=~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2009.09.14 10:04:33 =~=~=~=~=~=~=~=~=~=~=~= asterisk ]0;root@asterisk:/usr/src/asterisk-1.6.1.5[root@asterisk asterisk-1.6.1.5]# asterisk -r Asterisk 1.6.1.5, Copyright (C) 1999 - 2009 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 1.6.1.5 currently running on asterisk (pid = 15489) asterisk*CLI> core set verbose 10 asterisk*CLI> Verbosity was 0 and is now 10 asterisk*CLI> -- Registered SIP '8678' at 10.9.5.107 port 10474 > Saved useragent "X-Lite release 1103k stamp 53621" for peer 8678 asterisk*CLI> == Using SIP RTP CoS mark 5 asterisk*CLI> -- Executing [3300@default:1] Answer("SIP/8678-0a1ab498", "") in new stack asterisk*CLI> -- Executing [3300@default:2] Dial("SIP/8678-0a1ab498", "SIP/SIP_VM/3300") in new stack == Using SIP RTP CoS mark 5 asterisk*CLI> -- Called SIP_VM/3300 asterisk*CLI> -- SIP/SIP_VM-0a1ae788 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Executing [3300@default:3] Busy("SIP/8678-0a1ab498", "") in new stack asterisk*CLI> == Spawn extension (default, 3300, 3) exited non-zero on 'SIP/8678-0a1ab498' asterisk*CLI> sip set debug on asterisk*CLI> SIP Debugging enabled asterisk*CLI> <--- SIP read from UDP://10.9.5.107:10474 ---> INVITE sip:3300@10.9.1.121 SIP/2.0 Via: SIP/2.0/UDP 10.9.5.107:10474;branch=z9hG4bK-d8754z-6a75d142056b8e48-1---d8754z-;rport Max-Forwards: 70 Contact: To: "3300" From: ;tag=704e5b3d Call-ID: NTI0NGI5ZTE2M2M4M2JkMDVhYWU4MzAyOGQ0OGIwMTY. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: X-Lite release 1103k stamp 53621 Content-Length: 258 v=0 o=- 4 2 IN IP4 10.9.5.107 s=CounterPath X-Lite 3.0 c=IN IP4 10.9.5.107 t=0 0 m=audio 15570 RTP/AVP 107 0 8 101 a=alt:1 1 : jyrMnfov D9UyReZ9 10.9.5.107 15570 a=fmtp:101 0-15 a=rtpmap:107 BV32/16000 a=rtpmap:101 telephone-event/8000 a=sendrecv <-------------> --- (12 headers 11 lines) --- == Using SIP RTP CoS mark 5 Sending to 10.9.5.107 : 10474 (no NAT) Using INVITE request as basis request - NTI0NGI5ZTE2M2M4M2JkMDVhYWU4MzAyOGQ0OGIwMTY. Found peer '8678' for '8678' from 10.9.5.107:10474 <--- Reliably Transmitting (no NAT) to 10.9.5.107:10474 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.9.5.107:10474;branch=z9hG4bK-d8754z-6a75d142056b8e48-1---d8754z-;received=10.9.5.107;rport=10474 From: ;tag=704e5b3d To: "3300";tag=as430fb700 Call-ID: NTI0NGI5ZTE2M2M4M2JkMDVhYWU4MzAyOGQ0OGIwMTY. CSeq: 1 INVITE Server: Asterisk PBX 1.6.1.5 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="07d56872" Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'NTI0NGI5ZTE2M2M4M2JkMDVhYWU4MzAyOGQ0OGIwMTY.' in 32000 ms (Method: INVITE) asterisk*CLI> <--- SIP read from UDP://10.9.5.107:10474 ---> ACK sip:3300@10.9.1.121 SIP/2.0 