Asterisk SVN-trunk-r81159, Copyright (C) 1999 - 2007 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. ========================================================================= NOTE: This is a development version of Asterisk, and should not be used in production installations. [ Booting... [ Reading Master Configuration ] [ Initializing Custom Configuration Options ] [Aug 28 14:51:17] NOTICE[4949]: cdr.c:1349 do_reload: CDR simple logging enabled. [Aug 28 14:51:17] NOTICE[4949]: loader.c:831 load_modules: 156 modules will be loaded. [Aug 28 14:51:17] WARNING[4949]: res_smdi.c:715 load_module: No SMDI interfaces are available to listen on, not starting SDMI listener. ............SIP channel loading... ..........................................[Aug 28 14:51:17] WARNING[4949]: chan_iax2.c:11273 load_module: Unable to open IAX timing interface: No such file or directory ...................[Aug 28 14:51:17] WARNING[4949]: app_minivm.c:2394 load_config: Failed to load configuration file. Module activated with default settings. .....................................[Aug 28 14:51:17] NOTICE[4949]: config.c:1315 ast_config_engine_register: Registered Config Engine mysql .........[Aug 28 14:51:17] WARNING[4949]: cdr_sqlite3_custom.c:85 load_config: cdr_sqlite3_custom: Failed to load configuration file. Module not activated. [Aug 28 14:51:17] WARNING[4949]: cdr_sqlite3_custom.c:232 load_module: cdr_sqlite3_custom: near "(": syntax error. .....[Aug 28 14:51:17] NOTICE[4949]: pbx_ael.c:1680 pbx_load_module: Starting AEL load process. [Aug 28 14:51:17] NOTICE[4949]: pbx_ael.c:1687 pbx_load_module: AEL load process: calculated config file name '/etc/asterisk/extensions.ael'. [Aug 28 14:51:17] NOTICE[4949]: pbx_ael.c:1695 pbx_load_module: AEL load process: parsed config file name '/etc/asterisk/extensions.ael'. [Aug 28 14:51:17] NOTICE[4949]: pbx_ael.c:1698 pbx_load_module: AEL load process: checked config file name '/etc/asterisk/extensions.ael'. [Aug 28 14:51:17] NOTICE[4949]: pbx_ael.c:1700 pbx_load_module: AEL load process: compiled config file name '/etc/asterisk/extensions.ael'. [Aug 28 14:51:17] NOTICE[4949]: pbx_ael.c:1703 pbx_load_module: AEL load process: merged config file name '/etc/asterisk/extensions.ael'. [Aug 28 14:51:17] NOTICE[4949]: pbx_ael.c:1706 pbx_load_module: AEL load process: verified config file name '/etc/asterisk/extensions.ael'. ............................. ] Asterisk Ready. ]1;Asterisk]2;Asterisk Console on 'pbx-gr-backup.cc.huji.ac.il' (pid 4945)*CLI> core set verbose 5 Verbosity was 0 and is now 5 *CLI> sip set debug on SIP Debugging enabled *CLI> <--- SIP read from 128.139.26.6:58203 ---> INVITE sip:80609@132.64.9.162:5060 SIP/2.0 Via: SIP/2.0/UDP 128.139.26.6:5060;branch=z9hG4bK200D124 Remote-Party-ID: ;party=calling;screen=no;privacy=off From: ;tag=AAABF94C-49D To: Date: Tue, 28 Aug 2007 12:05:03 GMT Call-ID: BE5A119A-549511DC-B1DEEAD3-358FA890@128.139.26.6 Supported: 100rel,timer,resource-priority,replaces Min-SE: 1800 Cisco-Guid: 3193453705-1419055580-2568552474-801595720 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER CSeq: 101 INVITE Max-Forwards: 70 Timestamp: 1188302703 Contact: Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: 245 v=0 o=CiscoSystemsSIP-GW-UserAgent 235 777 IN IP4 128.139.26.6 s=SIP Call c=IN IP4 128.139.26.6 t=0 0 m=audio 17710 RTP/AVP 8 101 c=IN IP4 128.139.26.6 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 <-------------> --- (21 headers 11 lines) ---  == Using TOS bits 0  == Using CoS mark 5  == Using TOS bits 0  == Using CoS mark 6  == Using TOS bits 0  == Using CoS mark 5 Sending to 128.139.26.6 : 5060 (no NAT) Using INVITE request as basis request - BE5A119A-549511DC-B1DEEAD3-358FA890@128.139.26.6 No user '7106' in SIP users list Found peer '128.139.26.6' for '7106' from 128.139.26.6:58203 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 128.139.26.6:17710 Found description format PCMA for ID 8 Found description format telephone-event for ID 101 Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 128.139.26.6:17710 Looking for 80609 in huji-local (domain 132.64.9.162) list_route: hop: <--- Transmitting (no NAT) to 128.139.26.6:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 128.139.26.6:5060;branch=z9hG4bK200D124;received=128.139.26.6 From: ;tag=AAABF94C-49D To: Call-ID: BE5A119A-549511DC-B1DEEAD3-358FA890@128.139.26.6 CSeq: 101 INVITE User-Agent: Asterisk PBX SVN-trunk-r81159 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------>  -- Executing [80609@huji-local:1] NoOp("SIP/gr-pbx-link-087bbcc0", "") in new stack  -- Executing [80609@huji-local:2] Set("SIP/gr-pbx-link-087bbcc0", "_To=80609") in new stack  -- Executing [80609@huji-local:3] Set("SIP/gr-pbx-link-087bbcc0", "_From=7106") in new stack  -- Executing [80609@huji-local:4] Set("SIP/gr-pbx-link-087bbcc0", "DB(80609/LastCaller)=7106") in new stack  -- Executing [80609@huji-local:5] Set("SIP/gr-pbx-link-087bbcc0", "DB(7106/LastCalled)=80609") in new stack  -- Executing [80609@huji-local:6] MYSQL("SIP/gr-pbx-link-087bbcc0", "Connect connid localhost asterisk NotMe43e080 asterisk") in new stack  -- Executing [80609@huji-local:7] MYSQL("SIP/gr-pbx-link-087bbcc0", "Query resID 1 SELECT additional_numbers from shared_lines where orig_number='80609'") in new stack  -- Executing [80609@huji-local:8] MYSQL("SIP/gr-pbx-link-087bbcc0", "Fetch FetchId 2 aEXTEN") in new stack  -- Executing [80609@huji-local:9] NoOp("SIP/gr-pbx-link-087bbcc0", "") in new stack  -- Executing [80609@huji-local:10] MYSQL("SIP/gr-pbx-link-087bbcc0", "Clear 2") in new stack  -- Executing [80609@huji-local:11] MYSQL("SIP/gr-pbx-link-087bbcc0", "Query resID 1 SELECT callerid from sip_users where name='80609'") in new stack  -- Executing [80609@huji-local:12] MYSQL("SIP/gr-pbx-link-087bbcc0", "Fetch FetchId 2 CalledName") in new stack  -- Executing [80609@huji-local:13] MYSQL("SIP/gr-pbx-link-087bbcc0", "Clear 2") in new stack  -- Executing [80609@huji-local:14] GotoIf("SIP/gr-pbx-link-087bbcc0", "1?15:21") in new stack  -- Goto (huji-local,80609,15)  -- Executing [80609@huji-local:15] MYSQL("SIP/gr-pbx-link-087bbcc0", "Query resID 1 SELECT name from full_names where number=7106") in new stack  -- Executing [80609@huji-local:16] MYSQL("SIP/gr-pbx-link-087bbcc0", "Fetch FetchId 2 CallingName") in new stack  -- Executing [80609@huji-local:17] MYSQL("SIP/gr-pbx-link-087bbcc0", "Clear 2") in new stack  -- Executing [80609@huji-local:18] GotoIf("SIP/gr-pbx-link-087bbcc0", "0?19:20") in new stack  -- Goto (huji-local,80609,20)  -- Executing [80609@huji-local:20] NoOp("SIP/gr-pbx-link-087bbcc0", "Finish if-if-huji-local-4-5") in new stack  -- Executing [80609@huji-local:21] NoOp("SIP/gr-pbx-link-087bbcc0", "Finish if-huji-local-4") in new stack  -- Executing [80609@huji-local:22] MYSQL("SIP/gr-pbx-link-087bbcc0", "Disconnect 1") in new stack  -- Executing [80609@huji-local:23] Dial("SIP/gr-pbx-link-087bbcc0", "SIP/80609,20,L(3600000:60000:30000)") in new stack  -- Limit Data for this call:  > timelimit = 3600000  > play_warning = 60000  > play_to_caller = yes  > play_to_callee = no  > warning_freq = 30000  > start_sound =  > warning_sound = timeleft  > end_sound =  == Using TOS bits 0  == Using CoS mark 5  == Using TOS bits 0  == Using CoS mark 6  == Using TOS bits 0  == Using CoS mark 5 Audio is at 132.64.9.162 port 60134 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 132.64.4.137:2059: INVITE sip:80609@132.64.4.137:2059;line=5fnpelxl SIP/2.0 Via: SIP/2.0/UDP 132.64.9.162:5060;branch=z9hG4bK415cef09;rport Max-Forwards: 70 From: "7106" ;tag=as77f172fb To: Contact: Call-ID: 1b133afd3c4bb85f31b6580b4ef259dc@132.64.9.162 CSeq: 102 INVITE User-Agent: Asterisk PBX SVN-trunk-r81159 Remote-Party-ID: "7106" ;privacy=off;screen=no Date: Tue, 28 Aug 2007 11:51:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 250 v=0 o=root 2132778888 2132778888 IN IP4 128.139.26.6 s=session c=IN IP4 128.139.26.6 t=0 0 m=audio 17710 RTP/AVP 8 101 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 80609 <--- SIP read from 132.64.4.137:2059 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 132.64.9.162:5060;branch=z9hG4bK415cef09;rport=5060 From: "7106" ;tag=as77f172fb To: "Test" ;tag=mmnlvqxsvq Call-ID: 1b133afd3c4bb85f31b6580b4ef259dc@132.64.9.162 CSeq: 102 INVITE Contact: ;flow-id=1 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Content-Length: 0 <-------------> --- (10 headers 0 lines) ---  -- SIP/80609-087c5728 is ringing <--- Transmitting (no NAT) to 128.139.26.6:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 128.139.26.6:5060;branch=z9hG4bK200D124;received=128.139.26.6 From: ;tag=AAABF94C-49D To: ;tag=as07da1167 Call-ID: BE5A119A-549511DC-B1DEEAD3-358FA890@128.139.26.6 CSeq: 101 INVITE User-Agent: Asterisk PBX SVN-trunk-r81159 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> <--- SIP read from 132.64.4.137:2059 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 132.64.9.162:5060;branch=z9hG4bK415cef09;rport=5060 From: "7106" ;tag=as77f172fb To: "Test" ;tag=mmnlvqxsvq Call-ID: 1b133afd3c4bb85f31b6580b4ef259dc@132.64.9.162 CSeq: 102 INVITE Contact: ;flow-id=1 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Content-Length: 0 <-------------> --- (10 headers 0 lines) ---  -- SIP/80609-087c5728 is ringing <--- SIP read from 132.64.4.137:2059 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 132.64.9.162:5060;branch=z9hG4bK415cef09;rport=5060 From: "7106" ;tag=as77f172fb To: "Test" ;tag=mmnlvqxsvq Call-ID: 1b133afd3c4bb85f31b6580b4ef259dc@132.64.9.162 CSeq: 102 INVITE Contact: ;flow-id=1 User-Agent: snom320/6.2.2 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces, callerid Content-Type: application/sdp Content-Length: 327 v=0 o=root 1095695981 1095695982 IN IP4 132.64.4.137 s=call c=IN IP4 132.64.4.137 t=0 0 m=audio 56748 RTP/AVP 8 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:T2xcshdYx1E91R63pkwp9oyUlMExjnxiE9mY89Nv a=rtpmap:8 pcma/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=encryption:optional a=sendrecv <-------------> --- (13 headers 13 lines) --- Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 132.64.4.137:56748 Found description format pcma for ID 8 Found description format telephone-event for ID 101 Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 132.64.4.137:56748 --- set_address_from_contact host '132.64.4.137' list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 132.64.4.137, port 2059 Transmitting (no NAT) to 132.64.4.137:2059: ACK sip:80609@132.64.4.137:2059;line=5fnpelxl SIP/2.0 Via: SIP/2.0/UDP 132.64.9.162:5060;branch=z9hG4bK17d9392e;rport Max-Forwards: 70 From: "7106" ;tag=as77f172fb To: ;tag=mmnlvqxsvq Contact: Call-ID: 1b133afd3c4bb85f31b6580b4ef259dc@132.64.9.162 CSeq: 102 ACK User-Agent: Asterisk PBX SVN-trunk-r81159 Remote-Party-ID: "7106" ;privacy=off;screen=no Content-Length: 0 ---  -- SIP/80609-087c5728 answered SIP/gr-pbx-link-087bbcc0 Audio is at 132.64.9.162 port 36838 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 128.139.26.6:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 128.139.26.6:5060;branch=z9hG4bK200D124;received=128.139.26.6 From: ;tag=AAABF94C-49D To: ;tag=as07da1167 Call-ID: BE5A119A-549511DC-B1DEEAD3-358FA890@128.139.26.6 CSeq: 101 INVITE User-Agent: Asterisk PBX SVN-trunk-r81159 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 244 v=0 o=root 8792145 8792145 IN IP4 132.64.4.137 s=session c=IN IP4 132.64.4.137 t=0 0 m=audio 56748 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> <--- SIP read from 128.139.26.6:58203 ---> ACK sip:80609@132.64.9.162:5060 SIP/2.0 Via: SIP/2.0/UDP 128.139.26.6:5060;branch=z9hG4bK200E1F92 From: ;tag=AAABF94C-49D To: ;tag=as07da1167 Date: Tue, 28 Aug 2007 12:05:03 GMT Call-ID: BE5A119A-549511DC-B1DEEAD3-358FA890@128.139.26.6 Max-Forwards: 70 CSeq: 101 ACK Allow-Events: telephone-event Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 132.64.4.137:2059 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 132.64.9.162:5060;branch=z9hG4bK415cef09;rport=5060 From: "7106" ;tag=as77f172fb To: "Test" ;tag=mmnlvqxsvq Call-ID: 1b133afd3c4bb85f31b6580b4ef259dc@132.64.9.162 CSeq: 102 INVITE Contact: ;flow-id=1 User-Agent: snom320/6.2.2 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces, callerid Content-Type: application/sdp Content-Length: 327 v=0 o=root 1095695981 1095695982 IN IP4 132.64.4.137 s=call c=IN IP4 132.64.4.137 t=0 0 m=audio 56748 RTP/AVP 8 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:T2xcshdYx1E91R63pkwp9oyUlMExjnxiE9mY89Nv a=rtpmap:8 pcma/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=encryption:optional a=sendrecv <-------------> --- (13 headers 13 lines) --- Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 132.64.4.137:56748 Found description format pcma for ID 8 Found description format telephone-event for ID 101 Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 132.64.4.137:56748 --- set_address_from_contact host '132.64.4.137' set_destination: Parsing for address/port to send to set_destination: set destination to 132.64.4.137, port 2059 Transmitting (no NAT) to 132.64.4.137:2059: ACK sip:80609@132.64.4.137:2059;line=5fnpelxl SIP/2.0 Via: SIP/2.0/UDP 132.64.9.162:5060;branch=z9hG4bK6bf858d3;rport Max-Forwards: 70 From: "7106" ;tag=as77f172fb To: ;tag=mmnlvqxsvq Contact: Call-ID: 1b133afd3c4bb85f31b6580b4ef259dc@132.64.9.162 CSeq: 102 ACK User-Agent: Asterisk PBX SVN-trunk-r81159 Remote-Party-ID: "7106" ;privacy=off;screen=no Content-Length: 0 --- <--- SIP read from 132.64.4.137:2059 ---> INVITE sip:7106@132.64.9.162 SIP/2.0 Via: SIP/2.0/UDP 132.64.4.137:2059;branch=z9hG4bK-q31lbws2uyoe;rport From: "Test" ;tag=mmnlvqxsvq To: "7106" ;tag=as77f172fb Call-ID: 1b133afd3c4bb85f31b6580b4ef259dc@132.64.9.162 CSeq: 1 INVITE Max-Forwards: 70 Contact: ;flow-id=1 P-Key-Flags: keys="3" User-Agent: snom320/6.2.2 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces, callerid Session-Expires: 3600;refresher=uas Min-SE: 90 Content-Type: application/sdp Content-Length: 475 v=0 o=root 1095695981 1095695983 IN IP4 132.64.4.137 s=call c=IN IP4 132.64.4.137 t=0 0 m=audio 56748 RTP/AVP 0 8 9 2 3 18 4 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:T2xcshdYx1E91R63pkwp9oyUlMExjnxiE9mY89Nv a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:18 g729/8000 a=rtpmap:4 g723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=encryption:optional a=sendonly <-------------> --- (18 headers 19 lines) --- Sending to 132.64.4.137 : 2059 (NAT) Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 9 Found RTP audio format 2 Found RTP audio format 3 Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 101 Peer audio RTP is at port 132.64.4.137:56748 Found description format pcmu for ID 0 Found description format pcma for ID 8 Found description format g722 for ID 9 Found description format g726-32 for ID 2 Found description format gsm for ID 3 Found description format g729 for ID 18 Found description format g723 for ID 4 Found description format telephone-event for ID 101 Capabilities: us - 0x8 (alaw), peer - audio=0x190f (g723|gsm|ulaw|alaw|g726|g729|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 132.64.4.137:56748 <--- Transmitting (NAT) to 132.64.4.137:2059 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 132.64.4.137:2059;branch=z9hG4bK-q31lbws2uyoe;received=132.64.4.137;rport=2059 From: "Test" ;tag=mmnlvqxsvq To: "7106" ;tag=as77f172fb Call-ID: 1b133afd3c4bb85f31b6580b4ef259dc@132.64.9.162 CSeq: 1 INVITE User-Agent: Asterisk PBX SVN-trunk-r81159 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> Audio is at 132.64.9.162 port 60134 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Transmitting (NAT) to 132.64.4.137:2059 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 132.64.4.137:2059;branch=z9hG4bK-q31lbws2uyoe;received=132.64.4.137;rport=2059 From: "Test" ;tag=mmnlvqxsvq To: "7106" ;tag=as77f172fb Call-ID: 1b133afd3c4bb85f31b6580b4ef259dc@132.64.9.162 CSeq: 1 INVITE User-Agent: Asterisk PBX SVN-trunk-r81159 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 250 v=0 o=root 2132778888 2132778889 IN IP4 128.139.26.6 s=session c=IN IP4 128.139.26.6 t=0 0 m=audio 17710 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=recvonly <------------>  -- Started music on hold, class 'default', on SIP/gr-pbx-link-087bbcc0 <--- SIP read from 132.64.4.137:2059 ---> ACK sip:7106@132.64.9.162 SIP/2.0 Via: SIP/2.0/UDP 132.64.4.137:2059;branch=z9hG4bK-00976ge7orub;rport From: "Test" ;tag=mmnlvqxsvq To: "7106" ;tag=as77f172fb Call-ID: 1b133afd3c4bb85f31b6580b4ef259dc@132.64.9.162 CSeq: 1 ACK Max-Forwards: 70 Contact: ;flow-id=1 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- <--- SIP read from 132.64.4.137:2059 ---> INVITE sip:80620@pbx-gr-backup.cc.huji.ac.il;user=phone SIP/2.0 Via: SIP/2.0/UDP 132.64.4.137:2059;branch=z9hG4bK-xspnfia8812v;rport From: "Test" ;tag=5tui7rcix8 To: Call-ID: 3c36944683d6-ndhoic4byj9m@snom320-0004132480B4 CSeq: 1 INVITE Max-Forwards: 70 Contact: ;flow-id=1 P-Key-Flags: keys="3" User-Agent: snom320/6.2.2 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces, callerid Session-Expires: 3600;refresher=uas Min-SE: 90 Content-Type: application/sdp Content-Length: 475 v=0 o=root 1572994474 1572994474 IN IP4 132.64.4.137 s=call c=IN IP4 132.64.4.137 t=0 0 m=audio 65030 RTP/AVP 0 8 9 2 3 18 4 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:POt1KNvI7YHRzymCpj6SG+ZsLlYo51161JZIdVq9 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:18 g729/8000 a=rtpmap:4 g723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=encryption:optional a=sendrecv <-------------> --- (18 headers 19 lines) ---  == Using TOS bits 0  == Using CoS mark 5  == Using TOS bits 0  == Using CoS mark 6  == Using TOS bits 0  == Using CoS mark 5 Sending to 132.64.4.137 : 2059 (NAT) Using INVITE request as basis request - 3c36944683d6-ndhoic4byj9m@snom320-0004132480B4 Found user '80609' for '80609' <--- Reliably Transmitting (no NAT) to 132.64.4.137:2059 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 132.64.4.137:2059;branch=z9hG4bK-xspnfia8812v;received=132.64.4.137;rport=2059 From: "Test" ;tag=5tui7rcix8 To: ;tag=as09a14613 Call-ID: 3c36944683d6-ndhoic4byj9m@snom320-0004132480B4 CSeq: 1 INVITE User-Agent: Asterisk PBX SVN-trunk-r81159 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="cc.huji.ac.il", nonce="424ea6ff" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '3c36944683d6-ndhoic4byj9m@snom320-0004132480B4' in 32000 ms (Method: INVITE) <--- SIP read from 132.64.4.137:2059 ---> SUBSCRIBE sip:80620@pbx-gr-backup.cc.huji.ac.il;user=phone SIP/2.0 Via: SIP/2.0/UDP 132.64.4.137:2059;branch=z9hG4bK-suin2q8p52nh;rport From: ;tag=j1fbjrjd7x To: Call-ID: 3c369447a122-tyblrm7hxwun@snom320-0004132480B4 CSeq: 1 SUBSCRIBE Max-Forwards: 70 Contact: ;flow-id=1 Event: dialog;purpose=call-completion Accept: application/dialog-info+xml Expires: 3600 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Creating new subscription Sending to 132.64.4.137 : 2059 (NAT) Found peer '80609' for '80609' from 132.64.4.137:2059 <--- Transmitting (no NAT) to 132.64.4.137:2059 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 132.64.4.137:2059;branch=z9hG4bK-suin2q8p52nh;received=132.64.4.137;rport=2059 From: ;tag=j1fbjrjd7x To: ;tag=as7f4ab585 Call-ID: 3c369447a122-tyblrm7hxwun@snom320-0004132480B4 CSeq: 1 SUBSCRIBE User-Agent: Asterisk PBX SVN-trunk-r81159 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="cc.huji.ac.il", nonce="7b162dd9" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '3c369447a122-tyblrm7hxwun@snom320-0004132480B4' in 32000 ms (Method: SUBSCRIBE) <--- SIP read from 132.64.4.137:2059 ---> ACK sip:80620@pbx-gr-backup.cc.huji.ac.il;user=phone SIP/2.0 Via: SIP/2.0/UDP 132.64.4.137:2059;branch=z9hG4bK-xspnfia8812v;rport From: "Test" ;tag=5tui7rcix8 To: ;tag=as09a14613 Call-ID: 3c36944683d6-ndhoic4byj9m@snom320-0004132480B4 CSeq: 1 ACK Max-Forwards: 70 Contact: ;flow-id=1 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- <--- SIP read from 132.64.4.137:2059 ---> INVITE sip:80620@pbx-gr-backup.cc.huji.ac.il;user=phone