=~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2009.09.14 09:50:12 =~=~=~=~=~=~=~=~=~=~=~= aseterisk ]0;root@asterisk:/usr/src/asterisk-1.6.1.4[root@asterisk asterisk-1.6.1.4]# asterisk -r Asterisk 1.6.1.4, 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 1.6.1.4 currently running on asterisk (pid = 14047) 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 54440 > 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/8678-0998c3b0", "") in new stack asterisk*CLI> -- Executing [3300@default:2] Dial("SIP/8678-0998c3b0", "SIP/SIP_VM/3300") in new stack == Using SIP RTP CoS mark 5 asterisk*CLI> -- Called SIP_VM/3300 asterisk*CLI> -- Got SIP response 302 "Moved Temporarily" back from 10.9.1.13 -- Now forwarding SIP/8678-0998c3b0 to 'SIP/3300::::TCP@10.9.1.13:5067' (thanks to SIP/SIP_VM-099e8918) == Using SIP RTP CoS mark 5 asterisk*CLI> -- SIP/10.9.1.13:5067-099edc70 is ringing asterisk*CLI> -- SIP/10.9.1.13:5067-099edc70 answered SIP/8678-0998c3b0 -- Packet2Packet bridging SIP/8678-0998c3b0 and SIP/10.9.1.13:5067-099edc70 asterisk*CLI> == Spawn extension (default, 3300, 2) exited non-zero on 'SIP/8678-0998c3b0' asterisk*CLI> sip set debug on asterisk*CLI> SIP Debugging enabled asterisk*CLI> <--- SIP read from UDP://10.9.5.107:54440 ---> INVITE sip:3300@10.9.1.121 SIP/2.0 Via: SIP/2.0/UDP 10.9.5.107:54440;branch=z9hG4bK-d8754z-fb2ed271bb0c5459-1---d8754z-;rport Max-Forwards: 70 Contact: To: "3300" From: ;tag=4b09af29 Call-ID: MDcxZTJlOTA5NjA2NTc5ZjIxNTE0ZjNjNjg5YjRlOWI. 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=- 2 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 35418 RTP/AVP 107 0 8 101 a=alt:1 1 : 9TLd1mKw voQ29jJ3 10.9.5.107 35418 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 : 54440 (NAT) Using INVITE request as basis request - MDcxZTJlOTA5NjA2NTc5ZjIxNTE0ZjNjNjg5YjRlOWI. asterisk*CLI> Found peer '8678' for '8678' from 10.9.5.107:54440 <--- Reliably Transmitting (no NAT) to 10.9.5.107:54440 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.9.5.107:54440;branch=z9hG4bK-d8754z-fb2ed271bb0c5459-1---d8754z-;received=10.9.5.107;rport=54440 From: ;tag=4b09af29 To: "3300";tag=as27612628 Call-ID: MDcxZTJlOTA5NjA2NTc5ZjIxNTE0ZjNjNjg5YjRlOWI. CSeq: 1 INVITE Server: Asterisk PBX 1.6.1.4 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0077a127" Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'MDcxZTJlOTA5NjA2NTc5ZjIxNTE0ZjNjNjg5YjRlOWI.' in 32000 ms (Method: INVITE) asterisk*CLI> <--- SIP read from UDP://10.9.5.107:54440 ---> ACK sip:3300@10.9.1.121 SIP/2.0 Via: SIP/2.0/UDP 10.9.5.107:54440;branch=z9hG4bK-d8754z-fb2ed271bb0c5459-1---d8754z-;rport To: "3300";tag=as27612628 From: ;tag=4b09af29 Call-ID: MDcxZTJlOTA5NjA2NTc5ZjIxNTE0ZjNjNjg5YjRlOWI. CSeq: 1 ACK Content-Length: 0 <-------------> --- (7 headers 0 lines) --- asterisk*CLI> <--- SIP read from UDP://10.9.5.107:54440 ---> INVITE sip:3300@10.9.1.121 SIP/2.0 Via: SIP/2.0/UDP 10.9.5.107:54440;branch=z9hG4bK-d8754z-91483f0736245a74-1---d8754z-;rport Max-Forwards: 70 Contact: To: "3300" From: ;tag=4b09af29 Call-ID: MDcxZTJlOTA5NjA2NTc5ZjIxNTE0ZjNjNjg5YjRlOWI. 