=~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2011.09.13 13:23:25 =~=~=~=~=~=~=~=~=~=~=~= asterisk -r Asterisk 10.0.0-beta1, Copyright (C) 1999 - 2011 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 10.0.0-beta1 currently running on pbx (pid = 3286) pbx*CLI> Verbosity is at least 3 pbx*CLI> sip set debug on pbx*CLI> SIP Debugging re-enabled pbx*CLI>  <--- SIP read from UDP:192.168.1.128:5060 ---> <-------------> pbx*CLI> Reliably Transmitting (NAT) to 192.168.1.128:5060: OPTIONS sip:2000@192.168.1.128:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.195:5060;branch=z9hG4bK544cbe67;rport Max-Forwards: 70 From: "Unknown" ;tag=as4fabbf5d To: Contact: Call-ID: 30bf047825d5b74e237fa5bf403a24e6@192.168.1.195:5060 CSeq: 102 OPTIONS User-Agent: FPBX-2.9.0(10.0.0) Date: Tue, 13 Sep 2011 20:23:47 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- pbx*CLI>  <--- SIP read from UDP:192.168.1.128:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.195:5060;branch=z9hG4bK544cbe67;rport From: "Unknown" ;tag=as4fabbf5d To: ;tag=a7e8268e5124e104 Call-ID: 30bf047825d5b74e237fa5bf403a24e6@192.168.1.195:5060 CSeq: 102 OPTIONS User-Agent: Grandstream HT287 1.1.0.42 DevId 000b82307614 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE Supported: replaces, timer Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Really destroying SIP dialog '30bf047825d5b74e237fa5bf403a24e6@192.168.1.195:5060' Method: OPTIONS pbx*CLI>  <--- SIP read from UDP:192.168.1.128:5060 ---> INVITE sip:1001@192.168.1.195 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.128:5060;branch=z9hG4bK737dbb83fcc677fc From: ;tag=af73bc1ace4e08dd To: Contact: Supported: replaces, timer Call-ID: 8b22d8f3f8c11852@10.0.0.100 CSeq: 4499 INVITE User-Agent: Grandstream HT287 1.1.0.42 DevId 000b82307614 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE Content-Type: application/sdp Content-Length: 321 v=0 o=2000 8000 8000 IN IP4 10.0.0.100 s=SIP Call c=IN IP4 192.168.1.128 t=0 0 m=audio 5004 RTP/AVP 0 8 4 18 2 97 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:2 G726-32/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=20 a=ptime:20 <-------------> --- (13 headers 16 lines) --- == Using UDPTL TOS bits 184 == Using UDPTL CoS mark 5 pbx*CLI> Sending to 192.168.1.128:5060 (NAT) pbx*CLI> Using INVITE request as basis request - 8b22d8f3f8c11852@10.0.0.100 pbx*CLI> Found peer '2000' for '2000' from 192.168.1.128:5060 pbx*CLI>  <--- Reliably Transmitting (NAT) to 192.168.1.128:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.128:5060;branch=z9hG4bK737dbb83fcc677fc;received=192.168.1.128;rport=5060 From: ;tag=af73bc1ace4e08dd To: ;tag=as2a367eff Call-ID: 8b22d8f3f8c11852@10.0.0.100 CSeq: 4499 INVITE Server: FPBX-2.9.0(10.0.0) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="33e7d21d" Content-Length: 0 <------------> pbx*CLI> Scheduling destruction of SIP dialog '8b22d8f3f8c11852@10.0.0.100' in 6400 ms (Method: INVITE) pbx*CLI>  <--- SIP read from UDP:192.168.1.128:5060 ---> ACK sip:1001@192.168.1.195 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.128:5060;branch=z9hG4bK737dbb83fcc677fc From: ;tag=af73bc1ace4e08dd To: ;tag=as2a367eff Contact: Call-ID: 8b22d8f3f8c11852@10.0.0.100 CSeq: 4499 ACK User-Agent: Grandstream HT287 1.1.0.42 DevId 000b82307614 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE Content-Length: 0 <-------------> --- (11 headers 0 lines) --- pbx*CLI>  <--- SIP read from UDP:192.168.1.128:5060 ---> INVITE sip:1001@192.168.1.195 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.128:5060;branch=z9hG4bKbd7dfd1e2e39439d From: ;tag=af73bc1ace4e08dd To: Contact: Supported: replaces, timer Authorization: Digest username="2000", realm="asterisk", algorithm=MD5, uri="sip:1001@192.168.1.195", nonce="33e7d21d", response="291e47f4380a87257e38d69fde7ce59e" Call-ID: 8b22d8f3f8c11852@10.0.0.100 CSeq: 4500 INVITE User-Agent: Grandstream HT287 1.1.0.42 DevId 000b82307614 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE Content-Type: application/sdp Content-Length: 321 v=0 o=2000 8000 8001 IN IP4 10.0.0.100 s=SIP Call c=IN IP4 192.168.1.128 t=0 0 m=audio 5004 RTP/AVP 0 8 4 18 2 97 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:2 G726-32/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=20 a=ptime:20 <-------------> --- (14 headers 16 lines) --- Sending to 192.168.1.128:5060 (NAT) Using INVITE request as basis request - 8b22d8f3f8c11852@10.0.0.100 Found peer '2000' for '2000' from 192.168.1.128:5060 == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 4 Found RTP audio format 18 Found RTP audio format 2 Found RTP audio format 97 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format G723 for ID 4 Found audio description format G729 for ID 18 Found audio description format G726-32 for ID 2 Found audio description format iLBC for ID 97 pbx*CLI> Capabilities: us - (gsm|ulaw|alaw), peer - audio=(g723|ulaw|alaw|g726|g729|ilbc)/video=(nothing)/text=(nothing), combined - (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 192.168.1.128:5004 Looking for 1001 in from-internal (domain 192.168.1.195) list_route: hop: <--- Transmitting (NAT) to 192.168.1.128:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.128:5060;branch=z9hG4bKbd7dfd1e2e39439d;received=192.168.1.128;rport=5060 From: ;tag=af73bc1ace4e08dd To: Call-ID: 8b22d8f3f8c11852@10.0.0.100 CSeq: 4500 INVITE Server: FPBX-2.9.0(10.0.0) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> pbx*CLI>  -- Executing [1001@from-internal:1] ExecIf("SIP/2000-00000000", "0?Set(__RINGTIMER=0)") in new stack pbx*CLI>  -- Executing [1001@from-internal:2] Macro("SIP/2000-00000000", "exten-vm,novm,1001,1,1,1") in new stack pbx*CLI>  -- Executing [s@macro-exten-vm:1] Macro("SIP/2000-00000000", "user-callerid,") in new stack pbx*CLI>  -- Executing [s@macro-user-callerid:1] Set("SIP/2000-00000000", "AMPUSER=2000") in new stack pbx*CLI>  -- Executing [s@macro-user-callerid:2] GotoIf("SIP/2000-00000000", "0?report") in new stack pbx*CLI>  -- Executing [s@macro-user-callerid:3] ExecIf("SIP/2000-00000000", "1?Set(REALCALLERIDNUM=2000)") in new stack pbx*CLI>  -- Executing [s@macro-user-callerid:4] Set("SIP/2000-00000000", "AMPUSER=2000") in new stack pbx*CLI>  -- Executing [s@macro-user-callerid:5] Set("SIP/2000-00000000", "AMPUSERCIDNAME=ATA2") in new stack pbx*CLI>  -- Executing [s@macro-user-callerid:6] GotoIf("SIP/2000-00000000", "0?report") in new stack pbx*CLI>  -- Executing [s@macro-user-callerid:7] Set("SIP/2000-00000000", "AMPUSERCID=2000") in new stack pbx*CLI>  -- Executing [s@macro-user-callerid:8] Set("SIP/2000-00000000", "CALLERID(all)="ATA2" <2000>") in new stack pbx*CLI>  -- Executing [s@macro-user-callerid:9] GotoIf("SIP/2000-00000000", "0?limit") in new stack pbx*CLI>  -- Executing [s@macro-user-callerid:10] ExecIf("SIP/2000-00000000", "0?Set(GROUP(concurrency_limit)=2000)") in new stack pbx*CLI>  -- Executing [s@macro-user-callerid:11] GosubIf("SIP/2000-00000000", "7?sub-ccss,s,1(macro-exten-vm,1001)") in new stack pbx*CLI>  -- Executing [s@sub-ccss:1] ExecIf("SIP/2000-00000000", "0?Return()") in new stack pbx*CLI>  -- Executing [s@sub-ccss:2] Set("SIP/2000-00000000", "CCSS_SETUP=TRUE") in new stack pbx*CLI>  -- Executing [s@sub-ccss:3] GosubIf("SIP/2000-00000000", "7?monitor_config,1(macro-exten-vm,1001):monitor_default,1(macro-exten-vm,1001)") in new stack pbx*CLI>  -- Executing [monitor_config@sub-ccss:1] Set("SIP/2000-00000000", "CALLCOMPLETION(cc_monitor_policy)=generic") in new stack pbx*CLI>  -- Executing [monitor_config@sub-ccss:2] GotoIf("SIP/2000-00000000", "1?set_monitor") in new stack pbx*CLI>  -- Goto (sub-ccss,monitor_config,5) pbx*CLI>  -- Executing [monitor_config@sub-ccss:5] Set("SIP/2000-00000000", "CALLCOMPLETION(cc_max_monitors)=") in new stack pbx*CLI>  -- Executing [monitor_config@sub-ccss:6] Return("SIP/2000-00000000", "TRUE") in new stack pbx*CLI>  -- Executing [s@sub-ccss:4] GosubIf("SIP/2000-00000000", "7?agent_config,1:agent_default,1") in new stack pbx*CLI>  -- Executing [agent_config@sub-ccss:1] Set("SIP/2000-00000000", "CALLCOMPLETION(cc_agent_policy)=generic") in new stack pbx*CLI>  -- Executing [agent_config@sub-ccss:2] Set("SIP/2000-00000000", "CALLCOMPLETION(cc_offer_timer)=30") in new stack pbx*CLI>  -- Executing [agent_config@sub-ccss:3] Set("SIP/2000-00000000", "CALLCOMPLETION(ccbs_available_timer)=4800") in new stack pbx*CLI>  -- Executing [agent_config@sub-ccss:4] Set("SIP/2000-00000000", "CALLCOMPLETION(ccnr_available_timer)=7200") in new stack pbx*CLI>  -- Executing [agent_config@sub-ccss:5] Set("SIP/2000-00000000", "CALLCOMPLETION(cc_callback_macro)=ccss-default") in new stack pbx*CLI>  -- Executing [agent_config@sub-ccss:6] ExecIf("SIP/2000-00000000", "1?Set(CALLCOMPLETION(cc_recall_timer)=15)") in new stack pbx*CLI>  -- Executing [agent_config@sub-ccss:7] ExecIf("SIP/2000-00000000", "1?Set(CALLCOMPLETION(cc_max_agents)=5)") in new stack pbx*CLI>  -- Executing [agent_config@sub-ccss:8] ExecIf("SIP/2000-00000000", "1?Set(CALLCOMPLETION(cc_agent_dialstring)=Local/2000_1001@from-ccss-extension)") in new stack pbx*CLI>  -- Executing [agent_config@sub-ccss:9] Set("SIP/2000-00000000", "CALLCOMPLETION(cc_callback_macro)=ccss-default") in new stack pbx*CLI>  -- Executing [agent_config@sub-ccss:10] Return("SIP/2000-00000000", "") in new stack pbx*CLI>  -- Executing [s@sub-ccss:5] Set("SIP/2000-00000000", "DB(AMPUSER/2000/ccss/last_number)=1001") in new stack pbx*CLI>  -- Executing [s@sub-ccss:6] Return("SIP/2000-00000000", "") in new stack pbx*CLI>  -- Executing [s@macro-user-callerid:12] ExecIf("SIP/2000-00000000", "0?Set(CHANNEL(language)=)") in new stack pbx*CLI>  -- Executing [s@macro-user-callerid:13] GotoIf("SIP/2000-00000000", "0?continue") in new stack pbx*CLI>  -- Executing [s@macro-user-callerid:14] Set("SIP/2000-00000000", "__TTL=64") in new stack pbx*CLI>  -- Executing [s@macro-user-callerid:15] GotoIf("SIP/2000-00000000", "1?continue") in new stack pbx*CLI>  -- Goto (macro-user-callerid,s,26) pbx*CLI>  -- Executing [s@macro-user-callerid:26] Set("SIP/2000-00000000", "CALLERID(number)=2000") in new stack pbx*CLI>  -- Executing [s@macro-user-callerid:27] Set("SIP/2000-00000000", "CALLERID(name)=ATA2") in new stack pbx*CLI>  -- Executing [s@macro-user-callerid:28] Set("SIP/2000-00000000", "CHANNEL(language)=en") in new stack pbx*CLI>  -- Executing [s@macro-exten-vm:2] Set("SIP/2000-00000000", "RingGroupMethod=none") in new stack pbx*CLI>  -- Executing [s@macro-exten-vm:3] Set("SIP/2000-00000000", "__EXTTOCALL=1001") in new stack pbx*CLI>  -- Executing [s@macro-exten-vm:4] Set("SIP/2000-00000000", "__PICKUPMARK=1001") in new stack pbx*CLI>  -- Executing [s@macro-exten-vm:5] Set("SIP/2000-00000000", "RT=15") in new stack pbx*CLI>  -- Executing [s@macro-exten-vm:6] Macro("SIP/2000-00000000", "record-enable,1001,IN") in new stack pbx*CLI>  -- Executing [s@macro-record-enable:1] GotoIf("SIP/2000-00000000", "1?check") in new stack pbx*CLI>  -- Goto (macro-record-enable,s,4) pbx*CLI>  -- Executing [s@macro-record-enable:4] ExecIf("SIP/2000-00000000", "0?MacroExit()") in new stack pbx*CLI>  -- Executing [s@macro-record-enable:5] GotoIf("SIP/2000-00000000", "0?Group:OUT") in new stack pbx*CLI>  -- Goto (macro-record-enable,s,14) pbx*CLI>  -- Executing [s@macro-record-enable:14] GotoIf("SIP/2000-00000000", "1?IN") in new stack pbx*CLI>  -- Goto (macro-record-enable,s,18) pbx*CLI>  -- Executing [s@macro-record-enable:18] ExecIf("SIP/2000-00000000", "1?MacroExit()") in new stack pbx*CLI>  -- Executing [s@macro-exten-vm:7] Macro("SIP/2000-00000000", "dial-one,15,tr,1001") in new stack pbx*CLI>  -- Executing [s@macro-dial-one:1] Set("SIP/2000-00000000", "DEXTEN=1001") in new stack pbx*CLI>  -- Executing [s@macro-dial-one:2] Set("SIP/2000-00000000", "DIALSTATUS_CW=") in new stack pbx*CLI>  -- Executing [s@macro-dial-one:3] GosubIf("SIP/2000-00000000", "0?screen,1") in new stack pbx*CLI>  -- Executing [s@macro-dial-one:4] GosubIf("SIP/2000-00000000", "0?cf,1") in new stack pbx*CLI>  -- Executing [s@macro-dial-one:5] GotoIf("SIP/2000-00000000", "1?skip1") in new stack pbx*CLI>  -- Goto (macro-dial-one,s,8) pbx*CLI>  -- Executing [s@macro-dial-one:8] GotoIf("SIP/2000-00000000", "0?nodial") in new stack pbx*CLI>  -- Executing [s@macro-dial-one:9] GotoIf("SIP/2000-00000000", "0?continue") in new stack pbx*CLI>  -- Executing [s@macro-dial-one:10] Set("SIP/2000-00000000", "EXTHASCW=ENABLED") in new stack pbx*CLI>  -- Executing [s@macro-dial-one:11] GotoIf("SIP/2000-00000000", "0?next1:cwinusebusy") in new stack pbx*CLI>  -- Goto (macro-dial-one,s,23) pbx*CLI>  -- Executing [s@macro-dial-one:23] GotoIf("SIP/2000-00000000", "1?next3:continue") in new stack pbx*CLI>  -- Goto (macro-dial-one,s,24) pbx*CLI>  -- Executing [s@macro-dial-one:24] ExecIf("SIP/2000-00000000", "0?Set(DIALSTATUS_CW=BUSY)") in new stack pbx*CLI>  -- Executing [s@macro-dial-one:25] GotoIf("SIP/2000-00000000", "0?nodial") in new stack pbx*CLI>  -- Executing [s@macro-dial-one:26] GosubIf("SIP/2000-00000000", "1?dstring,1:dlocal,1") in new stack pbx*CLI>  -- Executing [dstring@macro-dial-one:1] Set("SIP/2000-00000000", "DSTRING=") in new stack pbx*CLI>  -- Executing [dstring@macro-dial-one:2] Set("SIP/2000-00000000", "DEVICES=") in new stack pbx*CLI>  -- Executing [dstring@macro-dial-one:3] ExecIf("SIP/2000-00000000", "1?Return()") in new