Via: SIP/2.0/UDP 10.9.5.107:10474;branch=z9hG4bK-d8754z-6a75d142056b8e48-1---d8754z-;rport To: "3300";tag=as430fb700 From: ;tag=704e5b3d Call-ID: NTI0NGI5ZTE2M2M4M2JkMDVhYWU4MzAyOGQ0OGIwMTY. CSeq: 1 ACK Content-Length: 0 <-------------> --- (7 headers 0 lines) --- asterisk*CLI> <--- SIP read from UDP://10.9.5.107:10474 ---> INVITE sip:3300@10.9.1.121 SIP/2.0 Via: SIP/2.0/UDP 10.9.5.107:10474;branch=z9hG4bK-d8754z-7d378406ba194700-1---d8754z-;rport Max-Forwards: 70 Contact: To: "3300" From: ;tag=704e5b3d Call-ID: NTI0NGI5ZTE2M2M4M2JkMDVhYWU4MzAyOGQ0OGIwMTY. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: X-Lite release 1103k stamp 53621 Authorization: Digest username="8678",realm="asterisk",nonce="07d56872",uri="sip:3300@10.9.1.121",response="61790fefbacc771a545fb7ebdf80ae6e",algorithm=MD5 Content-Length: 258 v=0 o=- 4 2 IN IP4 10.9.5.107 s=CounterPath X-Lite 3.0 c=IN IP4 10.9.5.107 t=0 0 m=audio 15570 RTP/AVP 107 0 8 101 a=alt:1 1 : jyrMnfov D9UyReZ9 10.9.5.107 15570 a=fmtp:101 0-15 a=rtpmap:107 BV32/16000 a=rtpmap:101 telephone-event/8000 a=sendrecv <-------------> --- (13 headers 11 lines) --- Sending to 10.9.5.107 : 10474 (no NAT) Using INVITE request as basis request - NTI0NGI5ZTE2M2M4M2JkMDVhYWU4MzAyOGQ0OGIwMTY. Found peer '8678' for '8678' from 10.9.5.107:10474 Found RTP audio format 107 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 10.9.5.107:15570 Found unknown media description format BV32 for ID 107 Found audio description format telephone-event for ID 101 Capabilities: us - 0x4 (ulaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.9.5.107:15570 Looking for 3300 in default (domain 10.9.1.121) list_route: hop: <--- Transmitting (no NAT) to 10.9.5.107:10474 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.9.5.107:10474;branch=z9hG4bK-d8754z-7d378406ba194700-1---d8754z-;received=10.9.5.107;rport=10474 From: ;tag=704e5b3d To: "3300" Call-ID: NTI0NGI5ZTE2M2M4M2JkMDVhYWU4MzAyOGQ0OGIwMTY. CSeq: 2 INVITE Server: Asterisk PBX 1.6.1.5 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Length: 0 <------------> asterisk*CLI> -- Executing [3300@default:1] Answer("SIP/8678-0a1a04c8", "") in new stack asterisk*CLI> Audio is at 10.9.1.121 port 12752 asterisk*CLI> Adding codec 0x4 (ulaw) to SDP asterisk*CLI> Adding non-codec 0x1 (telephone-event) to SDP asterisk*CLI> <--- Reliably Transmitting (no NAT) to 10.9.5.107:10474 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.9.5.107:10474;branch=z9hG4bK-d8754z-7d378406ba194700-1---d8754z-;received=10.9.5.107;rport=10474 From: ;tag=704e5b3d To: "3300";tag=as2e6c69ae Call-ID: NTI0NGI5ZTE2M2M4M2JkMDVhYWU4MzAyOGQ0OGIwMTY. CSeq: 2 INVITE Server: Asterisk PBX 1.6.1.5 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 