sip*CLI> Retransmitting #6 (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=z9hG4bK1d931064;rport From: "asterisk" ;tag=as03bf43ea To: Contact: Call-ID: 13e2792927cbe2b204bb0b8510994f49@192.168.192.1 CSeq: 102 OPTIONS User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Date: Sat, 12 Jan 2008 00:26:07 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- sip*CLI> [Jan 11 19:26:14] WARNING[21759]: chan_sip.c:1946 retrans_pkt: Cancelling retransmit of OPTIONs (call id 13e2792927cbe2b204bb0b8510994f49@192.168.192.1) sip*CLI> <--- SIP read from 192.168.192.19:2052 ---> INVITE sip:220@domain.com SIP/2.0 Via: SIP/2.0/UDP 192.168.192.19:2052;branch=z9hG4bK-12xswpgfj9nx;rport From: "TV Room" ;tag=aqpwnh9qqe To: Call-ID: 3c2678874f57-5j5078blkji7 CSeq: 1 INVITE Max-Forwards: 70 Contact: ;flow-id=1 P-Key-Flags: keys="3" User-Agent: snom320/7.1.30 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 Session-Expires: 3600;refresher=uas Min-SE: 90 Content-Type: application/sdp Content-Length: 479 v=0 o=root 1305685344 1305685344 IN IP4 192.168.192.19 s=call c=IN IP4 192.168.192.19 t=0 0 m=audio 17006 RTP/AVP 0 8 9 2 3 18 4 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:sulbschQhlgNV2CK60ngXK+ivBCtdtXcwmSixgjA 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) --- [Jan 11 19:26:14] WARNING[21759]: rtp.c:1987 ast_rtp_settos: Unable to set TOS to 184 Sending to 192.168.192.19 : 2052 (NAT) Using INVITE request as basis request - 3c2678874f57-5j5078blkji7 <--- Reliably Transmitting (no NAT) to 192.168.192.19:2052 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.192.19:2052;branch=z9hG4bK-12xswpgfj9nx;received=192.168.192.19;rport=2052 From: "TV Room" ;tag=aqpwnh9qqe To: ;tag=as406ec2f2 Call-ID: 3c2678874f57-5j5078blkji7 CSeq: 1 INVITE User-Agent: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces P sip*CLI> roxy-Authenticate: Digest algorithm=MD5, realm="domain.com", nonce="74386ad2" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '3c2678874f57-5j5078blkji7' in 32000 ms (Method: INVITE) Found user '233' sip*CLI> <--- SIP read from 192.168.192.19:2052 ---> ACK sip:220@domain.com SIP/2.0 Via: SIP/2.0/UDP 192.168.192.19:2052;branch=z9hG4bK-12xswpgfj9nx;rport From: "TV Room" ;tag=aqpwnh9qqe To: ;tag=as406ec2f2 Call-ID: 3c2678874f57-5j5078blkji7 CSeq: 1 ACK Max-Forwards: 70 Contact: ;flow-id=1 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- sip*CLI> <--- SIP read from 192.168.192.19:2052 ---> INVITE sip:220@domain.com SIP/2.0 Via: SIP/2.0/UDP 192.168.192.19:2052;branch=z9hG4bK-whl17gpzmzpn;rport From: "TV Room" ;tag=aqpwnh9qqe To: Call-ID: 3c2678874f57-5j5078blkji7 CSeq: 2 INVITE Max-Forwards: 70 Contact: ;flow-id=1 P-Key-Flags: keys="3" User-Agent: snom320/7.1.30 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 Session-Expires: 3600;refresher=uas Min-SE: 90 Proxy-Authorization: Digest username="233",realm="domain.com",nonce="74386ad2",uri="sip:220@domain.com",response="84f7be50b5610ee5b3598a53fdf841a3",algorithm=MD5 Content-Type: application/sdp Content-Length: 479 v=0 o=root 1305685344 1305685344 IN IP4 192.168.192.19 s=call c=IN IP4 192.168.192.19 t=0 0 m=audio 17006 RTP/AVP 0 8 9 2 3 18 4 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:sulbschQhlgNV2CK60ngXK+ivBCtdtXcwmSixgjA 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 192.168.192.19 : 2052 (NAT) Using INVITE request as basis request - 3c2678874f57-5j5078blkji7 Found user '233' Found RTP audio format 0 Found RTP audio format 8 sip*CLI> 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 192.168.192.19:17006 Found audio description format pcmu for ID 0 Found audio description format pcma for ID 8 Found audio description format g722 for ID 9 Found audio description format g726-32 for ID 2 Found audio description format gsm for ID 3 Found audio description format g729 for ID 18 Found audio description format g723 for ID 4 Found audio description format telephone-event for ID 101 Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x190f (g723|gsm|ulaw|alaw|g726|g729|g722)/video=0x0 (nothing), combined - 0x104 (ulaw|g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.192.19:17006 [Jan 11 19:26:14] DEBUG[21759]: chan_sip.c:3250 update_call_counter: Call from peer '233' is 1 out of 4 Looking for 220 in tenant-super (domain domain.com) list_route: hop: <--- Transmitting (no NAT) to 192.168.192.19:2052 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.192.19:2052;branch=z9hG4bK-whl17gpzmzpn;received=192.168.192.19;rport=2052 From: "TV Room" ;tag=aqpwnh9qqe To: Call-ID: 3c2678874f57-5j5078blkji7 CSeq: 2 INVITE User-Agent: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> sip*CLI> 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=z9hG4bK4cac9bca;rport From: ;tag=as2abf2206 To: Piano ;tag=7bc8c2e0aa Contact: Call-ID: 8648874b357ecd76 CSeq: 130 NOTIFY User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 213 confirmed --- sip*CLI> Extension Changed 233 new state InUse for Notify User 220 sip*CLI> -- Executing [220@tenant-super:1] GotoIf("SIP/233-09643248", "0?3") in new stack sip*CLI> -- Executing [220@tenant-super:2] Set("SIP/233-09643248", "GROUP(OUTGOING)=233") in new stack sip*CLI> -- Executing [220@tenant-super:3] Set("SIP/233-09643248", "OUTBOUND_GROUP_ONCE=220@INCOMING") in new stack sip*CLI> -- Executing [220@tenant-super:4] Set("SIP/233-09643248", "DB(tenant/wrapup/220/lastcall)=1200097574.52") in new stack sip*CLI> -- Executing [220@tenant-super:5] Macro("SIP/233-09643248", "tenant-dial|SIP/220|220|tenant|20|en||twWkKM(all-tapi^1200097574.52)||default|||vm") in new stack sip*CLI> -- Executing [s@macro-tenant-dial:1] NoOp("SIP/233-09643248", ""CALL TO LOCAL EXTENSION FROM 233(TV Room)"") in new stack sip*CLI> [Jan 11 19:26:14] DEBUG[1728]: app_macro.c:337 _macro_exec: Executed application: NoOp sip*CLI> -- Executing [s@macro-tenant-dial:2] UserEvent("SIP/233-09643248", "TAPI|TAPIEVENT: LINE_NEWCALL tenant") in new stack sip*CLI> [Jan 11 19:26:14] DEBUG[1728]: app_macro.c:337 _macro_exec: Executed application: UserEvent sip*CLI> -- Executing [s@macro-tenant-dial:3] UserEvent("SIP/233-09643248", "TAPI|TAPIEVENT: LINE_CALLSTATE LINECALLSTATE_OFFERING") in new stack sip*CLI> [Jan 11 19:26:14] DEBUG[1728]: app_macro.c:337 _macro_exec: Executed application: UserEvent sip*CLI> -- Executing [s@macro-tenant-dial:4] UserEvent("SIP/233-09643248", "TAPI|TAPIEVENT: SET CALLERID ") in new stack sip*CLI> [Jan 11 19:26:14] DEBUG[1728]: app_macro.c:337 _macro_exec: Executed application: UserEvent sip*CLI> -- Executing [s@macro-tenant-dial:5] UserEvent("SIP/233-09643248", "TAPI|TAPIEVENT: LINE_CALLINFO LINECALLINFOSTATE_CALLERID") in new stack sip*CLI> [Jan 11 19:26:14] DEBUG[1728]: app_macro.c:337 _macro_exec: Executed application: UserEvent sip*CLI> -- Executing [s@macro-tenant-dial:6] AGI("SIP/233-09643248", "/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> <--- SIP read from 192.168.192.27:5060 ---> SIP/2.0 500 CSeq Number Out of order Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK4cac9bca;rport=5060;received=192.168.192.1 From: ;tag=as2abf2206 To: Piano ;tag=7bc8c2e0aa Call-ID: 8648874b357ecd76 CSeq: 130 NOTIFY Server: Aastra 57i/2.1.2.30 Content-Length: 0 <-------------> sip*CLI> --- (8 headers 0 lines) --- sip*CLI> SIP Response message for INCOMING dialog NOTIFY arrived sip*CLI> -- Incoming call: Got SIP response 500 "CSeq Number Out of order" back from 192.168.192.27 sip*CLI> -- AGI Script Executing Application: (SetMusicOnHold) Options: (default) sip*CLI> == Parsing '/etc/asterisk/manager.conf': Found sip*CLI> /var/www/asterisk/telephony/scripts/agi/dial.php: Extension State for '220' is '0'. sip*CLI> -- AGI Script Executing Application: (NoOp) Options: (STATUS:) sip*CLI> -- /var/www/asterisk/telephony/scripts/agi/dial.php: Doing the action dial with params : 220 sip*CLI> /var/www/asterisk/telephony/scripts/agi/dial.php: Dial string is