SIP/2.0 Via: SIP/2.0/UDP 132.64.4.137:2059;branch=z9hG4bK-7yny0keushsf;rport From: "Test" ;tag=5tui7rcix8 To: Call-ID: 3c36944683d6-ndhoic4byj9m@snom320-0004132480B4 CSeq: 2 INVITE Max-Forwards: 70 Contact: ;flow-id=1 P-Key-Flags: keys="3" User-Agent: snom320/6.2.2 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces, callerid Session-Expires: 3600;refresher=uas Min-SE: 90 Authorization: Digest username="80609",realm="cc.huji.ac.il",nonce="424ea6ff",uri="sip:80620@pbx-gr-backup.cc.huji.ac.il;user=phone",response="b8b20efd6a0c298a18b89b18029dee5c",algorithm=md5 Content-Type: application/sdp Content-Length: 475 v=0 o=root 1572994474 1572994474 IN IP4 132.64.4.137 s=call c=IN IP4 132.64.4.137 t=0 0 m=audio 65030 RTP/AVP 0 8 9 2 3 18 4 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:POt1KNvI7YHRzymCpj6SG+ZsLlYo51161JZIdVq9 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:18 g729/8000 a=rtpmap:4 g723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=encryption:optional a=sendrecv <-------------> --- (19 headers 19 lines) --- Sending to 132.64.4.137 : 2059 (NAT) Using INVITE request as basis request - 3c36944683d6-ndhoic4byj9m@snom320-0004132480B4 Found user '80609' for '80609' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 9 Found RTP audio format 2 Found RTP audio format 3 Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 101 Peer audio RTP is at port 132.64.4.137:65030 Found description format pcmu for ID 0 Found description format pcma for ID 8 Found description format g722 for ID 9 Found description format g726-32 for ID 2 Found description format gsm for ID 3 Found description format g729 for ID 18 Found description format g723 for ID 4 Found description format telephone-event for ID 101 Capabilities: us - 0xc0008 (alaw|h261|h263), peer - audio=0x190f (g723|gsm|ulaw|alaw|g726|g729|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 132.64.4.137:65030 Looking for 80620 in huji-remote-gr (domain pbx-gr-backup.cc.huji.ac.il) list_route: hop: <--- Transmitting (no NAT) to 132.64.4.137:2059 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 132.64.4.137:2059;branch=z9hG4bK-7yny0keushsf;received=132.64.4.137;rport=2059 From: "Test" ;tag=5tui7rcix8 To: Call-ID: 3c36944683d6-ndhoic4byj9m@snom320-0004132480B4 CSeq: 2 INVITE User-Agent: Asterisk PBX SVN-trunk-r81159 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------>  -- Executing [80620@huji-remote-gr:1] NoOp("SIP/80609-087f74c0", "") in new stack  -- Executing [80620@huji-remote-gr:2] Set("SIP/80609-087f74c0", "_To=80620") in new stack  -- Executing [80620@huji-remote-gr:3] Set("SIP/80609-087f74c0", "_From=80609") in new stack  -- Executing [80620@huji-remote-gr:4] Set("SIP/80609-087f74c0", "DB(80620/LastCaller)=80609") in new stack  -- Executing [80620@huji-remote-gr:5] Set("SIP/80609-087f74c0", "DB(80609/LastCalled)=80620") in new stack  -- Executing [80620@huji-remote-gr:6] MYSQL("SIP/80609-087f74c0", "Connect connid localhost asterisk NotMe43e080 asterisk") in new stack  -- Executing [80620@huji-remote-gr:7] MYSQL("SIP/80609-087f74c0", "Query resID 1 SELECT additional_numbers from shared_lines where orig_number='80620'") in new stack  -- Executing [80620@huji-remote-gr:8] MYSQL("SIP/80609-087f74c0", "Fetch FetchId 2 aEXTEN") in new stack  -- Executing [80620@huji-remote-gr:9] NoOp("SIP/80609-087f74c0", "") in new stack  -- Executing [80620@huji-remote-gr:10] MYSQL("SIP/80609-087f74c0", "Clear 2") in new stack  -- Executing [80620@huji-remote-gr:11] MYSQL("SIP/80609-087f74c0", "Query resID 1 SELECT callerid from sip_users where name='80620'") in new stack  -- Executing [80620@huji-remote-gr:12] MYSQL("SIP/80609-087f74c0", "Fetch FetchId 2 CalledName") in new stack  -- Executing [80620@huji-remote-gr:13] MYSQL("SIP/80609-087f74c0", "Clear 2") in new stack  -- Executing [80620@huji-remote-gr:14] GotoIf("SIP/80609-087f74c0", "0?15:21") in new stack  -- Goto (huji-remote-gr,80620,21)  -- Executing [80620@huji-remote-gr:21] NoOp("SIP/80609-087f74c0", "Finish if-huji-local-4") in new stack  -- Executing [80620@huji-remote-gr:22] MYSQL("SIP/80609-087f74c0", "Disconnect 1") in new stack  -- Executing [80620@huji-remote-gr:23] Dial("SIP/80609-087f74c0", "SIP/80620,20,L(3600000:60000:30000)") in new stack  -- Limit Data for this call:  > timelimit = 3600000  > play_warning = 60000  > play_to_caller = yes  > play_to_callee = no  > warning_freq = 30000  > start_sound =  > warning_sound = timeleft  > end_sound =  == Using TOS bits 0  == Using CoS mark 5  == Using TOS bits 0  == Using CoS mark 6  == Using TOS bits 0  == Using CoS mark 5 Audio is at 132.64.9.162 port 16168 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 132.64.4.123:5060: INVITE sip:80620@132.64.4.123:5060 SIP/2.0 Via: SIP/2.0/UDP 132.64.9.162:5060;branch=z9hG4bK2caef865;rport Max-Forwards: 70 From: "Yehavi SNOM 320" ;tag=as29385b1b To: Contact: Call-ID: 7377f9415bd341b36ae8f8a22bcbafeb@132.64.9.162 CSeq: 102 INVITE User-Agent: Asterisk PBX SVN-trunk-r81159 Remote-Party-ID: "Yehavi