Connected to Asterisk 1.4.13 currently running on sip (pid = 32474) OPTIONS sip:230@10.0.0.102:5060 SIP/2.0 Via: SIP/2.0/UDP classified:5060;branch=z9hG4bK2c1cbed0;rport From: "asterisk" ;tag=as2c002d0a To: Contact: Call-ID: 3ce6bbb2614701f31a56f14b032b5ed4@classified CSeq: 102 OPTIONS User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Date: Thu, 10 Jan 2008 01:25:27 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- sip*CLI> Retransmitting #1 (NAT) to classified:5060: OPTIONS sip:230@10.0.0.102:5060 SIP/2.0 Via: SIP/2.0/UDP classified:5060;branch=z9hG4bK2c1cbed0;rport From: "asterisk" ;tag=as2c002d0a To: Contact: Call-ID: 3ce6bbb2614701f31a56f14b032b5ed4@classified CSeq: 102 OPTIONS User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Date: Thu, 10 Jan 2008 01:25:27 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- sip*CLI> <--- SIP read from 192.168.192.24:5060 ---> INVITE sip:220@sip.domain.com SIP/2.0 Via: SIP/2.0/UDP 192.168.192.24:5060;branch=z9hG4bK1954984950;rport From: "Brillert" ;tag=200181101 To: Supported: replaces, 100rel, timer Call-ID: 1238682216@192.168.192.24 CSeq: 20 INVITE Session-Expires: 1800 Contact: Max-Forwards: 70 User-Agent: WirelessIP5000-v2.2.6/00:03:2a:00:53:68 Expires: 180 Content-Type: application/sdp Content-Length: 274 v=0 o=222 741952389 509725487 IN IP4 192.168.192.24 s=A_converstion c=IN IP4 192.168.192.24 t=0 0 m=audio 15000 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:18 G729/8000/1 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 <-------------> --- (14 headers 12 lines) --- sip*CLI> [Jan 9 20:25:30] WARNING[32541]: rtp.c:1973 ast_rtp_settos: Unable to set TOS to 184 Sending to 192.168.192.24 : 5060 (NAT) Using INVITE request as basis request - 1238682216@192.168.192.24 <--- Reliably Transmitting (no NAT) to 192.168.192.24:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.192.24:5060;branch=z9hG4bK1954984950;received=192.168.192.24;rport=5060 From: "Brillert" ;tag=200181101 To: ;tag=as5f70277f Call-ID: 1238682216@192.168.192.24 CSeq: 20 INVITE User-Agent: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="sip.domain.com", nonce="5fa37274" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '1238682216@192.168.192.24' in 32000 ms (Method: INVITE) Found user '222' sip*CLI> <--- SIP read from 192.168.192.24:5060 ---> ACK sip:220@sip.domain.com SIP/2.0 Via: SIP/2.0/UDP 192.168.192.24:5060;branch=z9hG4bK1954984950;rport From: "Brillert" ;tag=200181101 To: ;tag=as5f70277f Call-ID: 1238682216@192.168.192.24 CSeq: 20 ACK Content-Length: 0 <-------------> --- (7 headers 0 lines) --- sip*CLI> <--- SIP read from 192.168.192.24:5060 ---> INVITE sip:220@sip.domain.com SIP/2.0 Via: SIP/2.0/UDP 192.168.192.24:5060;branch=z9hG4bK743436026 From: "Brillert" ;tag=200181101 To: Supported: replaces, 100rel, timer Call-ID: 1238682216@192.168.192.24 CSeq: 21 INVITE Session-Expires: 1800 Contact: Proxy-Authorization: Digest username="222", realm="sip.domain.com", nonce="5fa37274", uri="sip:220@sip.domain.com", response="27302d6f90a91486e8435ce26621e40a", algorithm=MD5 Max-Forwards: 70 User-Agent: WirelessIP5000-v2.2.6/00:03:2a:00:53:68 Expires: 180 Content-Type: application/sdp Content-Length: 274 v=0 o=222 741952389 509725487 IN IP4 192.168.192.24 s=A_converstion c=IN IP4 192.168.192.24 t=0 0 m=audio 15000 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:18 G729/8000/1 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 <-------------> --- (15 headers 12 lines) --- Sending to 192.168.192.24 : 5060 (no NAT) Using INVITE request as basis request - 1238682216@192.168.192.24 Found user '222' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 192.168.192.24:15000 Found description format PCMU for ID 0 Found description format PCMA for ID 8 Found description format G729 for ID 18 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x4 (ulaw), peer - audio=0x10c (ulaw|alaw|g729)/video=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 192.168.192.24:15000 Looking for 220 in tenant-super (domain sip.domain.com) list_route: hop: <--- Transmitting (no NAT) to 192.168.192.24:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.192.24:5060;branch=z9hG4bK743436026;received=192.168.192.24 From: "Brillert" ;tag=200181101 To: Call-ID: 1238682216@192.168.192.24 CSeq: 21 INVITE User-Agent: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> sip*CLI> -- Executing [220@tenant-super:1] GotoIf("SIP/222-0874a7d0", "0?3") in new stack sip*CLI> -- Executing [220@tenant-super:2] Set("SIP/222-0874a7d0", "GROUP(OUTGOING)=222") in new stack sip*CLI> -- Executing [220@tenant-super:3] Set("SIP/222-0874a7d0", "OUTBOUND_GROUP_ONCE=220@INCOMING") in new stack sip*CLI> -- Executing [220@tenant-super:4] Set("SIP/222-0874a7d0", "DB(tenant/wrapup/220/lastcall)=1199928330.0") in new stack sip*CLI> -- Executing [220@tenant-super:5] Macro("SIP/222-0874a7d0", "tenant-dial|SIP/220|220|tenant|20|en||twWkKM(all-tapi^1199928330.0)||default|||vm") in new stack sip*CLI> -- Executing [s@macro-tenant-dial:1] NoOp("SIP/222-0874a7d0", ""CALL TO LOCAL EXTENSION FROM 222(WiFi)"") in new stack sip*CLI> -- Executing [s@macro-tenant-dial:2] UserEvent("SIP/222-0874a7d0", "TAPI|TAPIEVENT: LINE_NEWCALL tenant") in new stack sip*CLI> -- Executing [s@macro-tenant-dial:3] UserEvent("SIP/222-0874a7d0", "TAPI|TAPIEVENT: LINE_CALLSTATE LINECALLSTATE_OFFERING") in new stack sip*CLI> -- Executing [s@macro-tenant-dial:4] UserEvent("SIP/222-0874a7d0", "TAPI|TAPIEVENT: SET CALLERID ") in new stack sip*CLI> -- Executing [s@macro-tenant-dial:5] UserEvent("SIP/222-0874a7d0", "TAPI|TAPIEVENT: LINE_CALLINFO LINECALLINFOSTATE_CALLERID") in new stack sip*CLI> -- Executing [s@macro-tenant-dial:6] AGI("SIP/222-0874a7d0", "/var/www/asterisk/telephony/scripts/agi/dial.php") in new