Asterisk 1.8.11.0-2darkbasic1~squeeze, Copyright (C) 1999 - 2012 Digium, Inc. and others. Created by Mark Spencer Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. ========================================================================= Connected to Asterisk 1.8.11.0-2darkbasic1~squeeze currently running on asterisk (pid = 1751) Verbosity is at least 10 <--- SIP read from UDP:192.168.3.10:5060 ---> <-------------> -- Starting simple switch on 'DAHDI/11-1' Reliably Transmitting (NAT) to 192.168.3.10:5060: OPTIONS sip:152@192.168.3.10:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.1:5060;branch=z9hG4bK1853fc51;rport Max-Forwards: 70 From: "asterisk" ;tag=as2a8d68e7 To: Contact: Call-ID: 063ee82a60adf3661275651269437b28@192.168.3.1:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.11.0-2darkbasic1~squeeze Date: Mon, 09 Apr 2012 19:55:44 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- <--- SIP read from UDP:192.168.3.10:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.1:5060;branch=z9hG4bK1853fc51;rport=5060 From: "asterisk" ;tag=as2a8d68e7 To: ;tag=2633592200 Call-ID: 063ee82a60adf3661275651269437b28@192.168.3.1:5060 CSeq: 102 OPTIONS Contact: "152" Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO Accept: application/sdp,application/dtmf-relay Accept-Encoding: identity Accept-Language: en Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Really destroying SIP dialog '063ee82a60adf3661275651269437b28@192.168.3.1:5060' Method: OPTIONS Reliably Transmitting (NAT) to 192.168.3.12:2048: OPTIONS sip:155@192.168.3.12:2048;line=qr8mjt5c SIP/2.0 Via: SIP/2.0/UDP 192.168.3.1:5060;branch=z9hG4bK03c45233;rport Max-Forwards: 70 From: "asterisk" ;tag=as3ba7b8c4 To: Contact: Call-ID: 5e48547624427960154c22e93a32a0b9@192.168.3.1:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.11.0-2darkbasic1~squeeze Date: Mon, 09 Apr 2012 19:55:47 GMT llow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- Retransmitting #1 (NAT) to 192.168.3.12:2048: OPTIONS sip:155@192.168.3.12:2048;line=qr8mjt5c SIP/2.0 Via: SIP/2.0/UDP 192.168.3.1:5060;branch=z9hG4bK03c45233;rport Max-Forwards: 70 From: "asterisk" ;tag=as3ba7b8c4 To: Contact: Call-ID: 5e48547624427960154c22e93a32a0b9@192.168.3.1:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.11.0-2darkbasic1~squeeze Date: Mon, 09 Apr 2012 19:55:47 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- Retransmitting #2 (NAT) to 192.168.3.12:2048: OPTIONS sip:155@192.168.3.12:2048;line=qr8mjt5c SIP/2.0 Via: SIP/2.0/UDP 192.168.3.1:5060;branch=z9hG4bK03c45233;rport Max-Forwards: 70 From: "asterisk" ;tag=as3ba7b8c4 To: Contact: Call-ID: 5e48547624427960154c22e93a32a0b9@192.168.3.1:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.11.0-2darkbasic1~squeeze Date: Mon, 09 Apr 2012 19:55:47 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- Retransmitting #3 (NAT) to 192.168.3.12:2048: OPTIONS sip:155@192.168.3.12:2048;line=qr8mjt5c SIP/2.0 Via: SIP/2.0/UDP 192.168.3.1:5060;branch=z9hG4bK03c45233;rport Max-Forwards: 70 From: "asterisk" ;tag=as3ba7b8c4 To: Contact: Call-ID: 5e48547624427960154c22e93a32a0b9@192.168.3.1:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.11.0-2darkbasic1~squeeze Date: Mon, 09 Apr 2012 19:55:47 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- Retransmitting #4 (NAT) to 192.168.3.12:2048: OPTIONS sip:155@192.168.3.12:2048;line=qr8mjt5c