SIP/220|20|twWkKM(all-tapi^1200097574.52)|. sip*CLI> -- AGI Script Executing Application: (Dial) Options: (SIP/220|20|twWkKM(all-tapi^1200097574.52)|) sip*CLI> [Jan 11 19:26:14] WARNING[1728]: rtp.c:1987 ast_rtp_settos: Unable to set TOS to 184 sip*CLI> [Jan 11 19:26:14] DEBUG[1728]: chan_sip.c:3250 update_call_counter: Call to peer '220' is 1 out of 4 sip*CLI> 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=z9hG4bK4c8ecf5a;rport From: ;tag=as5616628c To: Piano ;tag=5448de14ba Contact: Call-ID: 3500caa645923438 CSeq: 115 NOTIFY User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 231 early --- Extension Changed 220 new state Ringing for Notify User 220 sip*CLI> Audio is at 192.168.192.1 port 19158 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=z9hG4bK2d726eb3;rport From: "TV Room" ;tag=as55fff443 To: Contact: Call-ID: 12c6a24c458aa98d6919765b5da1bb72@192.168.192.1 CSeq: 102 INVITE User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Remote-Party-ID: "TV Room" ;privacy=off;screen=no Date: Sat, 12 Jan 2008 00:26:14 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 242 v=0 o=root 14822 14822 IN IP4 192.168.192.1 s=session c=IN IP4 192.168.192.1 t=0 0 m=audio 19158 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: NOTIFY sip:220@192.168.192.27:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK4c8ecf5a;rport From: ;tag=as5616628c To: Piano ;tag=5448de14ba Contact: Call-ID: 3500caa645923438 CSeq: 115 NOTIFY User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 231 early --- 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=z9hG4bK2d726eb3;rport From: "TV Room" ;tag=as55fff443 To: Contact: Call-ID: 12c6a24c458aa98d6919765b5da1bb72@192.168.192.1 CSeq: 102 INVITE User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Remote-Party-ID: "TV Room" ;privacy=off;screen=no Date: Sat, 12 Jan 2008 00:26:14 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 242 v=0 o=root 14822 14822 IN IP4 192.168.192.1 s=session c=IN IP4 192.168.192.1 t=0 0 m=audio 19158 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/2.0 500 CSeq Number Out of order Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK4c8ecf5a;rport=5060;received=192.168.192.1 From: ;tag=as5616628c To: Piano ;tag=5448de14ba Call-ID: 3500caa645923438 CSeq: 115 NOTIFY Server: Aastra 57i/2.1.2.30 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived -- Incoming call: Got SIP response 500 "CSeq Number Out of order" back from 192.168.192.27 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=z9hG4bK2d726eb3;rport=5060;received=192.168.192.1 From: "TV Room" ;tag=as55fff443 To: ;tag=1257917792 Call-ID: 12c6a24c458aa98d6919765b5da1bb72@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/220-096d8988 is ringing <--- Transmitting (no NAT) to 192.168.192.19:2052 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.192.19:2052;branch=z9hG4bK-whl17gpzmzpn;received=192.168.192.19;rport=2052 From: "TV Room" ;tag=aqpwnh9qqe To: ;tag=as70fb12c6 Call-ID: 3c2678874f57-5j5078blkji7 CSeq: 2 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/2.0 500 CSeq Number Out of order Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK4c8ecf5a;rport=5060;received=192.168.192.1 From: ;tag=as5616628c To: Piano ;tag=5448de14ba Call-ID: 3500caa645923438 CSeq: 115 NOTIFY Server: Aastra 57i/2.1.2.30 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived -- Incoming call: Got SIP response 500 "CSeq Number Out of order" back from 192.168.192.27 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=z9hG4bK2d726eb3;rport=5060;received=192.168.192.1 From: "TV Room" ;tag=as55fff443 To: ;tag=1257917792 Call-ID: 12c6a24c458aa98d6919765b5da1bb72@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/220-096d8988 is ringing 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=z9hG4bK6f4e0b9f;rport From: "asterisk" ;tag=as7e6ef3cd To: Contact: Call-ID: 30bc1f7919d774ec3555d4f55fe620cf@192.168.192.1 CSeq: 102 OPTIONS User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Date: Sat, 12 Jan 2008 00:26:17 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- sip*CLI> Reliably Transmitting (NAT) to 192.168.192.22:18902: OPTIONS sip:228@192.168.192.22:18902;rinstance=7bea254c9d1f3f2b SIP/2.0 Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK4c566d47;rport From: "asterisk" ;tag=as64c194d2 To: Contact: Call-ID: 51e10a6b6ac0f9f9074047724c3ac5dc@192.168.192.1 CSeq: 102 OPTIONS User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Date: Sat, 12 Jan 2008 00:26:17 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=z9hG4bK2d726eb3;rport=5060;received=192.168.192.1 From: "TV Room" ;tag=as55fff443 To: ;tag=1257917792 Call-ID: 12c6a24c458aa98d6919765b5da1bb72@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) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 192.168.192.27:3000 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 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: [Jan 11 19:26:18] DEBUG[21759]: chan_sip.c:5871 reqprep: Strict routing enforced for session 12c6a24c458aa98d6919765b5da1bb72@192.168.192.1 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=z9hG4bK2d0b1e7c;rport From: "TV Room" ;tag=as55fff443 To: ;tag=1257917792 Contact: Call-ID: 12c6a24c458aa98d6919765b5da1bb72@192.168.192.1 CSeq: 102 ACK User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Remote-Party-ID: "TV Room" ;privacy=off;screen=no Content-Length: 0 --- sip*CLI> 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=z9hG4bK0b83ff95;rport From: ;tag=as5616628c To: Piano ;tag=5448de14ba Contact: Call-ID: 3500caa645923438 CSeq: 116 NOTIFY User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 213 confirmed --- sip*CLI> Extension Changed 220 new state InUse for Notify User 220 sip*CLI> -- SIP/220-096d8988 answered SIP/233-09643248 sip*CLI> -- Executing [s@macro-all-tapi:1] UserEvent("SIP/220-096d8988", "TAPI|TAPIEVENT [~1200097574.52] LINE_CALLSTATE LINECALLSTATE_CONNECTED") in new stack sip*CLI> [Jan 11 19:26:18] DEBUG[1728]: app_macro.c:337 _macro_exec: Executed application: UserEvent sip*CLI> [Jan 11 19:26:18] DEBUG[1728]: app_dial.c:1575 dial_exec_full: Macro exited with status 0 sip*CLI> Audio is at 192.168.192.1 port 13686 sip*CLI> Adding codec 0x4 (ulaw) to SDP sip*CLI> Adding codec 0x100 (g729) to SDP sip*CLI> Adding non-codec 0x1 (telephone-event) to SDP sip*CLI> <--- Reliably Transmitting (no NAT) to 192.168.192.19:2052 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.192.19:2052;branch=z9hG4bK-whl17gpzmzpn;received=192.168.192.19;rport=2052 From: "TV Room" ;tag=aqpwnh9qqe To: ;tag=as70fb12c6 Call-ID: 3c2678874f57-5j5078blkji7 CSeq: 2 INVITE User-Agent: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 289 v=0 o=root 14822 14822 IN IP4 192.168.192.1 s=session c=IN IP4 192.168.192.1 t=0 0 m=audio 13686 RTP/AVP 0 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> sip*CLI> <--- SIP read from 192.168.192.19:2052 ---> ACK sip:220@192.168.192.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.192.19:2052;branch=z9hG4bK-eaho4vp7c2nm;rport From: "TV Room" ;tag=aqpwnh9qqe To: ;tag=as70fb12c6 Call-ID: 3c2678874f57-5j5078blkji7 CSeq: 2 ACK Max-Forwards: 70 Contact: ;flow-id=1 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- sip*CLI> <--- SIP read from 192.168.192.27:5060 ---> SIP/2.0 500 CSeq Number Out of order Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK0b83ff95;rport=5060;received=192.168.192.1 From: ;tag=as5616628c To: Piano ;tag=5448de14ba Call-ID: 3500caa645923438 CSeq: 116 NOTIFY Server: Aastra 57i/2.1.2.30 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived -- Incoming call: Got SIP response 500 "CSeq Number Out of order" back from 192.168.192.27 sip*CLI> Retransmitting #1 (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=z9hG4bK6f4e0b9f;rport From: "asterisk" ;tag=as7e6ef3cd To: Contact: Call-ID: 30bc1f7919d774ec3555d4f55fe620cf@192.168.192.1 CSeq: 102 OPTIONS User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Date: Sat, 12 Jan 2008 00:26:17 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- sip*CLI> Retransmitting #1 (NAT) to 192.168.192.22:18902: OPTIONS sip:228@192.168.192.22:18902;rinstance=7bea254c9d1f3f2b SIP/2.0 Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK4c566d47;rport From: "asterisk" ;tag=as64c194d2 To: Contact: Call-ID: 51e10a6b6ac0f9f9074047724c3ac5dc@192.168.192.1 CSeq: 102 OPTIONS User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Date: Sat, 12 Jan 2008 00:26:17 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 ---> INFO sip:233@192.168.192.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.192.27:5060;branch=z9hG4bK2f2c758139f18806e Max-Forwards: 70 From: ;tag=1257917792 To: "TV Room" ;tag=as55fff443 Call-ID: 12c6a24c458aa98d6919765b5da1bb72@192.168.192.1 CSeq: 18404 INFO Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference Contact: Piano Supported: timer User-Agent: Aastra 57i/2.1.2.30 Content-Type: application/dtmf-relay Content-Length: 25 Signal=10 Duration=160 <-------------> --- (14 headers 2 lines) --- Receiving INFO! * DTMF-relay event received: * <--- 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=z9hG4bK2f2c758139f18806e;received=192.168.192.27 From: ;tag=1257917792 To: "TV Room" ;tag=as55fff443 Call-ID: 12c6a24c458aa98d6919765b5da1bb72@192.168.192.1 CSeq: 18404 INFO User-Agent: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> sip*CLI> [Jan 11 19:26:19] DTMF[1728]: channel.c:2381 __ast_read: DTMF end '*' received on SIP/220-096d8988, duration 160 ms [Jan 11 19:26:19] DTMF[1728]: channel.c:2417 __ast_read: DTMF begin emulation of '*' with duration 160 queued on SIP/220-096d8988 sip*CLI> [Jan 11 19:26:19] DTMF[1728]: channel.c:2501 __ast_read: DTMF end emulation of '*' queued on SIP/220-096d8988 sip*CLI> Retransmitting #2 (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=z9hG4bK6f4e0b9f;rport From: "asterisk" ;tag=as7e6ef3cd To: Contact: Call-ID: 30bc1f7919d774ec3555d4f55fe620cf@192.168.192.1 CSeq: 102 OPTIONS User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Date: Sat, 12 Jan 2008 00:26:17 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 ---> INFO sip:233@192.168.192.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.192.27:5060;branch=z9hG4bKa5a5d82b386212f20 Max-Forwards: 70 From: ;tag=1257917792 To: "TV Room" ;tag=as55fff443 Call-ID: 12c6a24c458aa98d6919765b5da1bb72@192.168.192.1 CSeq: 18405 INFO Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference Contact: Piano Supported: timer User-Agent: Aastra 57i/2.1.2.30 Content-Type: application/dtmf-relay Content-Length: 24 Signal=2 Duration=160 <-------------> --- (14 headers 2 lines) --- Receiving INFO! * DTMF-relay event received: 2 <--- 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=z9hG4bKa5a5d82b386212f20;received=192.168.192.27 From: ;tag=1257917792 To: "TV Room" ;tag=as55fff443 Call-ID: 12c6a24c458aa98d6919765b5da1bb72@192.168.192.1 CSeq: 18405 INFO User-Agent: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> sip*CLI> [Jan 11 19:26:19] DTMF[1728]: channel.c:2381 __ast_read: DTMF end '2' received on SIP/220-096d8988, duration 160 ms sip*CLI> [Jan 11 19:26:19] DTMF[1728]: channel.c:2417 __ast_read: DTMF begin emulation of '2' with duration 160 queued on SIP/220-096d8988 sip*CLI> Retransmitting #2 (NAT) to 192.168.192.22:18902: OPTIONS sip:228@192.168.192.22:18902;rinstance=7bea254c9d1f3f2b SIP/2.0 Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK4c566d47;rport From: "asterisk" ;tag=as64c194d2 To: Contact: Call-ID: 51e10a6b6ac0f9f9074047724c3ac5dc@192.168.192.1 CSeq: 102 OPTIONS User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Date: Sat, 12 Jan 2008 00:26:17 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- sip*CLI> [Jan 11 19:26:19] DTMF[1728]: channel.c:2501 __ast_read: DTMF end emulation of '2' queued on SIP/220-096d8988 sip*CLI> Retransmitting #3 (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=z9hG4bK6f4e0b9f;rport