stack sip*CLI> -- Launched AGI Script /var/www/asterisk/telephony/scripts/agi/dial.php sip*CLI> AGI Tx >> agi_request: /var/www/asterisk/telephony/scripts/agi/dial.php sip*CLI> AGI Tx >> agi_channel: SIP/222-0874a7d0 sip*CLI> AGI Tx >> agi_language: en sip*CLI> AGI Tx >> agi_type: SIP sip*CLI> AGI Tx >> agi_uniqueid: 1199928330.0 sip*CLI> AGI Tx >> agi_callerid: 222 sip*CLI> AGI Tx >> agi_calleridname: WiFi sip*CLI> AGI Tx >> agi_callingpres: 0 sip*CLI> AGI Tx >> agi_callingani2: 0 sip*CLI> AGI Tx >> agi_callington: 0 sip*CLI> AGI Tx >> agi_callingtns: 0 sip*CLI> AGI Tx >> agi_dnid: 220 sip*CLI> AGI Tx >> agi_rdnis: unknown sip*CLI> AGI Tx >> agi_context: macro-tenant-dial sip*CLI> AGI Tx >> agi_extension: s sip*CLI> AGI Tx >> agi_priority: 6 sip*CLI> AGI Tx >> agi_enhanced: 0.0 sip*CLI> AGI Tx >> agi_accountcode: 222 sip*CLI> AGI Tx >> sip*CLI> AGI Rx << GET VARIABLE ARG1 sip*CLI> AGI Tx >> 200 result=1 (SIP/220) sip*CLI> AGI Rx << GET VARIABLE ARG2 sip*CLI> AGI Tx >> 200 result=1 (220) sip*CLI> AGI Rx << GET VARIABLE ARG3 sip*CLI> AGI Tx >> 200 result=1 (tenant) sip*CLI> AGI Rx << GET VARIABLE ARG4 AGI Tx >> 200 result=1 (20) sip*CLI> AGI Rx << GET VARIABLE ARG5 AGI Tx >> 200 result=1 (en) sip*CLI> AGI Rx << GET VARIABLE ARG6 AGI Tx >> 200 result=1 () sip*CLI> AGI Rx << GET VARIABLE ARG7 AGI Tx >> 200 result=1 (twWkKM(all-tapi^1199928330.0)) sip*CLI> AGI Rx << GET VARIABLE ARG8 AGI Tx >> 200 result=1 () sip*CLI> AGI Rx << GET VARIABLE ARG9 AGI Tx >> 200 result=1 (default) sip*CLI> AGI Rx << GET VARIABLE ARG10 sip*CLI> AGI Tx >> 200 result=1 () sip*CLI> AGI Rx << GET VARIABLE ARG11 sip*CLI> AGI Tx >> 200 result=1 () sip*CLI> AGI Rx << GET VARIABLE ARG12 sip*CLI> AGI Tx >> 200 result=1 (vm) sip*CLI> AGI Rx << GET VARIABLE MACRO_PRIORITY sip*CLI> AGI Tx >> 200 result=1 (5) sip*CLI> AGI Rx << DATABASE GET "tenant/220" "MonitorIncoming" sip*CLI> AGI Tx >> 200 result=0 sip*CLI> AGI Rx << DATABASE GET "tenant/220" "HotDesk" sip*CLI> AGI Tx >> 200 result=0 sip*CLI> AGI Rx << DATABASE GET "tenant/220" "DoNotDisturb" sip*CLI> AGI Tx >> 200 result=0 sip*CLI> AGI Rx << DATABASE GET "dnd" "SIP/220" sip*CLI> AGI Tx >> 200 result=0 sip*CLI> AGI Rx << DATABASE GET "tenant/220" "CallForward" sip*CLI> AGI Tx >> 200 result=1 (none) sip*CLI> AGI Rx << DATABASE GET "tenant/220" "CallForwardOnBusy" sip*CLI> AGI Tx >> 200 result=1 (none) sip*CLI> AGI Rx << DATABASE GET "tenant/220" "CallForwardOnNoAnswer" sip*CLI> AGI Tx >> 200 result=1 (none) sip*CLI> AGI Rx << DATABASE GET "tenant/220" "CallForwardOnUnavailable" sip*CLI> AGI Tx >> 200 result=1 (none) sip*CLI> AGI Rx << DATABASE GET "tenant/220" "CallWaiting" sip*CLI> AGI Tx >> 200 result=0 sip*CLI> AGI Rx << GET VARIABLE INCOMINGLINE sip*CLI> AGI Tx >> 200 result=0 sip*CLI> AGI Rx << DATABASE GET "tenant/220" "CallForward_Internal" sip*CLI> AGI Tx >> 200 result=0 sip*CLI> AGI Rx << DATABASE GET "tenant/220" "CallForwardOnBusy_Internal" sip*CLI> AGI Tx >> 200 result=0 sip*CLI> AGI Rx << DATABASE GET "tenant/220" "CallForwardOnNoAnswer_Internal" sip*CLI> AGI Tx >> 200 result=0 sip*CLI> AGI Rx << DATABASE GET "tenant/220" "CallForwardOnUnavailable_Internal" sip*CLI> AGI Tx >> 200 result=0 sip*CLI> AGI Rx << SET VARIABLE CHANNEL(language) "en" sip*CLI> AGI Tx >> 200 result=1 sip*CLI> AGI Rx << EXEC SetMusicOnHold default sip*CLI> -- AGI Script Executing Application: (SetMusicOnHold) Options: (default) sip*CLI> AGI Tx >> 200 result=0 sip*CLI> AGI Rx << GET VARIABLE FORCE_RECORDING sip*CLI> AGI Tx >> 200 result=0 sip*CLI> AGI Rx << SET VARIABLE TOUCH_MONITOR "1199928330.0" AGI Tx >> 200 result=1 sip*CLI> AGI Rx << GET VARIABLE LIMIT_IN_tenant_220 sip*CLI> AGI Tx >> 200 result=0 sip*CLI> == Parsing '/etc/asterisk/manager.conf': Found sip*CLI> AGI Rx << VERBOSE "Extension State for '220' is '0'." 1 sip*CLI> /var/www/asterisk/telephony/scripts/agi/dial.php: Extension State for '220' is '0'. sip*CLI> AGI Tx >> 200 result=1 sip*CLI> AGI Rx << EXEC NoOp STATUS: 0 sip*CLI> -- AGI Script Executing Application: (NoOp) Options: (STATUS:) sip*CLI> AGI Tx >> 200 result=0 sip*CLI> AGI Rx << VERBOSE "Doing the action dial with params : 220" 3 sip*CLI> -- /var/www/asterisk/telephony/scripts/agi/dial.php: Doing the action dial with params : 220 sip*CLI> AGI Tx >> 200 result=1 sip*CLI> AGI Rx << DATABASE GET "tenant/222" "CallerInternalNumber" sip*CLI> AGI Tx >> 200 result=0 sip*CLI> AGI Rx << DATABASE GET "tenant/222" "CallerInternalName" sip*CLI> AGI Tx >> 200 result=0 sip*CLI> AGI Rx << DATABASE GET "tenant/222" "HotDesk" AGI Tx >> 200 result=0 sip*CLI> AGI Rx << DATABASE PUT "tenant/220" "CallTrace" "222" sip*CLI> AGI Tx >> 200 result=1 sip*CLI> AGI Rx << VERBOSE "Dial string is SIP/220|20|twWkKM(all-tapi^1199928330.0)|." 1 sip*CLI> /var/www/asterisk/telephony/scripts/agi/dial.php: Dial string is SIP/220|20|twWkKM(all-tapi^1199928330.0)|. sip*CLI> AGI Tx >> 200 result=1 sip*CLI> AGI Rx << EXEC Dial SIP/220|20|twWkKM(all-tapi^1199928330.0)| sip*CLI> -- AGI Script Executing Application: (Dial) Options: (SIP/220|20|twWkKM(all-tapi^1199928330.0)|) sip*CLI> [Jan 9 20:25:30] WARNING[710]: rtp.c:1973 ast_rtp_settos: Unable to set TOS to 184 sip*CLI> Audio is at 192.168.192.1 port 12498 sip*CLI> Adding codec 0x4 (ulaw) to SDP sip*CLI> Adding non-codec 0x1 (telephone-event) to SDP sip*CLI> Reliably Transmitting (no NAT) to 192.168.192.27:5060: INVITE sip:220@192.168.192.27:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK5c3028e0;rport From: "WiFi" ;tag=as62bf3f8b To: Contact: Call-ID: 32158ad878e81c3c251daf81768f58c4@192.168.192.1 CSeq: 102 INVITE User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Remote-Party-ID: "WiFi" ;privacy=off;screen=no Date: Thu, 10 Jan 2008 01:25:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 242 v=0 o=root 32474 32474 IN