101 CSeq: 1865296332 INVITE Server: Asterisk PBX 1.8.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 258 v=0 o=root 1470846794 1470846794 IN IP4 10.244.43.41 s=Asterisk PBX 1.8.0 c=IN IP4 10.244.43.41 t=0 0 m=audio 15740 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- <--- SIP read from UDP:10.244.43.101:5060 ---> ACK sip:8998@10.244.43.41:5060 SIP/2.0 Via: SIP/2.0/UDP 10.244.43.101:5060;branch=z9hG4bK45936c149 Max-Forwards: 70 Content-Length: 0 To: 8998 ;tag=as34dcff8f From: 8881 ;tag=0a1455bce64f38c Call-ID: 2bf116d8f128af7633880a8abaac1d87@10.244.43.101 CSeq: 1865296332 ACK Contact: 8881 Authorization:Digest response="36c908a8d0df3a2645315987167e2d9f",username="8881",realm="asterisk",nonce="1f2abefd",algorithm=MD5,uri="sip:8998@10.244.43.41:5060" User-Agent: Aastra 9133i/1.4.3.1001 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45 <-------------> --- (11 headers 0 lines) --- <--- SIP read from UDP:10.244.43.101:5060 ---> ACK sip:8998@10.244.43.41:5060 SIP/2.0 Via: SIP/2.0/UDP 10.244.43.101:5060;branch=z9hG4bK45936c149 Max-Forwards: 70 Content-Length: 0 To: 8998 ;tag=as34dcff8f From: 8881 ;tag=0a1455bce64f38c Call-ID: 2bf116d8f128af7633880a8abaac1d87@10.244.43.101 CSeq: 1865296332 ACK Contact: 8881 Authorization:Digest response="36c908a8d0df3a2645315987167e2d9f",username="8881",realm="asterisk",nonce="1f2abefd",algorithm=MD5,uri="sip:8998@10.244.43.41:5060" User-Agent: Aastra 9133i/1.4.3.1001 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45 <-------------> --- (11 headers 0 lines) --- <--- SIP read from UDP:10.244.43.101:5060 ---> BYE sip:8998@10.244.43.41:5060 SIP/2.0 Via: SIP/2.0/UDP 10.244.43.101:5060;branch=z9hG4bK1b34edf3d Max-Forwards: 70 Content-Length: 0 To: 8998 ;tag=as34dcff8f From: 8881 ;tag=0a1455bce64f38c Call-ID: 2bf116d8f128af7633880a8abaac1d87@10.244.43.101 CSeq: 1865296333 BYE Supported: timer Supported: replaces Authorization:Digest response="e262884bb324257d1cde3aed11928ca9",username="8881",realm="asterisk",nonce="1f2abefd",algorithm=MD5,uri="sip:8998@10.244.43.41:5060" User-Agent: Aastra 9133i/1.4.3.1001 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45 <-------------> --- (12 headers 0 lines) --- Sending to 10.244.43.101:5060 (NAT) Scheduling destruction of SIP dialog '2bf116d8f128af7633880a8abaac1d87@10.244.43.101' in 6400 ms (Method: BYE) <--- Transmitting (NAT) to 10.244.43.101:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.244.43.101:5060;branch=z9hG4bK1b34edf3d;received=10.244.43.101;rport=5060 From: 8881 ;tag=0a1455bce64f38c To: 8998 ;tag=as34dcff8f Call-ID: 2bf116d8f128af7633880a8abaac1d87@10.244.43.101 CSeq: 1865296333 BYE Server: Asterisk PBX 1.8.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> [Nov 5 16:54:55] NOTICE[29960]: chan_sip.c:4034 sip_setoption: Unknown option: 5 [Nov 5 16:54:55] NOTICE[29960]: chan_sip.c:4034 sip_setoption: Unknown option: 6 -- Hungup 'DAHDI/pseudo-17985877' == Spawn extension (de-interne, 8998, 1) exited non-zero on 'SIP/8881-00000001' Really destroying SIP dialog '2bf116d8f128af7633880a8abaac1d87@10.244.43.101' Method: BYE *CLI> sip set debug off SIP Debugging Disabled *CLI> core set verbose 0 Verbosity is now OFF *CLI> core set debug 0 Core debug is now OFF *CLI>