259 v=0 o=root 1164266424 1164266424 IN IP4 10.9.1.121 s=Asterisk PBX 1.6.1.5 c=IN IP4 10.9.1.121 t=0 0 m=audio 12752 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 <------------> asterisk*CLI> -- Executing [3300@default:2] Dial("SIP/8678-0a1a04c8", "SIP/SIP_VM/3300") in new stack asterisk*CLI> == Using SIP RTP CoS mark 5 asterisk*CLI> Audio is at 10.9.1.121 port 10452 asterisk*CLI> Adding codec 0x4 (ulaw) to SDP asterisk*CLI> Adding codec 0x2 (gsm) to SDP asterisk*CLI> Adding codec 0x8 (alaw) to SDP asterisk*CLI> Adding non-codec 0x1 (telephone-event) to SDP asterisk*CLI> Reliably Transmitting (no NAT) to 10.9.1.13:5067: INVITE sip:3300@10.9.1.13:5067 SIP/2.0 Via: SIP/2.0/TCP 10.9.1.121:5060;branch=z9hG4bK67e12f58;rport Max-Forwards: 70 From: "Exchange User" ;tag=as2836cd22 To: Contact: Call-ID: 496e98ef565f1e3952b9009c72d0a6da@10.9.1.121 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.5 Date: Mon, 14 Sep 2009 14:07:09 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 306 v=0 o=root 1100115600 1100115600 IN IP4 10.9.1.121 s=Asterisk PBX 1.6.1.5 c=IN IP4 10.9.1.121 t=0 0 m=audio 10452 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/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 --- asterisk*CLI> -- Called SIP_VM/3300 asterisk*CLI> <--- SIP read from TCP://10.9.1.13:5067 ---> SIP/2.0 100 Trying FROM: "Exchange User";tag=as2836cd22 TO: CSEQ: 102 INVITE CALL-ID: 496e98ef565f1e3952b9009c72d0a6da@10.9.1.121 VIA: SIP/2.0/TCP 10.9.1.121:5060;branch=z9hG4bK67e12f58;rport CONTENT-LENGTH: 0 <-------------> --- (7 headers 0 lines) --- asterisk*CLI> <--- SIP read from TCP://10.9.1.13:5067 ---> SIP/2.0 180 Ringing FROM: "Exchange User";tag=as2836cd22 TO: ;epid=32BB1FF496;tag=7f3f84c250 CSEQ: 102 INVITE CALL-ID: 496e98ef565f1e3952b9009c72d0a6da@10.9.1.121 VIA: SIP/2.0/TCP 10.9.1.121:5060;branch=z9hG4bK67e12f58;rport CONTENT-LENGTH: 0 SERVER: RTCC/3.0.0.0 <-------------> --- (8 headers 0 lines) --- asterisk*CLI> <--- SIP read from UDP://10.9.5.107:10474 ---> ACK sip:3300@10.9.1.121 SIP/2.0 Via: SIP/2.0/UDP 10.9.5.107:10474;branch=z9hG4bK-d8754z-0d104453a737e845-1---d8754z-;rport Max-Forwards: 70 Contact: To: "3300";tag=as2e6c69ae From: ;tag=704e5b3d Call-ID: NTI0NGI5ZTE2M2M4M2JkMDVhYWU4MzAyOGQ0OGIwMTY. CSeq: 2 ACK User-Agent: X-Lite release 1103k stamp 53621 Authorization: Digest username="8678",realm="asterisk",nonce="07d56872",uri="sip:3300@10.9.1.121",response="61790fefbacc771a545fb7ebdf80ae6e",algorithm=MD5 Content-Length: 0 <-------------> --- (11 headers 0 lines) --- asterisk*CLI> <--- SIP read from TCP://10.9.1.13:5067 ---> SIP/2.0 200 OK FROM: "Exchange User";tag=as2836cd22 TO: ;epid=32BB1FF496;tag=7f3f84c250 