IP4 192.168.192.1 s=session c=IN IP4 192.168.192.1 t=0 0 m=audio 12498 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 --- sip*CLI> -- Called 220 sip*CLI> Retransmitting #1 (no NAT) to 192.168.192.27:5060: INVITE sip:220@192.168.192.27:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK5c3028e0;rport From: "WiFi" ;tag=as62bf3f8b To: Contact: Call-ID: 32158ad878e81c3c251daf81768f58c4@192.168.192.1 CSeq: 102 INVITE User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Remote-Party-ID: "WiFi" ;privacy=off;screen=no Date: Thu, 10 Jan 2008 01:25:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 242 v=0 o=root 32474 32474 IN IP4 192.168.192.1 s=session c=IN IP4 192.168.192.1 t=0 0 m=audio 12498 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 --- sip*CLI> <--- SIP read from 192.168.192.27:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK5c3028e0;rport=5060;received=192.168.192.1 From: "WiFi" ;tag=as62bf3f8b To: ;tag=2698138803 Call-ID: 32158ad878e81c3c251daf81768f58c4@192.168.192.1 CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference Call-Info: ;appearance-index=1 Contact: Piano Server: Aastra 57i/2.1.2.30 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- sip*CLI> -- SIP/220-08748058 is ringing <--- Transmitting (no NAT) to 192.168.192.24:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.192.24:5060;branch=z9hG4bK743436026;received=192.168.192.24 From: "Brillert" ;tag=200181101 To: ;tag=as5a647783 Call-ID: 1238682216@192.168.192.24 CSeq: 21 INVITE User-Agent: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> sip*CLI> <--- SIP read from 192.168.192.27:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK5c3028e0;rport=5060;received=192.168.192.1 From: "WiFi" ;tag=as62bf3f8b To: ;tag=2698138803 Call-ID: 32158ad878e81c3c251daf81768f58c4@192.168.192.1 CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference Call-Info: ;appearance-index=1 Contact: Piano Server: Aastra 57i/2.1.2.30 Content-Length: 0 <-------------> --- sip*CLI> <--- SIP read from 192.168.192.27:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK5c3028e0;rport=5060;received=192.168.192.1 From: "WiFi" ;tag=as62bf3f8b To: ;tag=2698138803 Call-ID: 32158ad878e81c3c251daf81768f58c4@192.168.192.1 CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference Call-Info: ;appearance-index=1 Contact: Piano Server: Aastra 57i/2.1.2.30 Session-Expires: 90;refresher=uas Supported: timer, replaces Content-Type: application/sdp Content-Length: 237 v=0 o=MxSIP 0 0 IN IP4 192.168.192.27 s=SIP Call c=IN IP4 192.168.192.27 t=0 0 m=audio 3000 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:off - - - - a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> --- (15 headers 12 lines) --- sip*CLI> Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 192.168.192.27:3000 Found description format PCMU for ID 0 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=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 192.168.192.27:3000 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.192.27, port 5060 Transmitting (no NAT) to 192.168.192.27:5060: ACK sip:220@192.168.192.27:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK767a9008;rport From: "WiFi" ;tag=as62bf3f8b To: ;tag=2698138803 Contact: Call-ID: 32158ad878e81c3c251daf81768f58c4@192.168.192.1 CSeq: 102 ACK User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Remote-Party-ID: "WiFi" ;privacy=off;screen=no Content-Length: 0 --- sip*CLI> -- SIP/220-08748058 answered SIP/222-0874a7d0 -- Executing [s@macro-all-tapi:1] UserEvent("SIP/220-08748058", "TAPI|TAPIEVENT [~1199928330.0] LINE_CALLSTATE LINECALLSTATE_CONNECTED") in new stack Audio is at 192.168.192.1 port 10546 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 192.168.192.24:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.192.24:5060;branch=z9hG4bK743436026;received=192.168.192.24 From: "Brillert" ;tag=200181101 To: ;tag=as5a647783 Call-ID: 1238682216@192.168.192.24 CSeq: 21 INVITE User-Agent: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 242 v=0 o=root 32474 32474 IN IP4 192.168.192.1 s=session c=IN IP4 192.168.192.1 t=0 0 m=audio 10546 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 <------------> sip*CLI> <--- SIP read from 192.168.192.24:5060 ---> ACK sip:220@192.168.192.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.192.24:5060;branch=z9hG4bK1576651754 From: "Brillert" ;tag=200181101 To: ;tag=as5a647783 Call-ID: 1238682216@192.168.192.24 CSeq: 21 ACK Max-Forwards: 70 User-Agent: WirelessIP5000-v2.2.6/00:03:2a:00:53:68 Content-Length: 0 <-------------> --- Really destroying SIP dialog '3ce6bbb2614701f31a56f14b032b5ed4@classified' Method: OPTIONS sip*CLI> <--- SIP read from 192.168.192.27:5060 ---> INVITE sip:222@192.168.192.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.192.27:5060;branch=z9hG4bK8fd0874caf93af426 Max-Forwards: 70 From: ;tag=2698138803 To: "WiFi" ;tag=as62bf3f8b Call-ID: 32158ad878e81c3c251daf81768f58c4@192.168.192.1 CSeq: 24924 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference Contact: Piano Session-Expires: 90 Supported: timer, 100rel, replaces User-Agent: Aastra 57i/2.1.2.30 Content-Type: application/sdp Content-Length: 599 v=0 o=MxSIP 0 1 IN IP4 192.168.192.27 s=SIP Call c=IN IP4 192.168.192.27 t=0 0 m=audio 3000 RTP/AVP 0 18 106 107 113 110 111 112 98 97 115 96 9 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:106 BV16/8000 a=rtpmap:107 BV32/16000 a=rtpmap:113 L16/16000 a=rtpmap:110 PCMU/16000 a=rtpmap:111 PCMA/16000 a sip*CLI> =rtpmap:112 L16/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:115 G726-32/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:off - - - - a=fmtp:101 0-15 a=ptime:30 a=sendonly <-------------> --- (15 headers 25 lines) --- Sending to 192.168.192.27 : 