From: "asterisk" ;tag=as7e6ef3cd To: Contact: Call-ID: 30bc1f7919d774ec3555d4f55fe620cf@192.168.192.1 CSeq: 102 OPTIONS User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Date: Sat, 12 Jan 2008 00:26:17 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- sip*CLI> Retransmitting #3 (NAT) to 192.168.192.22:18902: OPTIONS sip:228@192.168.192.22:18902;rinstance=7bea254c9d1f3f2b SIP/2.0 Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK4c566d47;rport From: "asterisk" ;tag=as64c194d2 To: Contact: Call-ID: 51e10a6b6ac0f9f9074047724c3ac5dc@192.168.192.1 CSeq: 102 OPTIONS User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Date: Sat, 12 Jan 2008 00:26:17 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- sip*CLI> Retransmitting #4 (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=z9hG4bK6f4e0b9f;rport From: "asterisk" ;tag=as7e6ef3cd To: Contact: Call-ID: 30bc1f7919d774ec3555d4f55fe620cf@192.168.192.1 CSeq: 102 OPTIONS User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Date: Sat, 12 Jan 2008 00:26:17 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- Really destroying SIP dialog '30bc1f7919d774ec3555d4f55fe620cf@192.168.192.1' Method: OPTIONS sip*CLI> Retransmitting #4 (NAT) to 192.168.192.22:18902: OPTIONS sip:228@192.168.192.22:18902;rinstance=7bea254c9d1f3f2b SIP/2.0 Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK4c566d47;rport From: "asterisk" ;tag=as64c194d2 To: Contact: Call-ID: 51e10a6b6ac0f9f9074047724c3ac5dc@192.168.192.1 CSeq: 102 OPTIONS User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Date: Sat, 12 Jan 2008 00:26:17 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- Really destroying SIP dialog '6f0a3d072e950fd134e3a2152cd35e0c@99.242.44.13' Method: OPTIONS sip*CLI> [Jan 11 19:26:25] DTMF[1728]: channel.c:2444 __ast_read: DTMF begin '*' received on SIP/233-09643248 [Jan 11 19:26:25] DTMF[1728]: channel.c:2454 __ast_read: DTMF begin passthrough '*' on SIP/233-09643248 sip*CLI> [Jan 11 19:26:25] DTMF[1728]: channel.c:2381 __ast_read: DTMF end '*' received on SIP/233-09643248, duration 60 ms [Jan 11 19:26:25] DTMF[1728]: channel.c:2422 __ast_read: DTMF end accepted with begin '*' on SIP/233-09643248 [Jan 11 19:26:25] DTMF[1728]: channel.c:2431 __ast_read: DTMF end '*' has duration 60 but want minimum 80, emulating on SIP/233-09643248 [Jan 11 19:26:25] DTMF[1728]: channel.c:2470 __ast_read: DTMF end emulation of '*' queued on SIP/233-09643248 sip*CLI> [Jan 11 19:26:25] DTMF[1728]: channel.c:2444 __ast_read: DTMF begin '2' received on SIP/233-09643248 [Jan 11 19:26:25] DTMF[1728]: channel.c:2454 __ast_read: DTMF begin passthrough '2' on SIP/233-09643248 sip*CLI> [Jan 11 19:26:25] DTMF[22005]: channel.c:2381 __ast_read: DTMF end '2' received on SIP/233-09643248, duration 60 ms [Jan 11 19:26:25] DTMF[22005]: channel.c:2422 __ast_read: DTMF end accepted with begin '2' on SIP/233-09643248 [Jan 11 19:26:25] DTMF[22005]: channel.c:2431 __ast_read: DTMF end '2' has duration 60 but want minimum 80, emulating on SIP/233-09643248 [Jan 11 19:26:25] DTMF[22005]: channel.c:2470 __ast_read: DTMF end emulation of '2' queued on SIP/233-09643248 sip*CLI> [Jan 11 19:26:25] DTMF[1728]: channel.c:2381 __ast_read: DTMF end '2' received on SIP/233-09643248, duration 87 ms [Jan 11 19:26:25] DTMF[1728]: channel.c:2417 __ast_read: DTMF begin emulation of '2' with duration 87 queued on SIP/233-09643248 sip*CLI> [Jan 11 19:26:25] DTMF[1728]: channel.c:2501 __ast_read: DTMF end emulation of '2' queued on SIP/233-09643248 sip*CLI> Really destroying SIP dialog '13e2792927cbe2b204bb0b8510994f49@192.168.192.1' Method: OPTIONS sip*CLI> <--- SIP read from 192.168.192.27:5060 ---> BYE sip:233@192.168.192.