=~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2009.09.14 09:55:03 =~=~=~=~=~=~=~=~=~=~=~= 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 = 15144) 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 30818 > 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/8678-08937548", "") in new stack asterisk*CLI> -- Executing [3300@default:2] Dial("SIP/8678-08937548", "SIP/SIP_VM/3300") in new stack == Using SIP RTP CoS mark 5 asterisk*CLI> -- Called SIP_VM/3300 asterisk*CLI> -- SIP/SIP_VM-0893a838 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Executing [3300@default:3] Busy("SIP/8678-08937548", "") in new stack asterisk*CLI> == Spawn extension (default, 3300, 3) exited non-zero on 'SIP/8678-08937548' asterisk*CLI> sip set debug on asterisk*CLI> SIP Debugging enabled asterisk*CLI> <--- SIP read from UDP://10.9.5.107:30818 ---> <-------------> asterisk*CLI> Really destroying SIP dialog 'd2cdd921389f48abb0deb8d1588ed210' Method: OPTIONS asterisk*CLI> <--- SIP read from UDP://10.9.5.107:30818 ---> INVITE sip:3300@10.9.1.121 SIP/2.0 Via: SIP/2.0/UDP 10.9.5.107:30818;branch=z9hG4bK-d8754z-8a3a326a2c37953a-1---d8754z-;rport Max-Forwards: 70 Contact: To: "3300" From: ;tag=305dea68 Call-ID: N2U0ZTAyMmQ0YzQzYjQ2ZDBkNjNjMWIyNWMyMGU4Nzg. 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=- 1 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 48198 RTP/AVP 107 0 8 101 a=alt:1 1 : lTp8p+jn nttlv0B0 10.9.5.107 48198 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 : 30818 (no NAT) Using INVITE request as basis request - N2U0ZTAyMmQ0YzQzYjQ2ZDBkNjNjMWIyNWMyMGU4Nzg. Found peer '8678' for '8678' from 10.9.5.107:30818 <--- Reliably Transmitting (no NAT) to 10.9.5.107:30818 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.9.5.107:30818;branch=z9hG4bK-d8754z-8a3a326a2c37953a-1---d8754z-;received=10.9.5.107;rport=30818 From: ;tag=305dea68 To: "3300";tag=as585e3f33 Call-ID: N2U0ZTAyMmQ0YzQzYjQ2ZDBkNjNjMWIyNWMyMGU4Nzg. 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="32d944e3" Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'N2U0ZTAyMmQ0YzQzYjQ2ZDBkNjNjMWIyNWMyMGU4Nzg.' in 32000 ms (Method: INVITE) asterisk*CLI> <--- SIP read from UDP://10.9.5.107:30818 ---> ACK sip:3300@10.9.1.121 SIP/2.0 Via: SIP/2.0/UDP 10.9.5.107:30818;branch=z9hG4bK-d8754z-8a3a326a2c37953a-1---d8754z-;rport To: "3300";tag=as585e3f33 From: ;tag=305dea68 Call-ID: N2U0ZTAyMmQ0YzQzYjQ2ZDBkNjNjMWIyNWMyMGU4Nzg. CSeq: 1 ACK Content-Length: 0 <-------------> --- (7 headers 0 lines) --- asterisk*CLI> <--- SIP read from UDP://10.9.5.107:30818 ---> INVITE sip:3300@10.9.1.121 SIP/2.0 Via: SIP/2.0/UDP 10.9.5.107:30818;branch=z9hG4bK-d8754z-957f4f3d61715216-1---d8754z-;rport Max-Forwards: 70 Contact: To: "3300" From: ;tag=305dea68 Call-ID: N2U0ZTAyMmQ0YzQzYjQ2ZDBkNjNjMWIyNWMyMGU4Nzg. 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="32d944e3",uri="sip:3300@10.9.1.121",response="34c521c4fc910a36e2ec03c6b0327ca4",algorithm=MD5 Content-Length: 258 v=0 o=- 1 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 48198 RTP/AVP 107 0 8 101 a=alt:1 1 : lTp8p+jn nttlv0B0 10.9.5.107 48198 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 : 30818 (no NAT) Using INVITE request as basis request - N2U0ZTAyMmQ0YzQzYjQ2ZDBkNjNjMWIyNWMyMGU4Nzg. Found peer '8678' for '8678' from 10.9.5.107:30818 asterisk*CLI> 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 10.9.5.107:48198 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 10.9.5.107:48198 Looking for 3300 in default (domain 10.9.1.121) list_route: hop: <--- Transmitting (no NAT) to 10.9.5.107:30818 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.9.5.107:30818;branch=z9hG4bK-d8754z-957f4f3d61715216-1---d8754z-;received=10.9.5.107;rport=30818 From: ;tag=305dea68 To: "3300" Call-ID: N2U0ZTAyMmQ0YzQzYjQ2ZDBkNjNjMWIyNWMyMGU4Nzg. 