5060 (no NAT) Found RTP audio format 0 Found RTP audio format 18 Found RTP audio format 106 Found RTP audio format 107 Found RTP audio format 113 Found RTP audio format 110 Found RTP audio format 111 Found RTP audio format 112 Found RTP audio format 98 Found RTP audio format 97 Found RTP audio format 115 Found RTP audio format 96 Found RTP audio format 9 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 192.168.192.27:3000 Found description format PCMU for ID 0 Found description format G729 for ID 18 Found description format BV16 for ID 106 Found description format BV32 for ID 107 Found description format L16 for ID 113 Found description format PCMU for ID 110 Found description format PCMA for ID 111 Found description format L16 for ID 112 Found description format G726-16 for ID 98 Found description format G726-24 for ID 97 Found description format G726-32 for ID 115 Found description format G726-40 for ID 96 Found description format G722 for ID 9 Found description format PCMA for ID 8 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x4 (ulaw), peer - audio=0x1d4c (ulaw|alaw|g726|slin|g729|ilbc|g722)/video=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 192.168.192.27:3000 <--- Transmitting (no NAT) to 192.168.192.27:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.192.27:5060;branch=z9hG4bK8fd0874caf93af426;received=192.168.192.27 From: ;tag=2698138803 To: "WiFi" ;tag=as62bf3f8b Call-ID: 32158ad878e81c3c251daf81768f58c4@192.168.192.1 CSeq: 24924 INVITE User-Agent: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> Audio is at 192.168.192.1 port 12498 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Transmitting (no NAT) to 192.168.192.27:5060 ---> SIP/2.0 200 OK sip*CLI> Via: SIP/2.0/UDP 192.168.192.27:5060;branch=z9hG4bK8fd0874caf93af426;received=192.168.192.27 From: ;tag=2698138803 To: "WiFi" ;tag=as62bf3f8b Call-ID: 32158ad878e81c3c251daf81768f58c4@192.168.192.1 CSeq: 24924 INVITE User-Agent: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 242 v=0 o=root 32474 32475 IN IP4 192.168.192.1 s=session c=IN IP4 192.168.192.1 t=0 0 m=audio 12498 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=recvonly <------------> sip*CLI> -- Started music on hold, class 'default', on channel 'SIP/222-0874a7d0' sip*CLI> <--- SIP read from 192.168.192.27:5060 ---> ACK sip:222@192.168.192.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.192.27:5060;branch=z9hG4bKe19cf725c904e9f9e Max-Forwards: 70 From: ;tag=2698138803 To: "WiFi" ;tag=as62bf3f8b Call-ID: 32158ad878e81c3c251daf81768f58c4@192.168.192.1 CSeq: 24924 ACK User-Agent: Aastra 57i/2.1.2.30 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- sip*CLI> <--- SIP read from 192.168.192.27:5060 ---> INVITE sip:700@sip.domain.com:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.192.27:5060;branch=z9hG4bK5cb87b07e6409c502 Max-Forwards: 70 From: Piano ;tag=32d0a1bffd To: 700 Call-ID: a46da0cfb8317a4c CSeq: 5309 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference Contact: Piano Session-Expires: 90 Supported: timer, 100rel, replaces User-Agent: Aastra 57i/2.1.2.30 Content-Type: application/sdp Content-Length: 599 v=0 o=MxSIP 0 0 IN IP4 192.168.192.27 s=SIP Call c=IN IP4 192.168.192.27 t=0 0 m=audio 3002 RTP/AVP 0 18 106 107 113 110 111 112 98 97 115 96 9 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:106 BV16/8000 a=rtpmap:107 BV32/16000 a=rtpmap:113 L16/16000 a=rtpmap:110 PCMU/16000 a=rtpmap:111 PCMA/16000 a=rtpmap:112 L16/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:115 G726-32/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:off - - - - a=fmtp:101 0-15 a=ptime:30 a=sendrecv <-------------> --- (15 headers 25 lines) --- [Jan 9 20:25:34] WARNING[32541]: rtp.c:1973 ast_rtp_settos: Unable to set TOS to 184 Sending to 192.168.192.27 : 5060 (no NAT) Using INVITE request as basis request - a46da0cfb8317a4c <--- Reliably Transmitting (no NAT) to 192.168.192.27:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.192.27:5060;branch=z9hG4bK5cb87b07e6409c502;received=192.168.192.27 From: Piano ;tag=32d0a1bffd To: 700 ;tag=as31d4ce67 Call-ID: a46da0cfb8317a4c CSeq: 5309 INVITE User-Agent: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="sip.domain.com", nonce="1721196e" Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'a46da0cfb8317a4c' in 32000 ms (Method: INVITE) Found user '220' sip*CLI> <--- SIP read from 192.168.192.27:5060 ---> ACK sip:700@sip.domain.com:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.192.27:5060;branch=z9hG4bK5cb87b07e6409c502 Max-Forwards: 70 From: Piano ;tag=32d0a1bffd To: 700 ;tag=as31d4ce67 Call-ID: a46da0cfb8317a4c CSeq: 5309 ACK User-Agent: Aastra 57i/2.1.2.30 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- sip*CLI> <--- SIP read from 192.168.192.27:5060 ---> INVITE sip:700@sip.domain.com:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.192.27:5060;branch=z9hG4bK2a2f4e1a43938d905 Proxy-Authorization: Digest username="220",realm="sip.domain.com",nonce="1721196e",uri="sip:700@sip.domain.com:5060",response="3add15324d5924a66fa87f4cde3a67f2",algorithm=MD5 Max-Forwards: 70 From: Piano ;tag=32d0a1bffd To: 700 Call-ID: a46da0cfb8317a4c CSeq: 5310 INVITE sip*CLI> Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference Contact: Piano Session-Expires: 90 Supported: timer, 100rel, replaces User-Agent: Aastra 57i/2.1.2.30 Content-Type: application/sdp Content-Length: 599 v=0 o=MxSIP 0 0 IN IP4 192.168.192.27 s=SIP Call c=IN IP4 192.168.192.27 t=0 0 m=audio 3002 RTP/AVP 0 18 106 107 113 110 111 112 98 97 115 96 9 8 101 a=rtpmap:0 PCMU/8000 a sip*CLI> =rtpmap:18 G729/8000 a=rtpmap:106 BV16/8000 a=rtpmap:107 BV32/16000 a=rtpmap:113 L16/16000 a=rtpmap:110 PCMU/16000 a=rtpmap:111 PCMA/16000 a=rtpmap:112 