stack pbx*CLI>  -- Executing [s@macro-dial-one:27] GotoIf("SIP/2000-00000000", "1?nodial") in new stack pbx*CLI>  -- Goto (macro-dial-one,s,46) pbx*CLI>  -- Executing [s@macro-dial-one:46] ExecIf("SIP/2000-00000000", "1?Set(DIALSTATUS=NOANSWER)") in new stack pbx*CLI>  -- Executing [s@macro-dial-one:47] NoOp("SIP/2000-00000000", "Returned from dial-one with nothing to call and DIALSTATUS: NOANSWER") in new stack pbx*CLI>  -- Executing [s@macro-dial-one:48] MacroExit("SIP/2000-00000000", "") in new stack pbx*CLI>  -- Executing [s@macro-exten-vm:8] GotoIf("SIP/2000-00000000", "0?exit") in new stack pbx*CLI>  -- Executing [s@macro-exten-vm:9] Set("SIP/2000-00000000", "SV_DIALSTATUS=NOANSWER") in new stack pbx*CLI>  -- Executing [s@macro-exten-vm:10] GosubIf("SIP/2000-00000000", "0?docfu,1") in new stack pbx*CLI>  -- Executing [s@macro-exten-vm:11] GosubIf("SIP/2000-00000000", "0?docfb,1") in new stack pbx*CLI>  -- Executing [s@macro-exten-vm:12] Set("SIP/2000-00000000", "DIALSTATUS=NOANSWER") in new stack pbx*CLI>  -- Executing [s@macro-exten-vm:13] ExecIf("SIP/2000-00000000", "1?MacroExit()") in new stack pbx*CLI>  -- Executing [1001@from-internal:3] Set("SIP/2000-00000000", "__PICKUPMARK=") in new stack pbx*CLI>  -- Executing [1001@from-internal:4] GotoIf("SIP/2000-00000000", "1?ext-fax,1001,1") in new stack pbx*CLI>  -- Goto (ext-fax,1001,1) pbx*CLI>  -- Executing [1001@ext-fax:1] NoOp("SIP/2000-00000000", "Receiving Fax for: Virtual1 (1001), From: "ATA2" <2000>") in new stack pbx*CLI>  -- Executing [1001@ext-fax:2] Set("SIP/2000-00000000", "FAX_RX_EMAIL=josh@netsteller.com") in new stack pbx*CLI>  -- Executing [1001@ext-fax:3] Goto("SIP/2000-00000000", "s,receivefax") in new stack pbx*CLI>  -- Goto (ext-fax,s,3) pbx*CLI>  -- Executing [s@ext-fax:3] StopPlayTones("SIP/2000-00000000", "") in new stack pbx*CLI>  -- Executing [s@ext-fax:4] ReceiveFAX("SIP/2000-00000000", "/var/spool/asterisk/fax/1315945434.0.tif,f") in new stack pbx*CLI>  -- Channel 'SIP/2000-00000000' receiving FAX '/var/spool/asterisk/fax/1315945434.0.tif' pbx*CLI> Audio is at 5060 pbx*CLI> Adding codec 100003 (ulaw) to SDP pbx*CLI> Adding codec 100004 (alaw) to SDP pbx*CLI>  <--- Reliably Transmitting (NAT) to 192.168.1.128:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.128:5060;branch=z9hG4bKbd7dfd1e2e39439d;received=192.168.1.128;rport=5060 From: ;tag=af73bc1ace4e08dd To: ;tag=as512f55a2 Call-ID: 8b22d8f3f8c11852@10.0.0.100 CSeq: 4500 INVITE Server: FPBX-2.9.0(10.0.0) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 209 v=0 o=root 100293917 100293917 IN IP4 192.168.1.195 s=Asterisk PBX 10.0.0-beta1 c=IN IP4 192.168.1.195 t=0 0 m=audio 10134 RTP/AVP 0 8 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=ptime:20 a=sendrecv <------------> pbx*CLI>  <--- SIP read from UDP:192.168.1.128:5060 ---> ACK sip:1001@192.168.1.195:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.128:5060;branch=z9hG4bK3f8111e1f90beb42 From: ;tag=af73bc1ace4e08dd To: ;tag=as512f55a2 Contact: Authorization: Digest username="2000", realm="asterisk", algorithm=MD5, uri="sip:1001@192.168.1.195:5060", nonce="33e7d21d", response="56fd7316a09f3de501c847f9879f36f4" Call-ID: 8b22d8f3f8c11852@10.0.0.100 CSeq: 4500 ACK User-Agent: Grandstream HT287 1.1.0.42 DevId 000b82307614 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE Content-Length: 0 <-------------> --- (12 headers 0 lines) --- pbx*CLI>  <--- SIP read from UDP:192.168.1.128:5060 ---> <-------------> pbx*CLI> set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.1.128:5060 Reliably Transmitting (NAT) to 192.168.1.128:5060: INVITE sip:2000@192.168.1.128:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.195:5060;branch=z9hG4bK3c3d308b;rport Max-Forwards: 70 From: ;tag=as512f55a2 To: ;tag=af73bc1ace4e08dd Contact: Call-ID: 8b22d8f3f8c11852@10.0.0.100 CSeq: 102 INVITE User-Agent: FPBX-2.9.0(10.0.0) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 269 v=0 o=root 100293917 100293918 IN IP4 192.168.1.195 s=Asterisk PBX 10.0.0-beta1 c=IN IP4 192.168.1.195 t=0 0 m=image 4290 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxDatagram:849 a=T38FaxUdpEC:t38UDPFEC --- pbx*CLI>  <--- SIP read from UDP:192.168.1.128:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.195:5060;branch=z9hG4bK3c3d308b;rport From: ;tag=as512f55a2 To: ;tag=af73bc1ace4e08dd Call-ID: 8b22d8f3f8c11852@10.0.0.100 CSeq: 102 INVITE User-Agent: Grandstream HT287 1.1.0.42 DevId 000b82307614 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE Content-Type: application/sdp Supported: replaces, timer Content-Length: 262 v=0 o=2000 8000 8002 IN IP4 10.0.0.100 s=SIP Call c=IN IP4 5004128 t=0 0 m=image 5004 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:9600 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:400 a=T38FaxMaxDatagram:280 a=T38FaxUdpEC:t38UDPRedundancy <-------------> --- (12 headers 12 lines) --- Got T.38 offer in SDP in dialog 8b22d8f3f8c11852@10.0.0.100 Capabilities: us - (gsm|ulaw|alaw), peer - audio=(nothing)/video=(nothing)/text=(nothing), combined - (nothing) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing) Got T.38 Re-invite without audio. Keeping RTP active during T.38 session. set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.1.128:5060 Transmitting (NAT) to 192.168.1.128:5060: ACK sip:2000@192.168.1.128:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.195:5060;branch=z9hG4bK45c98427;rport Max-Forwards: 70 From: ;tag=as512f55a2 To: ;tag=af73bc1ace4e08dd Contact: Call-ID: 8b22d8f3f8c11852@10.0.0.100 CSeq: 102 ACK User-Agent: FPBX-2.9.0(10.0.0) Content-Length: 0 --- pbx*CLI>  <--- SIP read from UDP:192.168.1.128:5060 ---> <-------------> pbx*CLI> set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.1.128:5060 Audio is at 5060 Adding codec 100003 (ulaw) to SDP Adding codec 100004 (alaw) to SDP Reliably Transmitting (NAT) to 192.168.1.128:5060: INVITE sip:2000@192.168.1.128:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.195:5060;branch=z9hG4bK088b7eda;rport Max-Forwards: 70 From: ;tag=as512f55a2 To: ;tag=af73bc1ace4e08dd Contact: Call-ID: 8b22d8f3f8c11852@10.0.0.100 CSeq: 103 INVITE User-Agent: FPBX-2.9.0(10.0.0) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 209 v=0 o=root 100293917 100293919 IN IP4 192.168.1.195 s=Asterisk PBX 10.0.0-beta1 c=IN IP4 192.168.1.195 t=0 0 m=audio 10134 RTP/AVP 0 8 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=ptime:20 a=sendrecv --- pbx*CLI>  <--- SIP read from UDP:192.168.1.128:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.195:5060;branch=z9hG4bK088b7eda;rport From: ;tag=as512f55a2 To: ;tag=af73bc1ace4e08dd Call-ID: 8b22d8f3f8c11852@10.0.0.100 CSeq: 103 