SIP/2.0 Via: SIP/2.0/UDP 192.168.3.1:5060;branch=z9hG4bK03c45233;rport Max-Forwards: 70 From: "asterisk" ;tag=as3ba7b8c4 To: Contact: Call-ID: 5e48547624427960154c22e93a32a0b9@192.168.3.1:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.11.0-2darkbasic1~squeeze Date: Mon, 09 Apr 2012 19:55:47 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- Really destroying SIP dialog '5e48547624427960154c22e93a32a0b9@192.168.3.1:5060' Method: OPTIONS <--- SIP read from UDP:192.168.3.4:5060 ---> <-------------> Reliably Transmitting (NAT) to 192.168.3.4:5060: OPTIONS sip:102@192.168.3.4:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.1:5060;branch=z9hG4bK08a30087;rport Max-Forwards: 70 From: "asterisk" ;tag=as63189fda To: Contact: Call-ID: 419478c27a793662166d3ef60bb49fcd@192.168.3.1:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.11.0-2darkbasic1~squeeze Date: Mon, 09 Apr 2012 19:55:52 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- <--- SIP read from UDP:192.168.3.4:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.1:5060;branch=z9hG4bK08a30087;rport=5060 From: "asterisk" ;tag=as63189fda To: ;tag=1747108023 Call-ID: 419478c27a793662166d3ef60bb49fcd@192.168.3.1:5060 CSeq: 102 OPTIONS Contact: "102" Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO Accept: application/sdp,application/dtmf-relay Accept-Encoding: identity Accept-Language: en Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Really destroying SIP dialog '419478c27a793662166d3ef60bb49fcd@192.168.3.1:5060' Method: OPTIONS -- Executing [071914032@faxes:1] Set("DAHDI/11-1", "FAXOPT(gateway)=yes") in new stack -- Executing [071914032@faxes:2] Dial("DAHDI/11-1", "SIP/eutelia/071914032,60") in new stack == Using SIP RTP CoS mark 5 Audio is at 18806 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 83.211.227.21:5060: INVITE sip:071914032@voip.eutelia.it SIP/2.0 Via: SIP/2.0/UDP 2.119.245.40:5060;branch=z9hG4bK031f3b23;rport Max-Forwards: 70 From: "asterisk" ;tag=as5b3a0f26 To: Contact: Call-ID: 020d85de7f95ac9073059f40754d5e1e@2.119.245.40:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.11.0-2darkbasic1~squeeze Date: Mon, 09 Apr 2012 19:55:53 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 255 v=0 o=root 985362844 985362844 IN IP4 2.119.245.40 s=Asterisk PBX 1.8.11.0-2darkbasic1~squeeze c=IN IP4 2.119.245.40 t=0 0 m=audio 18806 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- -- Called SIP/eutelia/071914032 <--- SIP read from UDP:83.211.227.21:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 2.119.245.40:5060;branch=z9hG4bK031f3b23;rport=5060 From: "asterisk" ;tag=as5b3a0f26 To: ;tag=c040a69dfc7733bdec8c921a7a9f2d3a.0501 Call-ID: 020d85de7f95ac9073059f40754d5e1e@2.119.245.40:5060 CSeq: 102 INVITE Proxy-Authenticate: Digest realm="voip.eutelia.it", nonce="4f833f0567b572b037a33dd74f8b73cfb7f8d5d2", qop="auth" Server: SPS EUT RM GW 03 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- set_destination: Parsing for address/port to send to set_destination: set destination to 83.211.227.21:5060 Transmitting (NAT) to 83.211.227.21:5060: ACK sip:071914032@voip.eutelia.it SIP/2.0 Via: SIP/2.0/UDP 2.119.245.40:5060;branch=z9hG4bK031f3b23;rport Max-Forwards: 70 From: "asterisk" ;tag=as5b3a0f26 To: ;tag=c040a69dfc7733bdec8c921a7a9f2d3a.0501 Contact: Call-ID: 