L16/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:115 G726-32/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:off - - - - a=fmtp:101 0-15 a=ptime:30 a=sendrecv <-------------> --- (16 headers 25 lines) --- Sending to 192.168.192.27 : 5060 (no NAT) sip*CLI> Using INVITE request as basis request - a46da0cfb8317a4c Found user '220' Found RTP audio format 0 Found RTP audio format 18 Found RTP audio format 106 Found RTP audio format 107 Found RTP audio format 113 Found RTP audio format 110 Found RTP audio format 111 Found RTP audio format 112 Found RTP audio format 98 Found RTP audio format 97 Found RTP audio format 115 Found RTP audio format 96 Found RTP audio format 9 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 192.168.192.27:3002 Found description format PCMU for ID 0 Found description format G729 for ID 18 Found description format BV16 for ID 106 Found description format BV32 for ID 107 Found description format L16 for ID 113 Found description format PCMU for ID 110 Found description format PCMA for ID 111 Found description format L16 for ID 112 Found description format G726-16 for ID 98 Found description format G726-24 for ID 97 Found description format G726-32 for ID 115 Found description format G726-40 for ID 96 Found description format G722 for ID 9 Found description format PCMA for ID 8 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x4 (ulaw), peer - audio=0x1d4c (ulaw|alaw|g726|slin|g729|ilbc|g722)/video=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 192.168.192.27:3002 Looking for 700 in tenant-super (domain sip.domain.com) sip*CLI> list_route: hop: <--- Transmitting (no NAT) to 192.168.192.27:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.192.27:5060;branch=z9hG4bK2a2f4e1a43938d905;received=192.168.192.27 From: Piano ;tag=32d0a1bffd To: 700 Call-ID: a46da0cfb8317a4c CSeq: 5310 INVITE User-Agent: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> sip*CLI> -- Executing [700@tenant-super:1] Park("SIP/220-086e1010", "") in new stack sip*CLI> Audio is at 192.168.192.1 port 11280 sip*CLI> Adding codec 0x4 (ulaw) to SDP sip*CLI> Adding non-codec 0x1 (telephone-event) to SDP sip*CLI> <--- Reliably Transmitting (no NAT) to 192.168.192.27:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.192.27:5060;branch=z9hG4bK2a2f4e1a43938d905;received=192.168.192.27 From: Piano ;tag=32d0a1bffd To: 700 ;tag=as040a34a1 Call-ID: a46da0cfb8317a4c CSeq: 5310 INVITE User-Agent: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 242 v=0 o=root 32474 32474 IN IP4 192.168.192.1 s=session c=IN IP4 192.168.192.1 t=0 0 m=audio 11280 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 <------------> sip*CLI> <--- SIP read from 192.168.192.27:5060 ---> ACK sip:700@192.168.192.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.192.27:5060;branch=z9hG4bK8ac70d51a2c415483 Proxy-Authorization: Digest username="220",realm="sip.domain.com",nonce="1721196e",uri="sip:700@192.168.192.1",response="93459e5dcf0bb6982ba4953b4e07a7b2",algorithm=MD5 Max-Forwards: 70 From: Piano ;tag=32d0a1bffd To: 700 ;tag=as040a34a1 Call-ID: a46da0cfb8317a4c CSeq: 5310 ACK User-Agent: Aastra 57i/2.1.2.30 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- sip*CLI> == Parked SIP/220-086e1010 on 701@parkedcalls. Will timeout back to extension [tenant-super] s, 1 in 45 seconds -- Playing 'digits/7' (language 'en') sip*CLI> -- Playing 'digits/0' (language 'en') sip*CLI> -- Playing 'digits/1' (language 'en') sip*CLI> -- Added extension '701' priority 1 to parkedcalls -- Started music on hold, class 'default', on channel 'SIP/220-086e1010' == Spawn extension (tenant-super, s, 1) exited KEEPALIVE on 'SIP/220-086e1010' sip*CLI> Reliably Transmitting (no NAT) to 192.168.192.19:2063: NOTIFY sip:233@192.168.192.19:2063;line=rngpzv8j SIP/2.0 Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK778b9da6;rport From: ;tag=as110bbcee To: ;tag=ro3em5aa41 Contact: Call-ID: 3c46f7f88191-ecjjxkove334 CSeq: 103 NOTIFY User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 207 confirmed --- sip*CLI> Extension Changed 701 new state InUse for Notify User 233 sip*CLI> <--- SIP read from 192.168.192.19:2063 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK778b9da6;rport=5060 From: ;tag=as110bbcee To: ;tag=ro3em5aa41 Call-ID: 3c46f7f88191-ecjjxkove334 CSeq: 103 NOTIFY Content-Length: 0 <-------------> --- (7 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived sip*CLI> <--- SIP read from 192.168.192.27:5060 ---> REFER sip:222@192.168.192.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.192.27:5060;branch=z9hG4bKb2b2ee915f9c8611f Max-Forwards: 70 From: ;tag=2698138803 To: "WiFi" ;tag=as62bf3f8b Call-ID: 32158ad878e81c3c251daf81768f58c4@192.168.192.1 CSeq: 24925 REFER Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference Contact: Piano Refer-To: 700 Referred-By: Supported: timer User-Agent: Aastra 57i/2.1.2.30 Content-Length: 0 <-------------> --- (15 headers 0 lines) --- Call 32158ad878e81c3c251daf81768f58c4@192.168.192.1 got a SIP call transfer from caller: (REFER)! sip*CLI> SIP transfer to extension 700@tenant-super by 220@sip.domain.com <--- Transmitting (no NAT) to 192.168.192.27:5060 ---> SIP/2.0 202 Accepted Via: SIP/2.0/UDP 192.168.192.27:5060;branch=z9hG4bKb2b2ee915f9c8611f;received=192.168.192.27 From: ;tag=2698138803 To: "WiFi" ;tag=as62bf3f8b Call-ID: 32158ad878e81c3c251daf81768f58c4@192.168.192.1 CSeq: 24925 REFER User-Agent: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> -- Stopped music on hold on SIP/220-086e1010 -- Stopped music on hold on SIP/222-0874a7d0 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.192.27, port 5060 Reliably Transmitting (no NAT) to 192.168.192.27:5060: NOTIFY sip:220@192.168.192.27:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK0e962441;rport