SNOM 320" ;privacy=off;screen=no Date: Tue, 28 Aug 2007 11:51:37 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 250 v=0 o=root 1140511044 1140511044 IN IP4 132.64.4.137 s=session c=IN IP4 132.64.4.137 t=0 0 m=audio 65030 RTP/AVP 8 101 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 80620 <--- SIP read from 132.64.4.137:2059 ---> SUBSCRIBE sip:80620@pbx-gr-backup.cc.huji.ac.il;user=phone SIP/2.0 Via: SIP/2.0/UDP 132.64.4.137:2059;branch=z9hG4bK-if4dghry4t9c;rport From: ;tag=j1fbjrjd7x To: Call-ID: 3c369447a122-tyblrm7hxwun@snom320-0004132480B4 CSeq: 2 SUBSCRIBE Max-Forwards: 70 Contact: ;flow-id=1 Event: dialog;purpose=call-completion Accept: application/dialog-info+xml Authorization: Digest username="80609",realm="cc.huji.ac.il",nonce="7b162dd9",uri="sip:80620@pbx-gr-backup.cc.huji.ac.il;user=phone",response="a1af9c1a0db6b2e5399c1f1f504e15b7",algorithm=md5 Expires: 3600 Content-Length: 0 <-------------> --- (13 headers 0 lines) --- Creating new subscription Sending to 132.64.4.137 : 2059 (NAT) Found peer '80609' for '80609' from 132.64.4.137:2059 Looking for 80620 in huji-remote-gr (domain pbx-gr-backup.cc.huji.ac.il) Scheduling destruction of SIP dialog '3c369447a122-tyblrm7hxwun@snom320-0004132480B4' in 3610000 ms (Method: SUBSCRIBE) [Aug 28 14:51:37] NOTICE[4949]: chan_sip.c:15889 handle_request_subscribe: Got SUBSCRIBE for extension 80620@huji-remote-gr from 132.64.4.137, but there is no hint for that extension. <--- Transmitting (no NAT) to 132.64.4.137:2059 ---> SIP/2.0 404 Not found Via: SIP/2.0/UDP 132.64.4.137:2059;branch=z9hG4bK-if4dghry4t9c;received=132.64.4.137;rport=2059 From: ;tag=j1fbjrjd7x To: ;tag=as7f4ab585 Call-ID: 3c369447a122-tyblrm7hxwun@snom320-0004132480B4 CSeq: 2 SUBSCRIBE User-Agent: Asterisk PBX SVN-trunk-r81159 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> Really destroying SIP dialog '3c369447a122-tyblrm7hxwun@snom320-0004132480B4' Method: SUBSCRIBE <--- SIP read from 132.64.4.123:5060 ---> SIP/2.0 100 Trying Call-ID: 7377f9415bd341b36ae8f8a22bcbafeb@132.64.9.162 CSeq: 102 INVITE From: "Yehavi SNOM 320" ;tag=as29385b1b To: ;tag=6495f1f0b9bb24e Via: SIP/2.0/UDP 132.64.9.162:5060;branch=z9hG4bK2caef865;rport Content-Length: 0 User-Agent: Avaya SIP R2.2 Endpoint Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 <-------------> --- (8 headers 0 lines) --- <--- SIP read from 132.64.4.123:5060 ---> SIP/2.0 180 Ringing Call-ID: 7377f9415bd341b36ae8f8a22bcbafeb@132.64.9.162 CSeq: 102 INVITE From: "Yehavi SNOM 320" ;tag=as29385b1b To: ;tag=6495f1f0b9bb24e Via: SIP/2.0/UDP 132.64.9.162:5060;branch=z9hG4bK2caef865;rport Content-Length: 0 Contact: User-Agent: Avaya SIP R2.2 Endpoint Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 <-------------> --- (9 headers 0 lines) ---  -- SIP/80620-0880c9e8 is ringing <--- Transmitting (no NAT) to 132.64.4.137:2059 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 132.64.4.137:2059;branch=z9hG4bK-7yny0keushsf;received=132.64.4.137;rport=2059 From: "Test" ;tag=5tui7rcix8 To: ;tag=as5baefe5e Call-ID: 3c36944683d6-ndhoic4byj9m@snom320-0004132480B4 CSeq: 2 INVITE User-Agent: Asterisk PBX SVN-trunk-r81159 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> <--- SIP read from 132.64.4.123:5060 ---> SIP/2.0 200 OK Call-ID: 7377f9415bd341b36ae8f8a22bcbafeb@132.64.9.162 CSeq: 102 INVITE From: "Yehavi SNOM 320" ;tag=as29385b1b To: ;tag=6495f1f0b9bb24e Via: SIP/2.0/UDP 132.64.9.162:5060;branch=z9hG4bK2caef865;rport Content-Length: 199 Allow: NOTIFY Allow: REFER Allow: OPTIONS Allow: INVITE Allow: ACK Allow: CANCEL Allow: BYE Content-Type: application/sdp Supported: replaces Contact: User-Agent: Avaya SIP R2.2 Endpoint Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 v=0 o=MxSIP 0 1039424330 IN IP4 132.64.4.123 s=SIP Call c=IN IP4 132.64.4.123 t=0 0 m=audio 34008 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=ptime:20 a=sendrecv <-------------> --- (18 headers 10 lines) --- Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 132.64.4.123:34008 Found description format PCMA for ID 8 Found description format telephone-event for ID 101 Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 132.64.4.123:34008 --- set_address_from_contact host '132.64.4.123' list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 132.64.4.123, port 5060 Transmitting (no NAT) to 132.64.4.123:5060: ACK sip:80620@132.64.4.123:5060 SIP/2.0 Via: SIP/2.0/UDP 132.64.9.162:5060;branch=z9hG4bK0ebb0aa7;rport Max-Forwards: 70 From: "Yehavi SNOM 320" ;tag=as29385b1b To: ;tag=6495f1f0b9bb24e Contact: Call-ID: 7377f9415bd341b36ae8f8a22bcbafeb@132.64.9.162 CSeq: 102 ACK User-Agent: Asterisk PBX SVN-trunk-r81159 Remote-Party-ID: "Yehavi SNOM 320" ;privacy=off;screen=no Content-Length: 0 ---  -- SIP/80620-0880c9e8 answered SIP/80609-087f74c0 Audio is at 132.64.9.162 port 51304 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 132.64.4.137:2059 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 132.64.4.137:2059;branch=z9hG4bK-7yny0keushsf;received=132.64.4.137;rport=2059 