CSEQ: 102 INVITE CALL-ID: 496e98ef565f1e3952b9009c72d0a6da@10.9.1.121 VIA: SIP/2.0/TCP 10.9.1.121:5060;branch=z9hG4bK67e12f58;rport CONTACT: ;automata CONTENT-LENGTH: 189 CONTENT-TYPE: application/sdp ALLOW: UPDATE SERVER: RTCC/3.0.0.0 ALLOW: Ack, Cancel, Bye,Invite,Message,Info,Service,Options,BeNotify v=0 o=- 0 0 IN IP4 10.9.1.13 s=Microsoft Exchange Speech Engine c=IN IP4 10.9.1.13 t=0 0 m=audio 30464 RTP/AVP 0 8 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 <-------------> --- (12 headers 9 lines) --- asterisk*CLI> <--- SIP read from TCP://10.9.1.13:42108 ---> OPTIONS sip:10.9.1.121:5060 SIP/2.0 FROM: ;epid=BCD2043730;tag=7df3ed778 TO: CSEQ: 12898 OPTIONS CALL-ID: c0a1d2a2f98c481f9eb12832fdd8cc40 MAX-FORWARDS: 70 VIA: SIP/2.0/TCP 10.9.1.13:42108;branch=z9hG4bK1a5659c5 ACCEPT: application/sdp CONTENT-LENGTH: 0 USER-AGENT: RTCC/3.0.0.0 <-------------> asterisk*CLI> --- (10 headers 0 lines) --- asterisk*CLI> Looking for s in default (domain 10.9.1.121) asterisk*CLI> <--- Transmitting (no NAT) to 10.9.1.13:42108 ---> SIP/2.0 404 Not Found Via: SIP/2.0/TCP 10.9.1.13:42108;branch=z9hG4bK1a5659c5;received=10.9.1.13 From: ;epid=BCD2043730;tag=7df3ed778 To: ;tag=as6c97d89d Call-ID: c0a1d2a2f98c481f9eb12832fdd8cc40 CSeq: 12898 OPTIONS Server: Asterisk PBX 1.6.1.5 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Accept: application/sdp Content-Length: 0 <------------> asterisk*CLI> Scheduling destruction of SIP dialog 'c0a1d2a2f98c481f9eb12832fdd8cc40' in 32000 ms (Method: OPTIONS) asterisk*CLI> <--- SIP read from UDP://10.9.5.107:10474 ---> <-------------> asterisk*CLI> Really destroying SIP dialog '6150895346bba5b6496d08401f908ea6@10.9.1.121' Method: BYE asterisk*CLI> -- SIP/SIP_VM-0a1ba1e0 is circuit-busy Scheduling destruction of SIP dialog '496e98ef565f1e3952b9009c72d0a6da@10.9.1.121' in 32000 ms (Method: INVITE) == Everyone is busy/congested at this time (1:0/1/0) -- Executing [3300@default:3] Busy("SIP/8678-0a1a04c8", "") in new stack asterisk*CLI> Really destroying SIP dialog 'c0a1d2a2f98c481f9eb12832fdd8cc40' Method: OPTIONS asterisk*CLI> <--- SIP read from TCP://10.9.1.13:5067 ---> BYE sip:8678@10.9.1.121;transport=TCP SIP/2.0 FROM: ;epid=32BB1FF496;tag=7f3f84c250 TO: ;tag=as2836cd22 CSEQ: 1 BYE CALL-ID: 496e98ef565f1e3952b9009c72d0a6da@10.9.1.121 MAX-FORWARDS: 70 VIA: SIP/2.0/TCP 10.9.1.13:5067;branch=z9hG4bKa3cae625 CONTENT-LENGTH: 0 USER-AGENT: RTCC/3.0.0.0 <-------------> --- (9 headers 0 lines) --- Sending to 10.9.1.13 : 5067 (no NAT) Scheduling destruction of SIP dialog '496e98ef565f1e3952b9009c72d0a6da@10.9.1.121' in 32000 ms (Method: BYE) <--- Transmitting (no NAT) to 10.9.1.13:5067 ---> SIP/2.0 200 OK Via: SIP/2.0/TCP 