From: "WiFi" ;tag=as62bf3f8b To: ;tag=2698138803 Contact: Call-ID: 32158ad878e81c3c251daf81768f58c4@192.168.192.1 CSeq: 103 NOTIFY User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Remote-Party-ID: "WiFi" ;privacy=off;screen=no Event: refer;id=24925 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 'a46da0cfb8317a4c' in 32000 ms (Method: ACK) set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.192.27, port 5060 Reliably Transmitting (no NAT) to 192.168.192.27:5060: BYE sip:220@192.168.192.27:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK76a85204;rport From: 700 ;tag=as040a34a1 To: Piano ;tag=32d0a1bffd Call-ID: a46da0cfb8317a4c CSeq: 102 BYE User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Content-Length: 0 --- -- Started music on hold, class 'default', on channel 'SIP/222-0874a7d0' sip*CLI> Scheduling destruction of SIP dialog '32158ad878e81c3c251daf81768f58c4@192.168.192.1' in 6400 ms (Method: REFER) AGI Tx >> 200 result=-1 == Spawn extension (macro-tenant-dial, s, 6) exited non-zero on 'SIP/220-086e1010' -- Executing [h@macro-tenant-dial:1] ResetCDR("SIP/220-086e1010", "w") in new stack sip*CLI> -- Executing [h@macro-tenant-dial:2] NoCDR("SIP/220-086e1010", "") in new stack sip*CLI> -- Executing [h@macro-tenant-dial:3] UserEvent("SIP/220-086e1010", "TAPI|TAPIEVENT: LINE_CALLSTATE LINECALLSTATE_IDLE") in new stack sip*CLI> -- Executing [h@macro-tenant-dial:4] System("SIP/220-086e1010", "/var/www/asterisk/telephony/scripts/billing/cdr.sh 1199928330.0") in new stack sip*CLI> Retransmitting #1 (no NAT) to 192.168.192.27:5060: NOTIFY sip:220@192.168.192.27:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK0e962441;rport From: "WiFi" ;tag=as62bf3f8b To: ;tag=2698138803 Contact: Call-ID: 32158ad878e81c3c251daf81768f58c4@192.168.192.1 CSeq: 103 NOTIFY User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Remote-Party-ID: "WiFi" ;privacy=off;screen=no Event: refer;id=24925 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 --- sip*CLI> Retransmitting #2 (no NAT) to 192.168.192.27:5060: NOTIFY sip:220@192.168.192.27:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK0e962441;rport From: "WiFi" ;tag=as62bf3f8b To: ;tag=2698138803 Contact: Call-ID: 32158ad878e81c3c251daf81768f58c4@192.168.192.1 CSeq: 103 NOTIFY User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Remote-Party-ID: "WiFi" ;privacy=off;screen=no Event: refer;id=24925 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 --- sip*CLI> <--- SIP read from 192.168.192.27:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK0e962441;rport=5060;received=192.168.192.1 From: "WiFi" ;tag=as62bf3f8b To: ;tag=2698138803 Call-ID: 32158ad878e81c3c251daf81768f58c4@192.168.192.1 CSeq: 103 NOTIFY Server: Aastra 57i/2.1.2.30 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived sip*CLI> <--- SIP read from 192.168.192.27:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK76a85204;rport=5060;received=192.168.192.1 From: 700 ;tag=as040a34a1 To: Piano ;tag=32d0a1bffd Call-ID: a46da0cfb8317a4c CSeq: 102 BYE Server: Aastra 57i/2.1.2.30 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- SIP Response message for INCOMING dialog BYE arrived sip*CLI> Really destroying SIP dialog 'a46da0cfb8317a4c' Method: ACK sip*CLI> <--- SIP read from 192.168.192.27:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK0e962441;rport=5060;received=192.168.192.1 From: "WiFi" ;tag=as62bf3f8b To: ;tag=2698138803 Call-ID: 32158ad878e81c3c251daf81768f58c4@192.168.192.1 CSeq: 103 NOTIFY Server: Aastra 57i/2.1.2.30 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived sip*CLI> <--- SIP read from 192.168.192.27:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK0e962441;rport=5060;received=192.168.192.1 From: "WiFi" ;tag=as62bf3f8b To: ;tag=2698138803 Call-ID: 32158ad878e81c3c251daf81768f58c4@192.168.192.1 CSeq: 103 NOTIFY Server: Aastra 57i/2.1.2.30 Content-Length: 0 --- sip*CLI> <--- SIP read from 192.168.192.24:5060 ---> BYE sip:220@192.168.192.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.192.24:5060;branch=z9hG4bK1838827424 From: "Brillert" ;tag=200181101 To: ;tag=as5a647783 Call-ID: 1238682216@192.168.192.24 CSeq: 22 BYE Max-Forwards: 70 User-Agent: WirelessIP5000-v2.2.6/00:03:2a:00:53:68 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Sending to 192.168.192.24 : 5060 (no NAT) <--- Transmitting (no NAT) to 192.168.192.24:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.192.24:5060;branch=z9hG4bK1838827424;received=192.168.192.24 From: "Brillert" ;tag=200181101 To: ;tag=as5a647783 Call-ID: 1238682216@192.168.192.24 CSeq: 22 BYE User-Agent: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> sip*CLI> -- Stopped music on hold on SIP/222-0874a7d0 == SIP/222-0874a7d0 got tired of being parked sip*CLI> Really destroying SIP dialog '1238682216@192.168.192.24' Method: BYE sip*CLI> Reliably Transmitting (no NAT) to 192.168.192.19:2063: NOTIFY sip:233@192.168.192.19:2063;line=rngpzv8j SIP/2.0 Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK410433b3;rport From: ;tag=as110bbcee To: ;tag=ro3em5aa41 Contact: Call-ID: 3c46f7f88191-ecjjxkove334 CSeq: 104 NOTIFY User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 208 terminated --- sip*CLI> Extension Changed 701 new state Idle for Notify User 233 sip*CLI> <--- SIP read from 192.168.192.19:2063 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK410433b3;rport=5060 From: ;tag=as110bbcee To: ;tag=ro3em5aa41 Call-ID: 3c46f7f88191-ecjjxkove334 CSeq: 104 NOTIFY Content-Length: 0 <-------------> --- sip*CLI> set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.192.27, port 5060 Reliably Transmitting (no NAT) to 192.168.192.27:5060: BYE sip:220@192.168.192.27:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK468a1c78;rport