From: "Test" ;tag=5tui7rcix8 To: ;tag=as5baefe5e Call-ID: 3c36944683d6-ndhoic4byj9m@snom320-0004132480B4 CSeq: 2 INVITE User-Agent: Asterisk PBX SVN-trunk-r81159 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 246 v=0 o=root 84551545 84551545 IN IP4 132.64.4.123 s=session c=IN IP4 132.64.4.123 t=0 0 m=audio 34008 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> <--- SIP read from 132.64.4.137:2059 ---> ACK sip:80620@132.64.9.162 SIP/2.0 Via: SIP/2.0/UDP 132.64.4.137:2059;branch=z9hG4bK-ieypfp8278nm;rport From: "Test" ;tag=5tui7rcix8 To: ;tag=as5baefe5e Call-ID: 3c36944683d6-ndhoic4byj9m@snom320-0004132480B4 CSeq: 2 ACK Max-Forwards: 70 Contact: ;flow-id=1 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- <--- SIP read from 132.64.4.137:2059 ---> SUBSCRIBE sip:80620@pbx-gr-backup.cc.huji.ac.il;user=phone SIP/2.0 Via: SIP/2.0/UDP 132.64.4.137:2059;branch=z9hG4bK-2f3eesewrit7;rport From: ;tag=j1fbjrjd7x To: Call-ID: 3c369447a122-tyblrm7hxwun@snom320-0004132480B4 CSeq: 3 SUBSCRIBE Max-Forwards: 70 Contact: ;flow-id=1 Event: dialog;purpose=call-completion Accept: application/dialog-info+xml Expires: 0 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Creating new subscription Sending to 132.64.4.137 : 2059 (NAT) Found peer '80609' for '80609' from 132.64.4.137:2059 <--- Transmitting (no NAT) to 132.64.4.137:2059 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 132.64.4.137:2059;branch=z9hG4bK-2f3eesewrit7;received=132.64.4.137;rport=2059 From: ;tag=j1fbjrjd7x To: ;tag=as6a05a87a Call-ID: 3c369447a122-tyblrm7hxwun@snom320-0004132480B4 CSeq: 3 SUBSCRIBE User-Agent: Asterisk PBX SVN-trunk-r81159 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="cc.huji.ac.il", nonce="256ff528" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '3c369447a122-tyblrm7hxwun@snom320-0004132480B4' in 32000 ms (Method: SUBSCRIBE) <--- SIP read from 132.64.4.137:2059 ---> SUBSCRIBE sip:80620@pbx-gr-backup.cc.huji.ac.il;user=phone SIP/2.0 Via: SIP/2.0/UDP 132.64.4.137:2059;branch=z9hG4bK-kqnxzk0x29dc;rport From: ;tag=j1fbjrjd7x To: Call-ID: 3c369447a122-tyblrm7hxwun@snom320-0004132480B4 CSeq: 4 SUBSCRIBE Max-Forwards: 70 Contact: ;flow-id=1 Event: dialog;purpose=call-completion Accept: application/dialog-info+xml Authorization: Digest username="80609",realm="cc.huji.ac.il",nonce="256ff528",uri="sip:80620@pbx-gr-backup.cc.huji.ac.il;user=phone",response="2fefdaf34e4caed6cafaea15a428ad94",algorithm=md5 Expires: 0 Content-Length: 0 <-------------> --- (13 headers 0 lines) --- Creating new subscription Sending to 132.64.4.137 : 2059 (NAT) Found peer '80609' for '80609' from 132.64.4.137:2059 Looking for 80620 in huji-remote-gr (domain pbx-gr-backup.cc.huji.ac.il) [Aug 28 14:51:38] NOTICE[4949]: chan_sip.c:15889 handle_request_subscribe: Got SUBSCRIBE for extension 80620@huji-remote-gr from 132.64.4.137, but there is no hint for that extension. <--- Transmitting (no NAT) to 132.64.4.137:2059 ---> SIP/2.0 404 Not found Via: SIP/2.0/UDP 132.64.4.137:2059;branch=z9hG4bK-kqnxzk0x29dc;received=132.64.4.137;rport=2059 From: ;tag=j1fbjrjd7x To: ;tag=as6a05a87a Call-ID: 3c369447a122-tyblrm7hxwun@snom320-0004132480B4 CSeq: 4 SUBSCRIBE User-Agent: Asterisk PBX SVN-trunk-r81159 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> Really destroying SIP dialog '3c369447a122-tyblrm7hxwun@snom320-0004132480B4' Method: SUBSCRIBE <--- SIP read from 132.64.4.137:2059 ---> REFER sip:7106@132.64.9.162 SIP/2.0 Via: SIP/2.0/UDP 132.64.4.137:2059;branch=z9hG4bK-o4qseeqxiauk;rport From: "Test" ;tag=mmnlvqxsvq To: "7106" ;tag=as77f172fb Call-ID: 1b133afd3c4bb85f31b6580b4ef259dc@132.64.9.162 CSeq: 2 REFER Max-Forwards: 70 Contact: ;flow-id=1 Refer-To: sip:80620@132.64.9.162?Replaces=3c36944683d6-ndhoic4byj9m%40snom320-0004132480B4%3Bto-tag%3Das5baefe5e%3Bfrom-tag%3D5tui7rcix8 Referred-By: sip:80609@pbx-gr-backup.cc.huji.ac.il User-Agent: snom320/6.2.2 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Call 1b133afd3c4bb85f31b6580b4ef259dc@132.64.9.162 got a SIP call transfer from caller: (REFER)! SIP transfer to extension 80620@huji-remote-gr by 80609@pbx-gr-backup.cc.huji.ac.il <--- Transmitting (NAT) to 132.64.4.137:2059 ---> SIP/2.0 202 Accepted Via: SIP/2.0/UDP 132.64.4.137:2059;branch=z9hG4bK-o4qseeqxiauk;received=132.64.4.137;rport=2059 From: "Test" ;tag=mmnlvqxsvq To: "7106" ;tag=as77f172fb Call-ID: 1b133afd3c4bb85f31b6580b4ef259dc@132.64.9.162 CSeq: 2 REFER User-Agent: Asterisk PBX SVN-trunk-r81159 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------>  -- Stopped music on hold on SIP/gr-pbx-link-087bbcc0 set_destination: Parsing for address/port to send to set_destination: set destination to 132.64.4.137, port 2059 Reliably Transmitting (NAT) to 132.64.4.137:2059: NOTIFY sip:80609@132.64.4.137:2059;line=5fnpelxl SIP/2.0 Via: SIP/2.0/UDP 132.64.9.162:5060;branch=z9hG4bK3f1a43f4;rport Max-Forwards: 70 From: "7106" ;tag=as77f172fb To: "Test" ;tag=mmnlvqxsvq Contact: Call-ID: 1b133afd3c4bb85f31b6580b4ef259dc@132.64.9.162 CSeq: 103 NOTIFY User-Agent: Asterisk PBX SVN-trunk-r81159 Remote-Party-ID: "7106" ;privacy=off;screen=no Event: refer;id=2 Subscription-state: terminated;reason=noresource Content-Type: message/sipfrag;version=2.