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.192.27:5060;branch=z9hG4bKde5a4bdf69f7edd5e Max-Forwards: 70 From: ;tag=1257917792 To: "TV Room" ;tag=as55fff443 Call-ID: 12c6a24c458aa98d6919765b5da1bb72@192.168.192.1 CSeq: 18406 BYE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference Call-Info: ;appearance-index=1 Supported: timer User-Agent: Aastra 57i/2.1.2.30 Content-Length: 0 <-------------> --- (13 headers 0 lines) --- Sending to 192.168.192.27 : 5060 (no NAT) <--- 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=z9hG4bKde5a4bdf69f7edd5e;received=192.168.192.27 From: ;tag=1257917792 To: "TV Room" ;tag=as55fff443 Call-ID: 12c6a24c458aa98d6919765b5da1bb72@192.168.192.1 CSeq: 18406 BYE User-Agent: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> sip*CLI> [Jan 11 19:26:30] DEBUG[1728]: chan_sip.c:3224 update_call_counter: Call to peer '220' removed from call limit 4 Really destroying SIP dialog '12c6a24c458aa98d6919765b5da1bb72@192.168.192.1' Method: BYE sip*CLI> 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=z9hG4bK5a5c44f3;rport From: ;tag=as5616628c To: Piano ;tag=5448de14ba Contact: Call-ID: 3500caa645923438 CSeq: 117 NOTIFY User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 214 terminated --- Extension Changed 220 new state Idle for Notify User 220 sip*CLI> -- AGI Script /var/www/asterisk/telephony/scripts/agi/dial.php completed, returning 0 [Jan 11 19:26:30] DEBUG[1728]: app_macro.c:337 _macro_exec: Executed application: AGI sip*CLI> <--- SIP read from 192.168.192.27:5060 ---> SIP/2.0 500 CSeq Number Out of order Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK5a5c44f3;rport=5060;received=192.168.192.1 From: ;tag=as5616628c To: Piano ;tag=5448de14ba Call-ID: 3500caa645923438 CSeq: 117 NOTIFY Server: Aastra 57i/2.1.2.30 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived -- Incoming call: Got SIP response 500 "CSeq Number Out of order" back from 192.168.192.27 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=z9hG4bK4bef00cd;rport From: "asterisk" ;tag=as20d2ef04 To: Contact: Call-ID: 15be8fb20db5a6197c4c55a347434cf9@192.168.192.1 CSeq: 102 OPTIONS User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Date: Sat, 12 Jan 2008 00:26:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- sip*CLI> Reliably Transmitting (NAT) to 192.168.192.22:18902: OPTIONS sip:228@192.168.192.22:18902;rinstance=7bea254c9d1f3f2b SIP/2.0 Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK5dc1ef87;rport From: "asterisk" ;tag=as0f4236cd To: Contact: Call-ID: 07ff1ef24d89c88036a33edb1444fa4d@192.168.192.1 CSeq: 102 OPTIONS User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Date: Sat, 12 Jan 2008 00:26:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- sip*CLI> Retransmitting #1 (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=z9hG4bK4bef00cd;rport From: "asterisk" ;tag=as20d2ef04 To: Contact: Call-ID: 15be8fb20db5a6197c4c55a347434cf9@192.168.192.1 CSeq: 102 OPTIONS User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Date: Sat, 12 Jan 2008 00:26:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- sip*CLI> Retransmitting #1 (NAT) to 192.168.192.22:18902: OPTIONS sip:228@192.168.192.22:18902;rinstance=7bea254c9d1f3f2b SIP/2.0 Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK5dc1ef87;rport From: "asterisk" ;tag=as0f4236cd To: Contact: Call-ID: 07ff1ef24d89c88036a33edb1444fa4d@192.168.192.1 CSeq: 102 OPTIONS User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Date: Sat, 12 Jan 2008 00:26:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- sip*CLI> <--- SIP read from 192.168.192.19:2052 ---> BYE sip:220@192.168.192.