10.9.1.13:5067;branch=z9hG4bKa3cae625;received=10.9.1.13 From: ;epid=32BB1FF496;tag=7f3f84c250 To: ;tag=as2836cd22 Call-ID: 496e98ef565f1e3952b9009c72d0a6da@10.9.1.121 CSeq: 1 BYE Server: Asterisk PBX 1.6.1.5 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> asterisk*CLI> <--- SIP read from UDP://10.9.5.107:10474 ---> BYE sip:3300@10.9.1.121 SIP/2.0 Via: SIP/2.0/UDP 10.9.5.107:10474;branch=z9hG4bK-d8754z-6a0e6508c2324c01-1---d8754z-;rport Max-Forwards: 70 Contact: To: "3300";tag=as2e6c69ae From: ;tag=704e5b3d Call-ID: NTI0NGI5ZTE2M2M4M2JkMDVhYWU4MzAyOGQ0OGIwMTY. CSeq: 3 BYE User-Agent: X-Lite release 1103k stamp 53621 Authorization: Digest username="8678",realm="asterisk",nonce="07d56872",uri="sip:3300@10.9.1.121",response="ceb16608e4dbf60f54ddd9038a2f8a37",algorithm=MD5 Reason: SIP;description="User Hung Up" Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Sending to 10.9.5.107 : 10474 (no NAT) <--- Transmitting (no NAT) to 10.9.5.107:10474 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.9.5.107:10474;branch=z9hG4bK-d8754z-6a0e6508c2324c01-1---d8754z-;received=10.9.5.107;rport=10474 From: ;tag=704e5b3d To: "3300";tag=as2e6c69ae Call-ID: NTI0NGI5ZTE2M2M4M2JkMDVhYWU4MzAyOGQ0OGIwMTY. CSeq: 3 BYE Server: Asterisk PBX 1.6.1.5 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> == Spawn extension (default, 3300, 3) exited non-zero on 'SIP/8678-0a1a04c8' asterisk*CLI> <--- SIP read from UDP://10.9.5.107:10474 ---> BYE sip:3300@10.9.1.121 SIP/2.0 Via: SIP/2.0/UDP 10.9.5.107:10474;branch=z9hG4bK-d8754z-6a0e6508c2324c01-1---d8754z-;rport Max-Forwards: 70 Contact: To: "3300";tag=as2e6c69ae From: ;tag=704e5b3d Call-ID: NTI0NGI5ZTE2M2M4M2JkMDVhYWU4MzAyOGQ0OGIwMTY. CSeq: 3 BYE User-Agent: X-Lite release 1103k stamp 53621 Authorization: Digest username="8678",realm="asterisk",nonce="07d56872",uri="sip:3300@10.9.1.121",response="ceb16608e4dbf60f54ddd9038a2f8a37",algorithm=MD5 Reason: SIP;description="User Hung Up" Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Sending to 10.9.5.107 : 10474 (no NAT) Scheduling destruction of SIP dialog 'NTI0NGI5ZTE2M2M4M2JkMDVhYWU4MzAyOGQ0OGIwMTY.' in 32000 ms (Method: BYE) <--- Transmitting (no NAT) to 10.9.5.107:10474 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.9.5.107:10474;branch=z9hG4bK-d8754z-6a0e6508c2324c01-1---d8754z-;received=10.9.5.107;rport=10474 From: ;tag=704e5b3d To: "3300";tag=as2e6c69ae Call-ID: NTI0NGI5ZTE2M2M4M2JkMDVhYWU4MzAyOGQ0OGIwMTY. CSeq: 3 BYE Server: Asterisk PBX 1.6.1.5 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> Really destroying SIP dialog 'NTI0NGI5ZTE2M2M4M2JkMDVhYWU4MzAyOGQ0OGIwMTY.' Method: BYE asterisk*CLI> exit ]0;root@asterisk:/usr/src/asterisk-1.6.1.5[root@asterisk asterisk-1.6.1.5]#