020d85de7f95ac9073059f40754d5e1e@2.119.245.40:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 1.8.11.0-2darkbasic1~squeeze Content-Length: 0 --- Audio is at 18806 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 83.211.227.21:5060: INVITE sip:071914032@voip.eutelia.it SIP/2.0 Via: SIP/2.0/UDP 2.119.245.40:5060;branch=z9hG4bK5cc667dc;rport Max-Forwards: 70 From: "asterisk" ;tag=as5b3a0f26 To: Contact: Call-ID: 020d85de7f95ac9073059f40754d5e1e@2.119.245.40:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 1.8.11.0-2darkbasic1~squeeze Proxy-Authorization: Digest username="0719206651", realm="voip.eutelia.it", algorithm=MD5, uri="sip:071914032@voip.eutelia.it", nonce="4f833f0567b572b037a33dd74f8b73cfb7f8d5d2", response="8c2fe9aea75af51445b370956a3f7381", qop=auth, cnonce="07ffc349", nc=00000001 Date: Mon, 09 Apr 2012 19:55:53 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 255 v=0 o=root 985362844 985362845 IN IP4 2.119.245.40 s=Asterisk PBX 1.8.11.0-2darkbasic1~squeeze c=IN IP4 2.119.245.40 t=0 0 m=audio 18806 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- <--- SIP read from UDP:83.211.227.21:5060 ---> SIP/2.0 100 Giving a try Via: SIP/2.0/UDP 2.119.245.40:5060;branch=z9hG4bK5cc667dc;rport=5060 From: "asterisk" ;tag=as5b3a0f26 To: Call-ID: 020d85de7f95ac9073059f40754d5e1e@2.119.245.40:5060 CSeq: 103 INVITE Server: SPS EUT RM GW 03 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- -- Span 2: Extension 71914032@from-isdn-2 does not exist. Rejecting call from ''. <--- SIP read from UDP:83.211.227.21:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 2.119.245.40:5060;received=2.119.245.40;branch=z9hG4bK5cc667dc;rport=5060 From: "asterisk" ;tag=as5b3a0f26 To: ;tag=2FE5A038-2207 Date: Mon, 09 Apr 2012 19:55:53 GMT Call-ID: 020d85de7f95ac9073059f40754d5e1e@2.119.245.40:5060 Server: Cisco-SIPGateway/IOS-12.x CSeq: 103 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER Allow-Events: telephone-event Contact: Record-Route: Content-Disposition: session;handling=required Content-Type: application/sdp Content-Length: 246 v=0 o=CiscoSystemsSIP-GW-UserAgent 861 4253 IN IP4 83.211.2.220 s=SIP Call c=IN IP4 62.94.199.35 t=0 0 m=audio 52950 RTP/AVP 8 101 c=IN IP4 62.94.199.35 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 <-------------> --- (15 headers 11 lines) --- list_route: hop: Found RTP audio format 8 Found RTP audio format 101 Found audio description format PCMA for ID 8 Found audio description format telephone-event for ID 101 Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 62.94.199.35:52950 -- SIP/eutelia-00000002 is making progress passing it to DAHDI/11-1 <--- SIP read from UDP:192.168.3.14:2048 ---> SUBSCRIBE sip:154Casa@192.168.3.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.14:2048;branch=z9hG4bK-5qelt216ke4b;rport From: ;tag=asvybwur93 To: Call-ID: 3c268318cdb1-cerji572lrrh CSeq: 11 SUBSCRIBE Max-Forwards: 70 Contact: ;reg-id=1 Event: dialog Accept: application/dialog-info+xml User-Agent: snom300/8.4.34 Expires: 3600 Content-Length: 0 <-------------> --- (13 headers 0 lines) --- Creating new subscription Sending to 192.168.3.14:2048 (NAT) list_route: hop: Found peer '154' for '154' from 192.168.3.14:2048 <--- Transmitting (NAT) to 192.168.3.14:2048 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.3.14:2048;branch=z9hG4bK-5qelt216ke4b;received=192.168.3.14;rport=2048 From: ;tag=asvybwur93 To: ;tag=as70d12163 Call-ID: 3c268318cdb1-cerji572lrrh CSeq: 11 SUBSCRIBE Server: Asterisk PBX 1.8.11.0-2darkbasic1~squeeze Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk.linuxsystems.it", nonce="1b13690b" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '3c268318cdb1-cerji572lrrh' in 6400 ms (Method: SUBSCRIBE) <--- SIP read from UDP:192.168.3.14:2048 ---> SUBSCRIBE sip:154Casa@192.168.3.