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/8678-0893efd0", "") in new stack asterisk*CLI> Audio is at 10.9.1.121 port 16220 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 10.9.5.107:30818 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.9.5.107:30818;branch=z9hG4bK-d8754z-957f4f3d61715216-1---d8754z-;received=10.9.5.107;rport=30818 From: ;tag=305dea68 To: "3300";tag=as785ce36a Call-ID: N2U0ZTAyMmQ0YzQzYjQ2ZDBkNjNjMWIyNWMyMGU4Nzg. 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 1629086808 1629086808 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 16220 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/8678-0893efd0", "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 18698 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 10.9.1.13:5060: INVITE sip:3300@10.9.1.13 SIP/2.0 Via: SIP/2.0/TCP 10.9.1.121:5060;branch=z9hG4bK02d739b0;rport Max-Forwards: 70 From: "Exchange User" ;tag=as3b707316 To: Contact: Call-ID: 0346a7e95dfdf4f633066e1642c1c024@10.9.1.121 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.5 Date: Mon, 14 Sep 2009 13:57:43 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 1988435507 1988435507 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 18698 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://10.9.1.13:5060 ---> SIP/2.0 100 Trying FROM: "Exchange User";tag=as3b707316 TO: CSEQ: 102 INVITE CALL-ID: 0346a7e95dfdf4f633066e1642c1c024@10.9.1.121 VIA: SIP/2.0/TCP 10.9.1.121:5060;branch=z9hG4bK02d739b0;rport CONTENT-LENGTH: 0 <-------------> --- (7 headers 0 lines) --- asterisk*CLI> <--- SIP read from TCP://10.9.1.13:5060 ---> SIP/2.0 302 Moved Temporarily FROM: "Exchange User";tag=as3b707316 TO: ;tag=8336d1eb6e CSEQ: 102 INVITE CALL-ID: 0346a7e95dfdf4f633066e1642c1c024@10.9.1.121 VIA: SIP/2.0/TCP 10.9.1.121:5060;branch=z9hG4bK02d739b0;rport CONTACT: CONTENT-LENGTH: 0 SERVER: RTCC/3.0.0.0 <-------------> --- (9 headers 0 lines) --- asterisk*CLI> <--- SIP read from UDP://10.9.5.107:30818 ---> ACK sip:3300@10.9.1.121 SIP/2.0 Via: SIP/2.0/UDP 10.9.5.107:30818;branch=z9hG4bK-d8754z-6d7c2d632d4b3624-1---d8754z-;rport Max-Forwards: 70 Contact: To: "3300";tag=as785ce36a From: ;tag=305dea68 Call-ID: N2U0ZTAyMmQ0YzQzYjQ2ZDBkNjNjMWIyNWMyMGU4Nzg. CSeq: 2 ACK User-Agent: X-Lite release 1103k stamp 53621 Authorization: Digest username="8678",realm="asterisk",nonce="32d944e3",uri="sip:3300@10.9.1.121",response="34c521c4fc910a36e2ec03c6b0327ca4",algorithm=MD5 Content-Length: 0 <-------------> --- (11 headers 0 lines) --- asterisk*CLI> Really destroying SIP dialog '7990b4631e93d78b2b5ea9b949f9fb06@10.9.1.121' Method: INVITE asterisk*CLI> <--- SIP read from UDP://10.9.5.107:30818 ---> <-------------> asterisk*CLI> -- SIP/SIP_VM-089428d8 is circuit-busy Scheduling destruction of SIP dialog '0346a7e95dfdf4f633066e1642c1c024@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/8678-0893efd0", "") in new stack asterisk*CLI> <--- SIP read from UDP://10.9.5.107:30818 ---> BYE sip:3300@10.9.1.121 SIP/2.0 Via: SIP/2.0/UDP 10.9.5.107:30818;branch=z9hG4bK-d8754z-3c2b3e01764b7136-1---d8754z-;rport Max-Forwards: 70 Contact: To: "3300";tag=as785ce36a From: ;tag=305dea68 Call-ID: N2U0ZTAyMmQ0YzQzYjQ2ZDBkNjNjMWIyNWMyMGU4Nzg. CSeq: 3 BYE User-Agent: X-Lite release 1103k stamp 53621 Authorization: Digest username="8678",realm="asterisk",nonce="32d944e3",uri="sip:3300@10.9.1.121",response="fd3abdbcafedab38db00bc3efcda46c2",algorithm=MD5 Reason: SIP;description="User Hung Up" Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Sending to 10.9.5.107 : 30818 (no NAT) <--- Transmitting (no NAT) to 10.9.5.107:30818 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.9.5.107:30818;branch=z9hG4bK-d8754z-3c2b3e01764b7136-1---d8754z-;received=10.9.5.107;rport=30818 From: ;tag=305dea68 To: "3300";tag=as785ce36a Call-ID: N2U0ZTAyMmQ0YzQzYjQ2ZDBkNjNjMWIyNWMyMGU4Nzg. 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/8678-0893efd0' asterisk*CLI> Really destroying SIP dialog 'N2U0ZTAyMmQ0YzQzYjQ2ZDBkNjNjMWIyNWMyMGU4Nzg.' Method: BYE asterisk*CLI> exit ]0;root@asterisk:/usr/src/asterisk-1.6.1.5[root@asterisk asterisk-1.6.1.5]#