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="0077a127",uri="sip:3300@10.9.1.121",response="1728403bb6b8700ed223be8cfdab4994",algorithm=MD5 Content-Length: 258 v=0 o=- 2 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 35418 RTP/AVP 107 0 8 101 a=alt:1 1 : 9TLd1mKw voQ29jJ3 10.9.5.107 35418 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 : 54440 (NAT) Using INVITE request as basis request - MDcxZTJlOTA5NjA2NTc5ZjIxNTE0ZjNjNjg5YjRlOWI. Found peer '8678' for '8678' from 10.9.5.107:54440 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 10.9.5.107:35418 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 10.9.5.107:35418 Looking for 3300 in default (domain 10.9.1.121) list_route: hop: <--- Transmitting (no NAT) to 10.9.5.107:54440 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.9.5.107:54440;branch=z9hG4bK-d8754z-91483f0736245a74-1---d8754z-;received=10.9.5.107;rport=54440 asterisk*CLI> From: ;tag=4b09af29 To: "3300" Call-ID: MDcxZTJlOTA5NjA2NTc5ZjIxNTE0ZjNjNjg5YjRlOWI. CSeq: 2 INVITE Server: Asterisk PBX 1.6.1.4 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: Content-Length: 0 <------------> -- Executing [3300@default:1] Answer("SIP/8678-099e42f0", "") in new stack Audio is at 10.9.1.121 port 19752 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 10.9.5.107:54440 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.9.5.107:54440;branch=z9hG4bK-d8754z-91483f0736245a74-1---d8754z-;received=10.9.5.107;rport=54440 From: ;tag=4b09af29 To: "3300";tag=as234c30a0 Call-ID: MDcxZTJlOTA5NjA2NTc5ZjIxNTE0ZjNjNjg5YjRlOWI. CSeq: 2 INVITE Server: Asterisk PBX 1.6.1.4 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 259 v=0 o=root 2102784026 2102784026 IN IP4 10.9.1.121 s=Asterisk PBX 1.6.1.4 c=IN IP4 10.9.1.121 t=0 0 m=audio 19752 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/8678-099e42f0", "SIP/SIP_VM/3300") in new stack == Using SIP RTP CoS mark 5 Audio is at 10.9.1.121 port 13264 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP 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=z9hG4bK0a76c402;rport Max-Forwards: 70 From: "Exchange User" ;tag=as7fc14ec1 To: Contact: Call-ID: 3b9bc03363bb030d63eb4c8265e6acde@10.9.1.121 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.4 Date: Mon, 14 Sep 2009 13:52:12 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Type: application/sdp Content-Length: 306 v=0 o=root 1364515244 1364515244 IN IP4 10.9.1.121 s=Asterisk PBX 1.6.1.4 c=IN IP4 10.9.1.121 t=0 0 m=audio 13264 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 --- -- Called SIP_VM/3300 asterisk*CLI> <--- SIP read from TCP://10.9.1.13:5060 ---> SIP/2.0 100 Trying FROM: "Exchange User";tag=as7fc14ec1 TO: CSEQ: 102 INVITE CALL-ID: 3b9bc03363bb030d63eb4c8265e6acde@10.9.1.121 VIA: SIP/2.0/TCP 10.9.1.121:5060;branch=z9hG4bK0a76c402;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=as7fc14ec1 TO: ;tag=26859f5f5f CSEQ: 102 INVITE CALL-ID: 3b9bc03363bb030d63eb4c8265e6acde@10.9.1.121 VIA: SIP/2.0/TCP 10.9.1.121:5060;branch=z9hG4bK0a76c402;rport CONTACT: CONTENT-LENGTH: 0 SERVER: RTCC/3.0.0.0 <-------------> --- (9 headers 0 lines) --- -- Got SIP response 302 "Moved Temporarily" back from 10.9.1.13 Transmitting (no NAT) to 10.9.1.13:5060: ACK sip:3300@10.9.1.13 SIP/2.0 Via: SIP/2.0/TCP 10.9.1.121:5060;branch=z9hG4bK0a76c402;rport Max-Forwards: 70 From: "Exchange User" ;tag=as7fc14ec1 To: ;tag=26859f5f5f Contact: Call-ID: 3b9bc03363bb030d63eb4c8265e6acde@10.9.1.121 CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.1.4 Content-Length: 0 --- -- Now forwarding SIP/8678-099e42f0 to 'SIP/3300::::TCP@10.9.1.13:5067' (thanks to SIP/SIP_VM-099eda48) == Using SIP RTP CoS mark 5 Audio is at 10.9.1.121 port 12410 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP 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=z9hG4bK27dd9e5d;rport Max-Forwards: 70 From: "Exchange User" ;tag=as568e72a1 To: Contact: Call-ID: 58cce21f785e433062591cd351fc3966@10.9.1.121 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.4 Date: Mon, 14 Sep 2009 13:52:12 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Type: application/sdp Content-Length: 306 v=0 o=root 1283861641 1283861641 IN IP4 10.9.1.121 s=Asterisk PBX 1.6.1.4 c=IN IP4 10.9.1.121 t=0 0 m=audio 12410 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> <--- SIP read from TCP://10.9.1.13:5067 ---> SIP/2.0 100 Trying FROM: "Exchange User";tag=as568e72a1 TO: CSEQ: 102 INVITE CALL-ID: 58cce21f785e433062591cd351fc3966@10.9.1.121 VIA: SIP/2.0/TCP 10.9.1.121:5060;branch=z9hG4bK27dd9e5d;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=as568e72a1 TO: ;epid=32BB1FF496;tag=d04db19e17 CSEQ: 102 INVITE CALL-ID: 58cce21f785e433062591cd351fc3966@10.9.1.121 VIA: SIP/2.0/TCP 10.9.1.121:5060;branch=z9hG4bK27dd9e5d;rport CONTENT-LENGTH: 0 SERVER: RTCC/3.0.0.0 <-------------> --- (8 headers 0 lines) --- -- SIP/10.9.1.13:5067-099eac00 is ringing asterisk*CLI> <--- SIP read from UDP://10.9.5.107:54440 ---> ACK sip:3300@10.9.1.121 SIP/2.0 Via: SIP/2.0/UDP 10.9.5.107:54440;branch=z9hG4bK-d8754z-045bb53b5f5f4627-1---d8754z-;rport Max-Forwards: 70 Contact: To: "3300";tag=as234c30a0 From: ;tag=4b09af29 Call-ID: MDcxZTJlOTA5NjA2NTc5ZjIxNTE0ZjNjNjg5YjRlOWI. CSeq: 2 ACK User-Agent: X-Lite release 1103k stamp 53621 Authorization: Digest username="8678",realm="asterisk",nonce="0077a127",uri="sip:3300@10.9.1.121",response="1728403bb6b8700ed223be8cfdab4994",algorithm=MD5 Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Really destroying SIP dialog '3b9bc03363bb030d63eb4c8265e6acde@10.9.1.121' Method: INVITE asterisk*CLI> <--- SIP read from TCP://10.9.1.13:5067 ---> SIP/2.0 200 OK FROM: "Exchange User";tag=as568e72a1 TO: ;epid=32BB1FF496;tag=d04db19e17 CSEQ: 102 INVITE CALL-ID: 58cce21f785e433062591cd351fc3966@10.9.1.121 VIA: SIP/2.0/TCP 10.9.1.121:5060;branch=z9hG4bK27dd9e5d;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 49920 RTP/AVP 0 8 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 <-------------> --- (12 headers 9 lines) --- Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 10.9.1.13:49920 Found audio description format telephone-event for ID 101 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xc (ulaw|alaw)/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 10.9.1.13:49920 list_route: hop: asterisk*CLI> set_destination: Parsing for address/port to send to set_destination: set destination to 