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.14:2048;branch=z9hG4bK-ho8mzj3a0qcz;rport From: ;tag=asvybwur93 To: Call-ID: 3c268318cdb1-cerji572lrrh CSeq: 12 SUBSCRIBE Max-Forwards: 70 Contact: ;reg-id=1 Event: dialog Accept: application/dialog-info+xml User-Agent: snom300/8.4.34 Authorization: Digest username="154",realm="asterisk.linuxsystems.it",nonce="1b13690b",uri="sip:154Casa@192.168.3.1",response="901537ecb56e16065471951da6db2251",algorithm=MD5 Expires: 3600 Content-Length: 0 <-------------> --- (14 headers 0 lines) --- Creating new subscription Sending to 192.168.3.14:2048 (NAT) Found peer '154' for '154' from 192.168.3.14:2048 Looking for 154Casa in phones-casa (domain 192.168.3.1) -- Added extension '154Casa' priority -1 to interni-casa Scheduling destruction of SIP dialog '3c268318cdb1-cerji572lrrh' in 3610000 ms (Method: SUBSCRIBE) <--- Transmitting (NAT) to 192.168.3.14:2048 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.14:2048;branch=z9hG4bK-ho8mzj3a0qcz;received=192.168.3.14;rport=2048 From: ;tag=asvybwur93 To: ;tag=as70d12163 Call-ID: 3c268318cdb1-cerji572lrrh CSeq: 12 SUBSCRIBE Server: Asterisk PBX 1.8.11.0-2darkbasic1~squeeze Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Expires: 3600 Contact: ;expires=3600 Content-Length: 0 <------------> set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.3.14:2048 Reliably Transmitting (NAT) to 192.168.3.14:2048: NOTIFY sip:154@192.168.3.14:2048;line=d0iogtg6 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.1:5060;branch=z9hG4bK565410a5;rport Max-Forwards: 70 From: ;tag=as70d12163 To: ;tag=asvybwur93 Contact: Call-ID: 3c268318cdb1-cerji572lrrh CSeq: 102 NOTIFY User-Agent: Asterisk PBX 1.8.11.0-2darkbasic1~squeeze Subscription-State: active Event: dialog Content-Type: application/dialog-info+xml Content-Length: 209 confirmed --- <--- SIP read from UDP:192.168.3.14:2048 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.3.1:5060;branch=z9hG4bK565410a5;rport=5060 From: ;tag=as70d12163 To: ;tag=asvybwur93 Call-ID: 3c268318cdb1-cerji572lrrh CSeq: 102 NOTIFY Content-Length: 0 <-------------> --- (7 headers 0 lines) --- Reliably Transmitting (NAT) to 192.168.3.12:2048: OPTIONS sip:155@192.168.3.12:2048;line=qr8mjt5c SIP/2.0 Via: SIP/2.0/UDP 192.168.3.1:5060;branch=z9hG4bK65f63d97;rport Max-Forwards: 70 From: "asterisk" ;tag=as7c7e36c3 To: Contact: Call-ID: 5ecccd8e440ef1000f8fabc51b94ead1@192.168.3.1:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.11.0-2darkbasic1~squeeze Date: Mon, 09 Apr 2012 19:56:01 GMT llow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- Retransmitting #1 (NAT) to 192.168.3.12:2048: OPTIONS sip:155@192.168.3.12:2048;line=qr8mjt5c SIP/2.0 Via: SIP/2.0/UDP 192.168.3.1:5060;branch=z9hG4bK65f63d97;rport Max-Forwards: 70 From: "asterisk" ;tag=as7c7e36c3 To: Contact: Call-ID: 5ecccd8e440ef1000f8fabc51b94ead1@192.168.3.1:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.11.0-2darkbasic1~squeeze