From: "WiFi" ;tag=as62bf3f8b To: ;tag=2698138803 Call-ID: 32158ad878e81c3c251daf81768f58c4@192.168.192.1 sip*CLI> CSeq: 104 BYE User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Remote-Party-ID: "WiFi" ;privacy=off;screen=no Content-Length: 0 --- Scheduling destruction of SIP dialog '32158ad878e81c3c251daf81768f58c4@192.168.192.1' in 6400 ms (Method: REFER) sip*CLI> <--- SIP read from 192.168.192.27:5060 ---> SIP/2.0 481 Call Leg/Transaction Does Not Exist Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK468a1c78;rport=5060;received=192.168.192.1 From: "WiFi" ;tag=as62bf3f8b To: ;tag=2698138803 Call-ID: 32158ad878e81c3c251daf81768f58c4@192.168.192.1 CSeq: 104 BYE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference Server: Aastra 57i/2.1.2.30 Content-Length: 0 --- Really destroying SIP dialog '4af1d1122a61f8917cf915fd0e10733e@classified' Method: OPTIONS sip*CLI> Reliably Transmitting (no NAT) to 192.168.192.19:2063: OPTIONS sip:233@192.168.192.19:2063;line=rngpzv8j SIP/2.0 Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK7214a87a;rport From: "asterisk" ;tag=as459a55ef To: Contact: Call-ID: 520cc4b7148993a57748d3301456723c@192.168.192.1 CSeq: 102 OPTIONS User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Date: Thu, 10 Jan 2008 01:25:49 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- sip*CLI> <--- SIP read from 192.168.192.19:2063 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK7214a87a;rport=5060 From: "asterisk" ;tag=as459a55ef To: Call-ID: 520cc4b7148993a57748d3301456723c@192.168.192.1 CSeq: 102 OPTIONS Contact: ;flow-id=1 User-Agent: snom320/7.1.30 Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, callerid Content-Length: 0 <-------------> --- (14 headers 0 lines) --- sip*CLI> Really destroying SIP dialog '520cc4b7148993a57748d3301456723c@192.168.192.1' Method: OPTIONS sip*CLI> Reliably Transmitting (no NAT) to 192.168.192.28:5060: OPTIONS sip:225@192.168.192.28;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK74946fe0;rport From: "asterisk" ;tag=as786c82f2 To: Contact: Call-ID: 30bbaa3d59a8218e0bbdac6604a5c8a2@192.168.192.1 CSeq: 102 OPTIONS User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Date: Thu, 10 Jan 2008 01:25:50 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- sip*CLI> <--- SIP read from 192.168.192.28:5060 ---> SIP/2.0 200 OK Call-ID: 30bbaa3d59a8218e0bbdac6604a5c8a2@192.168.192.1 CSeq: 102 OPTIONS From: "asterisk" ;tag=as786c82f2 To: ;tag=d5e4f731904ee7f Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK74946fe0;rport Content-Length: 0 Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO Contact: Supported: replaces User-Agent: Aastra 480i/1.4.2.3000 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45 <-------------> --- (11 headers 0 lines) --- sip*CLI> Really destroying SIP dialog '30bbaa3d59a8218e0bbdac6604a5c8a2@192.168.192.1' Method: OPTIONS sip*CLI> Reliably Transmitting (no NAT) to 192.168.192.21:5060: OPTIONS sip:226@192.168.192.21:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK099e1e7e;rport From: "asterisk" ;tag=as3e9fb6f5 To: Contact: Call-ID: 622a23f87b1da50a269fdab7636c002e@192.168.192.1 CSeq: 102 OPTIONS User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Date: Thu, 10 Jan 2008 01:25:50 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- sip*CLI> <--- SIP read from 192.168.192.21:1033 ---> SIP/2.0 405 Method Not Allowed Call-ID: 622a23f87b1da50a269fdab7636c002e@192.168.192.1 CSeq: 102 OPTIONS From: "asterisk" ;tag=as3e9fb6f5 To: ;tag=9a2a433ab74dfc5 Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK099e1e7e;rport Content-Length: 0 Allow: INVITE Allow: ACK Allow: CANCEL Allow: BYE Allow: REFER Allow: NOTIFY Allow: MESSAGE User-Agent: optiPoint 400 standard <-------------> --- (15 headers 0 lines) --- sip*CLI> Really destroying SIP dialog '622a23f87b1da50a269fdab7636c002e@192.168.192.1' Method: OPTIONS sip*CLI> Reliably Transmitting (no NAT) to 192.168.192.29:5060: OPTIONS sip:221@192.168.192.29;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK29140535;rport From: "asterisk" ;tag=as51648f24 To: Contact: Call-ID: 77e5fd69494e6a73606f83e2341d903c@192.168.192.1 CSeq: 102 OPTIONS User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Date: Thu, 10 Jan 2008 01:25:50 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- sip*CLI> Reliably Transmitting (no NAT) to 192.168.192.26:5060: OPTIONS sip:223@192.168.192.26:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK4ce04918;rport From: "asterisk" ;tag=as44d694cc To: Contact: Call-ID: 0aefc0e959536fe908fe20dc52ffc31a@192.168.192.1 CSeq: 102 OPTIONS User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Date: Thu, 10 Jan 2008 01:25:50 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- sip*CLI> <--- SIP read from 192.168.192.29:5060 ---> SIP/2.0 200 OK Call-ID: 77e5fd69494e6a73606f83e2341d903c@192.168.192.1 CSeq: 102 OPTIONS From: "asterisk" ;tag=as51648f24 To: ;tag=d3ed5f628dc3e84 Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK29140535;rport Content-Length: 0 Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO Contact: Supported: replaces User-Agent: Aastra 480i/1.4.2.3000 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45 <-------------> --- (11 headers 0 lines) --- sip*CLI> Really destroying SIP dialog '77e5fd69494e6a73606f83e2341d903c@192.168.192.1' Method: OPTIONS sip*CLI> Reliably Transmitting (no NAT) to 192.168.192.27:5060: OPTIONS sip:220@192.168.192.27:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK7bc74d68;rport From: "asterisk" ;tag=as4611178c To: Contact: Call-ID: 0812d37f502f0bb83625ef337bc04522@192.168.192.1 CSeq: 102 OPTIONS User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Date: Thu, 10 Jan 2008 01:25:50 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- sip*CLI> Reliably