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.192.19:2052;branch=z9hG4bK-im5r92mq48rv;rport From: "TV Room" ;tag=aqpwnh9qqe To: ;tag=as70fb12c6 Call-ID: 3c2678874f57-5j5078blkji7 CSeq: 3 BYE Max-Forwards: 70 Contact: ;flow-id=1 User-Agent: snom320/7.1.30 RTP-RxStat: Total_Rx_Pkts=559,Rx_Pkts=559,Rx_Pkts_Lost=-4,Remote_Rx_Pkts_Lost=0 RTP-TxStat: Total_Tx_Pkts=734,Tx_Pkts=734,Remote_Tx_Pkts=447 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Sending to 192.168.192.19 : 2052 (NAT) <--- Transmitting (NAT) to 192.168.192.19:2052 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.192.19:2052;branch=z9hG4bK-im5r92mq48rv;received=192.168.192.19;rport=2052 From: "TV Room" ;tag=aqpwnh9qqe To: ;tag=as70fb12c6 Call-ID: 3c2678874f57-5j5078blkji7 CSeq: 3 BYE User-Agent: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> sip*CLI> [Jan 11 19:26:32] DEBUG[1728]: chan_sip.c:3224 update_call_counter: Call from peer '233' removed from call limit 4 sip*CLI> Really destroying SIP dialog '3c2678874f57-5j5078blkji7' Method: BYE sip*CLI> 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=z9hG4bK0edb486d;rport From: ;tag=as2abf2206 To: Piano ;tag=7bc8c2e0aa Contact: Call-ID: 8648874b357ecd76 CSeq: 131 NOTIFY User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 214 terminated --- sip*CLI> Extension Changed 233 new state Idle for Notify User 220 sip*CLI> <--- SIP read from 192.168.192.27:5060 ---> SIP/2.0 500 CSeq Number Out of order Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK0edb486d;rport=5060;received=192.168.192.1 From: ;tag=as2abf2206 To: Piano ;tag=7bc8c2e0aa Call-ID: 8648874b357ecd76 CSeq: 131 NOTIFY Server: Aastra 57i/2.1.2.30 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived -- Incoming call: Got SIP response 500 "CSeq Number Out of order" back from 192.168.192.27 sip*CLI> Retransmitting #2 (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=z9hG4bK4bef00cd;rport From: "asterisk" ;tag=as20d2ef04 To: Contact: Call-ID: 15be8fb20db5a6197c4c55a347434cf9@192.168.192.1 CSeq: 102 OPTIONS User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Date: Sat, 12 Jan 2008 00:26:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- sip*CLI> Retransmitting #2 (NAT) to 192.168.192.22:18902: OPTIONS sip:228@192.168.192.22:18902;rinstance=7bea254c9d1f3f2b SIP/2.0 Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK5dc1ef87;rport From: "asterisk" ;tag=as0f4236cd To: Contact: Call-ID: 07ff1ef24d89c88036a33edb1444fa4d@192.168.192.1 CSeq: 102 OPTIONS User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Date: Sat, 12 Jan 2008 00:26:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- sip*CLI> Retransmitting #3 (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=z9hG4bK4bef00cd;rport From: "asterisk" ;tag=as20d2ef04 To: Contact: Call-ID: 15be8fb20db5a6197c4c55a347434cf9@192.168.192.1 CSeq: 102 OPTIONS User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Date: Sat, 12 Jan 2008 00:26:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- sip*CLI> Retransmitting #3 (NAT) to 192.168.192.22:18902: OPTIONS sip:228@192.168.192.22:18902;rinstance=7bea254c9d1f3f2b SIP/2.0 Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK5dc1ef87;rport From: "asterisk" ;tag=as0f4236cd To: Contact: Call-ID: 07ff1ef24d89c88036a33edb1444fa4d@192.168.192.1 CSeq: 102 OPTIONS User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Date: Sat, 12 Jan 2008 00:26:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- sip*CLI> Retransmitting #4 (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=z9hG4bK4bef00cd;rport From: "asterisk" ;tag=as20d2ef04 To: Contact: Call-ID: 15be8fb20db5a6197c4c55a347434cf9@192.168.192.1 CSeq: 102 OPTIONS User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Date: Sat, 12 Jan 2008 00:26:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- Really destroying SIP dialog '15be8fb20db5a6197c4c55a347434cf9@192.168.192.1' Method: OPTIONS sip*CLI> Retransmitting #4 (NAT) to 192.168.192.22:18902: OPTIONS sip:228@192.168.192.22:18902;rinstance=7bea254c9d1f3f2b SIP/2.0 Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK5dc1ef87;rport From: "asterisk" ;tag=as0f4236cd To: Contact: Call-ID: 07ff1ef24d89c88036a33edb1444fa4d@192.168.192.1 CSeq: 102 OPTIONS User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Date: Sat, 12 Jan 2008 00:26:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- Really destroying SIP dialog '07ff1ef24d89c88036a33edb1444fa4d@192.168.192.1' Method: OPTIONS sip*CLI> ]0;root@sip:~[root@sip ~]# logout