Date: Mon, 09 Apr 2012 19:56:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- <--- SIP read from UDP:83.211.227.21:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 2.119.245.40:5060;received=2.119.245.40;branch=z9hG4bK5cc667dc;rport=5060 From: "asterisk" ;tag=as5b3a0f26 To: ;tag=2FE5A038-2207 Date: Mon, 09 Apr 2012 19:55:53 GMT Call-ID: 020d85de7f95ac9073059f40754d5e1e@2.119.245.40:5060 Server: Cisco-SIPGateway/IOS-12.x CSeq: 103 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER Supported: replaces Allow-Events: telephone-event Contact: Record-Route: Content-Type: application/sdp Content-Length: 246 v=0 o=CiscoSystemsSIP-GW-UserAgent 861 4253 IN IP4 83.211.2.220 s=SIP Call c=IN IP4 62.94.199.35 t=0 0 m=audio 52950 RTP/AVP 8 101 c=IN IP4 62.94.199.35 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 <-------------> --- (15 headers 11 lines) --- list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 83.211.227.21:5060 Transmitting (NAT) to 83.211.227.21:5060: ACK sip:494071914032@83.211.2.220:5060 SIP/2.0 Via: SIP/2.0/UDP 2.119.245.40:5060;branch=z9hG4bK3f9d76f8;rport Route: Max-Forwards: 70 From: "asterisk" ;tag=as5b3a0f26 To: ;tag=2FE5A038-2207 Contact: Call-ID: 020d85de7f95ac9073059f40754d5e1e@2.119.245.40:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 1.8.11.0-2darkbasic1~squeeze Content-Length: 0 --- -- SIP/eutelia-00000002 answered DAHDI/11-1 Retransmitting #2 (NAT) to 192.168.3.12:2048: OPTIONS sip:155@192.168.3.12:2048;line=qr8mjt5c SIP/2.0 Via: SIP/2.0/UDP 192.168.3.1:5060;branch=z9hG4bK65f63d97;rport Max-Forwards: 70 From: "asterisk" ;tag=as7c7e36c3 To: Contact: Call-ID: 5ecccd8e440ef1000f8fabc51b94ead1@192.168.3.1:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.11.0-2darkbasic1~squeeze Date: Mon, 09 Apr 2012 19:56:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- -- Channel 11 echo canceler disabled its NLP. -- Channel 11 detected a CED tone towards the network. Retransmitting #3 (NAT) to 192.168.3.12:2048: OPTIONS sip:155@192.168.3.12:2048;line=qr8mjt5c SIP/2.0 Via: SIP/2.0/UDP 192.168.3.1:5060;branch=z9hG4bK65f63d97;rport Max-Forwards: 70 From: "asterisk" ;tag=as7c7e36c3 To: Contact: Call-ID: 5ecccd8e440ef1000f8fabc51b94ead1@192.168.3.1:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.11.0-2darkbasic1~squeeze Date: Mon, 09 Apr 2012 19:56:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- <--- SIP read from UDP:192.168.3.10:5060 ---> <-------------> Retransmitting #4 (NAT) to 192.168.3.12:2048: OPTIONS sip:155@192.168.3.12:2048;line=qr8mjt5c SIP/2.0 Via: SIP/2.0/UDP 192.168.3.1:5060;branch=z9hG4bK65f63d97;rport Max-Forwards: 70 From: "asterisk" ;tag=as7c7e36c3 To: Contact: Call-ID: 5ecccd8e440ef1000f8fabc51b94ead1@192.168.3.1:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.11.0-2darkbasic1~squeeze Date: Mon, 09 Apr 2012 19:56:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- Really destroying SIP dialog '5ecccd8e440ef1000f8fabc51b94ead1@192.168.3.1:5060' Method: OPTIONS [Apr 9 21:56:05] NOTICE[1839]: channel.c:4152 __ast_read: Dropping incompatible voice frame on SIP/eutelia-00000002 of format slin since our native format has changed to 0x8 (alaw) <--- SIP read from UDP:83.211.227.21:5060 ---> INVITE sip:0719206651@2.119.245.40:5060 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 83.211.227.21;branch=z9hG4bK702b.b6b11312.0 