INVITE User-Agent: Grandstream HT287 1.1.0.42 DevId 000b82307614 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE Content-Type: application/sdp Supported: replaces, timer Content-Length: 154 v=0 o=2000 8000 8003 IN IP4 10.0.0.100 s=SIP Call c=IN IP4 192.168.1.128 t=0 0 m=audio 5004 RTP/AVP 0 a=sendrecv a=rtpmap:0 PCMU/8000 a=ptime:20 <-------------> --- (12 headers 9 lines) --- Found RTP audio format 0 Found audio description format PCMU for ID 0 Capabilities: us - (gsm|ulaw|alaw), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 192.168.1.128:5004 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.1.128:5060 Transmitting (NAT) to 192.168.1.128:5060: ACK sip:2000@192.168.1.128:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.195:5060;branch=z9hG4bK5cfd7ef3;rport Max-Forwards: 70 From: ;tag=as512f55a2 To: ;tag=af73bc1ace4e08dd Contact: Call-ID: 8b22d8f3f8c11852@10.0.0.100 CSeq: 103 ACK User-Agent: FPBX-2.9.0(10.0.0) Content-Length: 0 --- pbx*CLI>  -- Executing [s@ext-fax:5] ExecIf("SIP/2000-00000000", "1?Set(FAXSTATUS="FAILED: error: Disconnected after permitted retries statusstr: Disconnected after permitted retries")") in new stack pbx*CLI>  -- Executing [s@ext-fax:6] Hangup("SIP/2000-00000000", "") in new stack == Spawn extension (ext-fax, s, 6) exited non-zero on 'SIP/2000-00000000' -- Executing [h@ext-fax:1] GotoIf("SIP/2000-00000000", "1?failed") in new stack -- Goto (ext-fax,h,103) -- Executing [h@ext-fax:103] NoOp("SIP/2000-00000000", "FAX "FAILED: error: Disconnected after permitted retries statusstr: Disconnected after permitted retries" for: josh@netsteller.com , From: "ATA2" <2000>") in new stack -- Executing [h@ext-fax:104] Macro("SIP/2000-00000000", "hangupcall,") in new stack -- Executing [s@macro-hangupcall:1] GotoIf("SIP/2000-00000000", "1?theend") in new stack -- Goto (macro-hangupcall,s,3) -- Executing [s@macro-hangupcall:3] Hangup("SIP/2000-00000000", "") in new stack == Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'SIP/2000-00000000' in macro 'hangupcall' == Spawn extension (ext-fax, h, 104) exited non-zero on 'SIP/2000-00000000' Scheduling destruction of SIP dialog '8b22d8f3f8c11852@10.0.0.100' in 6400 ms (Method: ACK) set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.1.128:5060 Reliably Transmitting (NAT) to 192.168.1.128:5060: BYE sip:2000@192.168.1.128:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.195:5060;branch=z9hG4bK72903f13;rport Max-Forwards: 70 From: ;tag=as512f55a2 To: ;tag=af73bc1ace4e08dd Call-ID: 8b22d8f3f8c11852@10.0.0.100 CSeq: 104 BYE User-Agent: FPBX-2.9.0(10.0.0) Proxy-Authorization: Digest username="2000", realm="asterisk", algorithm=MD5, uri="sip:192.168.1.195", nonce="", response="d8d24bdd3d5bbb81e11af9b8f53a365f" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- pbx*CLI>  <--- SIP read from UDP:192.168.1.128:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.195:5060;branch=z9hG4bK72903f13;rport From: ;tag=as512f55a2 To: ;tag=af73bc1ace4e08dd Call-ID: 8b22d8f3f8c11852@10.0.0.100 CSeq: 104 BYE User-Agent: Grandstream HT287 1.1.0.42 DevId 000b82307614 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE Supported: replaces, timer Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Really destroying SIP dialog '8b22d8f3f8c11852@10.0.0.100' Method: ACK pbx*CLI> quit ]0;root@pbx:~root@pbx:~ $ (B