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 16 SIP/2.0 200 OK --- Scheduling destruction of SIP dialog '3c36944683d6-ndhoic4byj9m@snom320-0004132480B4' in 32000 ms (Method: ACK) set_destination: Parsing for address/port to send to set_destination: set destination to 132.64.4.137, port 2059 Reliably Transmitting (no NAT) to 132.64.4.137:2059: BYE sip:80609@132.64.4.137:2059;line=5fnpelxl SIP/2.0 Via: SIP/2.0/UDP 132.64.9.162:5060;branch=z9hG4bK6067d4b3;rport Max-Forwards: 70 From: ;tag=as5baefe5e To: "Test" ;tag=5tui7rcix8 Call-ID: 3c36944683d6-ndhoic4byj9m@snom320-0004132480B4 CSeq: 102 BYE User-Agent: Asterisk PBX SVN-trunk-r81159 Content-Length: 0 --- Scheduling destruction of SIP dialog '1b133afd3c4bb85f31b6580b4ef259dc@132.64.9.162' in 32000 ms (Method: REFER)  == Spawn extension (huji-local, 80609, 23) exited non-zero on 'SIP/80609-087f74c0' <--- SIP read from 132.64.4.137:2059 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 132.64.9.162:5060;branch=z9hG4bK3f1a43f4;rport=5060 From: "7106" ;tag=as77f172fb To: "Test" ;tag=mmnlvqxsvq Call-ID: 1b133afd3c4bb85f31b6580b4ef259dc@132.64.9.162 CSeq: 103 NOTIFY Contact: ;flow-id=1 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived <--- SIP read from 132.64.4.137:2059 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 132.64.9.162:5060;branch=z9hG4bK6067d4b3;rport=5060 From: ;tag=as5baefe5e To: "Test" ;tag=5tui7rcix8 Call-ID: 3c36944683d6-ndhoic4byj9m@snom320-0004132480B4 CSeq: 102 BYE Contact: ;flow-id=1 User-Agent: snom320/6.2.2 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- SIP Response message for INCOMING dialog BYE arrived Really destroying SIP dialog '3c36944683d6-ndhoic4byj9m@snom320-0004132480B4' Method: ACK <--- SIP read from 132.64.4.137:2059 ---> BYE sip:7106@132.64.9.162 SIP/2.0 Via: SIP/2.0/UDP 132.64.4.137:2059;branch=z9hG4bK-j7g170qtp2fi;rport From: "Test" ;tag=mmnlvqxsvq To: "7106" ;tag=as77f172fb Call-ID: 1b133afd3c4bb85f31b6580b4ef259dc@132.64.9.162 CSeq: 3 BYE Max-Forwards: 70 Contact: ;flow-id=1 User-Agent: snom320/6.2.2 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 132.64.4.137 : 2059 (NAT) Scheduling destruction of SIP dialog '1b133afd3c4bb85f31b6580b4ef259dc@132.64.9.162' in 32000 ms (Method: BYE) <--- Transmitting (NAT) to 132.64.4.137:2059 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 132.64.4.137:2059;branch=z9hG4bK-j7g170qtp2fi;received=132.64.4.137;rport=2059 From: "Test" ;tag=mmnlvqxsvq To: "7106" ;tag=as77f172fb Call-ID: 1b133afd3c4bb85f31b6580b4ef259dc@132.64.9.162 CSeq: 3 BYE User-Agent: Asterisk PBX SVN-trunk-r81159 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> <--- SIP read from 128.139.26.6:58203 ---> BYE sip:80609@132.64.9.162:5060 SIP/2.0 Via: SIP/2.0/UDP 128.139.26.6:5060;branch=z9hG4bK200F479 From: ;tag=AAABF94C-49D To: ;tag=as07da1167 Date: Tue, 28 Aug 2007 12:05:03 GMT Call-ID: BE5A119A-549511DC-B1DEEAD3-358FA890@128.139.26.6 User-Agent: Cisco-SIPGateway/IOS-12.x Max-Forwards: 70 Timestamp: 1188302716 CSeq: 102 BYE Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Sending to 128.139.26.6 : 5060 (no NAT) <--- Transmitting (no NAT) to 128.139.26.6:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 128.139.26.6:5060;branch=z9hG4bK200F479;received=128.139.26.6 From: ;tag=AAABF94C-49D To: ;tag=as07da1167 Call-ID: BE5A119A-549511DC-B1DEEAD3-358FA890@128.139.26.6 CSeq: 102 BYE User-Agent: Asterisk PBX SVN-trunk-r81159 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> Scheduling destruction of SIP dialog '7377f9415bd341b36ae8f8a22bcbafeb@132.64.9.162' in 32000 ms (Method: INVITE) set_destination: Parsing for address/port to send to set_destination: set destination to 132.64.4.123, port 5060 Reliably Transmitting (no NAT) to 132.64.4.123:5060: BYE sip:80620@132.64.4.123:5060 SIP/2.0 Via: SIP/2.0/UDP 132.64.9.162:5060;branch=z9hG4bK6e020053;rport Max-Forwards: 70 From: "Yehavi SNOM 320" ;tag=as29385b1b To: ;tag=6495f1f0b9bb24e Call-ID: 7377f9415bd341b36ae8f8a22bcbafeb@132.64.9.162 CSeq: 103 BYE User-Agent: Asterisk PBX SVN-trunk-r81159 Remote-Party-ID: "Yehavi SNOM 320" ;privacy=off;screen=no Content-Length: 0 ---  == Spawn extension (huji-remote-gr, 80620, 23) exited non-zero on 'SIP/gr-pbx-link-087bbcc0' <--- SIP read from 132.64.4.123:5060 ---> SIP/2.0 200 OK Call-ID: 7377f9415bd341b36ae8f8a22bcbafeb@132.64.9.162 CSeq: 103 BYE From: "Yehavi SNOM 320" ;tag=as29385b1b To: ;tag=6495f1f0b9bb24e Via: SIP/2.0/UDP 132.64.9.162:5060;branch=z9hG4bK6e020053;rport Content-Length: 0 Supported: replaces User-Agent: Avaya SIP R2.2 Endpoint Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 <-------------> --- (9 headers 0 lines) --- Really destroying SIP dialog '7377f9415bd341b36ae8f8a22bcbafeb@132.64.9.162' Method: INVITE Really destroying SIP dialog 'BE5A119A-549511DC-B1DEEAD3-358FA890@128.139.26.6' Method: BYE stop now