Via: SIP/2.0/UDP 83.211.2.220:5060;rport=55968;received=83.211.2.220;branch=z9hG4bK31B00C10DF From: ;tag=2FE5A038-2207 To: "asterisk" ;tag=as5b3a0f26 Call-ID: 020d85de7f95ac9073059f40754d5e1e@2.119.245.40:5060 User-Agent: Cisco-SIPGateway/IOS-12.x CSeq: 101 INVITE Max-Forwards: 9 Contact: Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Length: 395 P-hint: rr-enforced v=0 o=CiscoSystemsSIP-GW-UserAgent 861 4254 IN IP4 83.211.2.220 s=SIP Call c=IN IP4 62.94.199.35 t=0 0 m=image 58394 udptl t38 c=IN IP4 62.94.199.35 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxFillBitRemoval:0 a=T38FaxTranscodingMMR:0 a=T38FaxTranscodingJBIG:0 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:200 a=T38FaxMaxDatagram:72 a=T38FaxUdpEC:t38UDPRedundancy <-------------> --- (16 headers 16 lines) --- Sending to 83.211.227.21:5060 (NAT) == Using UDPTL CoS mark 5 Got T.38 offer in SDP in dialog 020d85de7f95ac9073059f40754d5e1e@2.119.245.40:5060 Capabilities: us - 0x8 (alaw), peer - audio=0x0 (nothing)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x0 (nothing) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing) Got T.38 Re-invite without audio. Keeping RTP active during T.38 session. <--- Transmitting (NAT) to 83.211.227.21:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 83.211.227.21;branch=z9hG4bK702b.b6b11312.0;received=83.211.227.21;rport=5060 Via: SIP/2.0/UDP 83.211.2.220:5060;rport=55968;received=83.211.2.220;branch=z9hG4bK31B00C10DF Record-Route: From: ;tag=2FE5A038-2207 To: "asterisk" ;tag=as5b3a0f26 Call-ID: 020d85de7f95ac9073059f40754d5e1e@2.119.245.40:5060 CSeq: 101 INVITE Server: Asterisk PBX 1.8.11.0-2darkbasic1~squeeze Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> [Apr 9 21:56:07] ERROR[1839]: astobj2.c:110 INTERNAL_OBJ: user_data is NULL <--- Reliably Transmitting (NAT) to 83.211.227.21:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 83.211.227.21;branch=z9hG4bK702b.b6b11312.0;received=83.211.227.21;rport=5060 Via: SIP/2.0/UDP 83.211.2.220:5060;rport=55968;received=83.211.2.220;branch=z9hG4bK31B00C10DF Record-Route: From: ;tag=2FE5A038-2207 To: "asterisk" ;tag=as5b3a0f26 Call-ID: 020d85de7f95ac9073059f40754d5e1e@2.119.245.40:5060 CSeq: 101 INVITE Server: Asterisk PBX 1.8.11.0-2darkbasic1~squeeze Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 291 v=0 o=root 985362844 985362846 IN IP4 2.119.245.40 s=Asterisk PBX 1.8.11.0-2darkbasic1~squeeze c=IN IP4 2.119.245.40 t=0 0 m=image 4939 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxDatagram:1400 a=T38FaxUdpEC:t38UDPRedundancy <------------> [Apr 9 21:56:07] WARNING[1839]: channel.c:4913 ast_write: Codec mismatch on channel DAHDI/11-1 setting write format to slin from alaw native formats 0x8 (alaw) <--- SIP read from UDP:83.211.227.21:5060 ---> ACK sip:0719206651@2.119.245.40:5060 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 83.211.227.21;branch=z9hG4bK702b.b6b11312.2 Via: SIP/2.0/UDP 83.211.2.220:5060;rport=55968;received=83.211.2.220;branch=z9hG4bK31B00D1686 From: ;tag=2FE5A038-2207 To: "asterisk" ;tag=as5b3a0f26 Call-ID: 020d85de7f95ac9073059f40754d5e1e@2.119.245.40:5060 Max-Forwards: 9 CSeq: 101 ACK Content-Length: 0 P-hint: rr-enforced <-------------> --- (11 headers 0 lines) --- <--- SIP read from UDP:192.168.3.4:5060 ---> <-------------> Reliably Transmitting (NAT) to 192.168.3.12:2048: OPTIONS