10.9.1.13, port 5067 Transmitting (no NAT) to 10.9.1.13:5067: ACK sip:dialtone.Carahsoft.com:5067;transport=Tcp;maddr=10.9.1.13 SIP/2.0 Via: SIP/2.0/TCP 10.9.1.121:5060;branch=z9hG4bK74575bd6;rport Max-Forwards: 70 From: "Exchange User" ;tag=as568e72a1 To: ;tag=d04db19e17 Contact: Call-ID: 58cce21f785e433062591cd351fc3966@10.9.1.121 CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.1.4 Content-Length: 0 --- -- SIP/10.9.1.13:5067-099eac00 answered SIP/8678-099e42f0 -- Packet2Packet bridging SIP/8678-099e42f0 and SIP/10.9.1.13:5067-099eac00 asterisk*CLI> <--- SIP read from UDP://10.9.5.107:54440 ---> <-------------> asterisk*CLI> Really destroying SIP dialog 'OWJiYWFhMTViMmQyYjBlMjFjNDQ5ZDFlOTQ2NmQxODM.' Method: REGISTER asterisk*CLI> <--- SIP read from UDP://10.9.5.107:54440 ---> BYE sip:3300@10.9.1.121 SIP/2.0 Via: SIP/2.0/UDP 10.9.5.107:54440;branch=z9hG4bK-d8754z-be6c042e3767a16f-1---d8754z-;rport Max-Forwards: 70 Contact: To: "3300";tag=as234c30a0 From: ;tag=4b09af29 Call-ID: MDcxZTJlOTA5NjA2NTc5ZjIxNTE0ZjNjNjg5YjRlOWI. CSeq: 3 BYE User-Agent: X-Lite release 1103k stamp 53621 Authorization: Digest username="8678",realm="asterisk",nonce="0077a127",uri="sip:3300@10.9.1.121",response="2bbf75d61310a090d315a5ba333e9577",algorithm=MD5 Reason: SIP;description="User Hung Up" Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Sending to 10.9.5.107 : 54440 (NAT) <--- Transmitting (NAT) to 10.9.5.107:54440 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.9.5.107:54440;branch=z9hG4bK-d8754z-be6c042e3767a16f-1---d8754z-;received=10.9.5.107;rport=54440 From: ;tag=4b09af29 To: "3300";tag=as234c30a0 Call-ID: MDcxZTJlOTA5NjA2NTc5ZjIxNTE0ZjNjNjg5YjRlOWI. CSeq: 3 BYE Server: Asterisk PBX 1.6.1.4 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Length: 0 <------------> asterisk*CLI> Scheduling destruction of SIP dialog '58cce21f785e433062591cd351fc3966@10.9.1.121' in 32000 ms (Method: INVITE) set_destination: Parsing for address/port to send to set_destination: set destination to 10.9.1.13, port 5067 Reliably Transmitting (no NAT) to 10.9.1.13:5067: BYE sip:dialtone.Carahsoft.com:5067;transport=Tcp;maddr=10.9.1.13 SIP/2.0 Via: SIP/2.0/TCP 10.9.1.121:5060;branch=z9hG4bK00e82a82;rport Max-Forwards: 70 From: "Exchange User" ;tag=as568e72a1 To: ;tag=d04db19e17 Call-ID: 58cce21f785e433062591cd351fc3966@10.9.1.121 CSeq: 103 BYE User-Agent: Asterisk PBX 1.6.1.4 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- == Spawn extension (default, 3300, 2) exited non-zero on 'SIP/8678-099e42f0' asterisk*CLI> <--- SIP read from TCP://10.9.1.13:5067 ---> SIP/2.0 200 OK FROM: "Exchange User";tag=as568e72a1 TO: ;tag=d04db19e17;epid=32BB1FF496 CSEQ: 103 BYE CALL-ID: 58cce21f785e433062591cd351fc3966@10.9.1.121 VIA: SIP/2.0/TCP 10.9.1.121:5060;branch=z9hG4bK00e82a82;rport CONTENT-LENGTH: 0 SERVER: RTCC/3.0.0.0 <-------------> --- (8 headers 0 lines) --- asterisk*CLI> Really destroying SIP dialog '58cce21f785e433062591cd351fc3966@10.9.1.121' Method: INVITE Really destroying SIP dialog 'MDcxZTJlOTA5NjA2NTc5ZjIxNTE0ZjNjNjg5YjRlOWI.' Method: BYE asterisk*CLI> exit ]0;root@asterisk:/usr/src/asterisk-1.6.1.4[root@asterisk asterisk-1.6.1.4]#