Transmitting (no NAT) to 192.168.192.19:2063: OPTIONS sip:133@192.168.192.19:2063;line=h8l8tsqw SIP/2.0 Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK464d247b;rport From: "asterisk" ;tag=as05816ee8 To: Contact: Call-ID: 6f5f59e40741a4e66498bca95d053f9b@192.168.192.1 CSeq: 102 OPTIONS User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Date: Thu, 10 Jan 2008 01:25:50 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- sip*CLI> <--- SIP read from 192.168.192.27:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK7bc74d68;rport=5060;received=192.168.192.1 From: "asterisk" ;tag=as4611178c To: ;tag=297395798 Call-ID: 0812d37f502f0bb83625ef337bc04522@192.168.192.1 CSeq: 102 OPTIONS Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Server: Aastra 57i/2.1.2.30 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- sip*CLI> Really destroying SIP dialog '0812d37f502f0bb83625ef337bc04522@192.168.192.1' Method: OPTIONS sip*CLI> <--- SIP read from 192.168.192.26:1080 ---> SIP/2.0 405 Method Not Allowed Call-ID: 0aefc0e959536fe908fe20dc52ffc31a@192.168.192.1 CSeq: 102 OPTIONS From: "asterisk" ;tag=as44d694cc To: ;tag=1c89294d597ebe7 Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK4ce04918;rport Content-Length: 0 Allow: INVITE Allow: ACK Allow: CANCEL Allow: BYE Allow: REFER Allow: NOTIFY Allow: MESSAGE User-Agent: optiPoint 400 standard <-------------> --- (15 headers 0 lines) --- sip*CLI> Really destroying SIP dialog '0aefc0e959536fe908fe20dc52ffc31a@192.168.192.1' Method: OPTIONS sip*CLI> <--- SIP read from 192.168.192.19:2063 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK464d247b;rport=5060 From: "asterisk" ;tag=as05816ee8 To: Call-ID: 6f5f59e40741a4e66498bca95d053f9b@192.168.192.1 CSeq: 102 OPTIONS Contact: ;flow-id=1 User-Agent: snom320/7.1.30 Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, callerid Content-Length: 0 <-------------> --- (14 headers 0 lines) --- sip*CLI> Really destroying SIP dialog '6f5f59e40741a4e66498bca95d053f9b@192.168.192.1' Method: OPTIONS sip*CLI> Reliably Transmitting (no NAT) to 192.168.192.28:5060: OPTIONS sip:125@192.168.192.28;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK7acf92ed;rport From: "asterisk" ;tag=as7b770873 To: Contact: Call-ID: 3d59b5e6671d2e0a076ffef1460fb5e7@192.168.192.1 CSeq: 102 OPTIONS User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Date: Thu, 10 Jan 2008 01:25:50 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- sip*CLI> <--- SIP read from 192.168.192.28:5060 ---> SIP/2.0 200 OK Call-ID: 3d59b5e6671d2e0a076ffef1460fb5e7@192.168.192.1 CSeq: 102 OPTIONS From: "asterisk" ;tag=as7b770873 To: ;tag=2226593b820fb8a Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK7acf92ed;rport Content-Length: 0 Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO Contact: Supported: replaces User-Agent: Aastra 480i/1.4.2.3000 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45 <-------------> --- (11 headers 0 lines) --- sip*CLI> Really destroying SIP dialog '3d59b5e6671d2e0a076ffef1460fb5e7@192.168.192.1' Method: OPTIONS sip*CLI> Reliably Transmitting (no NAT) to 192.168.192.29:5060: OPTIONS sip:121@192.168.192.29;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK7a308ebc;rport From: "asterisk" ;tag=as63b7ef1f To: Contact: Call-ID: 02eafe162cb7ccaf4c478d6c1dac1c90@192.168.192.1 CSeq: 102 OPTIONS User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Date: Thu, 10 Jan 2008 01:25:50 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- sip*CLI> <--- SIP read from 192.168.192.29:5060 ---> SIP/2.0 200 OK Call-ID: 02eafe162cb7ccaf4c478d6c1dac1c90@192.168.192.1 CSeq: 102 OPTIONS From: "asterisk" ;tag=as63b7ef1f To: ;tag=1d4f01ebf18481c Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK7a308ebc;rport Content-Length: 0 Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO Contact: Supported: replaces User-Agent: Aastra 480i/1.4.2.3000 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45 <-------------> --- (11 headers 0 lines) --- sip*CLI> Really destroying SIP dialog '02eafe162cb7ccaf4c478d6c1dac1c90@192.168.192.1' Method: OPTIONS sip*CLI> Reliably Transmitting (no NAT) to 192.168.192.27:5060: OPTIONS sip:120@192.168.192.27:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK1c728152;rport From: "asterisk" ;tag=as0091622b To: Contact: Call-ID: 56f4b2520d9077db7a5d05103f3399dd@192.168.192.1 CSeq: 102 OPTIONS User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Date: Thu, 10 Jan 2008 01:25:50 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- sip*CLI> <--- SIP read from 192.168.192.27:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK1c728152;rport=5060;received=192.168.192.1 From: "asterisk" ;tag=as0091622b To: ;tag=1143504651 Call-ID: 56f4b2520d9077db7a5d05103f3399dd@192.168.192.1 CSeq: 102 OPTIONS Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Server: Aastra 57i/2.1.2.30 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- sip*CLI> Really destroying SIP dialog '56f4b2520d9077db7a5d05103f3399dd@192.168.192.1' Method: OPTIONS sip*CLI> Reliably Transmitting (no NAT) to 192.168.192.24:5060: OPTIONS sip:222@192.168.192.24 SIP/2.0 Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK0393c2d7;rport From: "asterisk" ;tag=as7b3809ea To: Contact: Call-ID: 34ebec4e5518b07c5037f6f616490e83@192.168.192.1 CSeq: 102 OPTIONS User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Date: Thu, 10 Jan 2008 01:25:51 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- sip*CLI> <--- SIP read from 192.168.192.24:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK0393c2d7;rport=5060 From: "asterisk" ;tag=as7b3809ea To: ;tag=1341516696 Call-ID: 34ebec4e5518b07c5037f6f616490e83@192.168.192.1 CSeq: 102 OPTIONS Accept: application/sdp Content-Length: 0