sip:155@192.168.3.12:2048;line=qr8mjt5c SIP/2.0 Via: SIP/2.0/UDP 192.168.3.1:5060;branch=z9hG4bK637432be;rport Max-Forwards: 70 From: "asterisk" ;tag=as49635a25 To: Contact: Call-ID: 15ba4f233ff249805f815b5952e1a3a1@192.168.3.1:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.11.0-2darkbasic1~squeeze Date: Mon, 09 Apr 2012 19:56:15 GMT llow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- Retransmitting #1 (NAT) to 192.168.3.12:2048: OPTIONS sip:155@192.168.3.12:2048;line=qr8mjt5c SIP/2.0 Via: SIP/2.0/UDP 192.168.3.1:5060;branch=z9hG4bK637432be;rport Max-Forwards: 70 From: "asterisk" ;tag=as49635a25 To: Contact: Call-ID: 15ba4f233ff249805f815b5952e1a3a1@192.168.3.1:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.11.0-2darkbasic1~squeeze Date: Mon, 09 Apr 2012 19:56:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- Retransmitting #2 (NAT) to 192.168.3.12:2048: OPTIONS sip:155@192.168.3.12:2048;line=qr8mjt5c SIP/2.0 Via: SIP/2.0/UDP 192.168.3.1:5060;branch=z9hG4bK637432be;rport Max-Forwards: 70 From: "asterisk" ;tag=as49635a25 To: Contact: Call-ID: 15ba4f233ff249805f815b5952e1a3a1@192.168.3.1:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.11.0-2darkbasic1~squeeze Date: Mon, 09 Apr 2012 19:56:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- Retransmitting #3 (NAT) to 192.168.3.12:2048: OPTIONS sip:155@192.168.3.12:2048;line=qr8mjt5c SIP/2.0 Via: SIP/2.0/UDP 192.168.3.1:5060;branch=z9hG4bK637432be;rport Max-Forwards: 70 From: "asterisk" ;tag=as49635a25 To: Contact: Call-ID: 15ba4f233ff249805f815b5952e1a3a1@192.168.3.1:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.11.0-2darkbasic1~squeeze Date: Mon, 09 Apr 2012 19:56:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- Retransmitting #4 (NAT) to 192.168.3.12:2048: OPTIONS sip:155@192.168.3.12:2048;line=qr8mjt5c SIP/2.0 Via: SIP/2.0/UDP 192.168.3.1:5060;branch=z9hG4bK637432be;rport Max-Forwards: 70 From: "asterisk" ;tag=as49635a25 To: Contact: Call-ID: 15ba4f233ff249805f815b5952e1a3a1@192.168.3.1:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.11.0-2darkbasic1~squeeze Date: Mon, 09 Apr 2012 19:56:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- Really destroying SIP dialog '15ba4f233ff249805f815b5952e1a3a1@192.168.3.1:5060' Method: OPTIONS [Apr 9 21:56:23] NOTICE[1782]: chan_sip.c:13077 sip_reregister: -- Re-registration for 5260447@sip.messagenet.it > doing dnsmgr_lookup for 'sip.messagenet.it' REGISTER 11 headers, 0 lines Reliably Transmitting (NAT) to 212.97.59.76:5061: REGISTER sip:sip.messagenet.it:5061 SIP/2.0 Via: SIP/2.0/UDP 2.119.245.40:5060;branch=z9hG4bK5762f9ce;rport Max-Forwards: 70 From: ;tag=as5fae34f5 To: Call-ID: 6f897dd35b4304763e6009ed4e97ee1b@127.0.1.1 CSeq: 114 REGISTER User-Agent: Asterisk PBX 1.8.11.0-2darkbasic1~squeeze Authorization: Digest username="5260447", realm="sip.messagenet.it", algorithm=MD5, uri="sip:sip.messagenet.it:5061", nonce="T4M+0U+DPaUBHTT9nPvpuo+ASj6QwVH7po3yT5zLdlVvw4W8eadTDD+BWiyB", response="3288db37aced286822d8a5a2c98505f8", qop=auth, cnonce="2f368814", nc=00000004 Expires: 120 Contact: Content-Length: 0 --- <--- SIP read from UDP:212.97.59.76:5061 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 2.119.245.40:5060;branch=z9hG4bK5762f9ce;rport=5060 From: ;tag=as5fae34f5 To: ;tag=ac2f3091de46227321b1af258e10b75e-c704 Call-ID: 6f897dd35b4304763e6009ed4e97ee1b@127.0.1.1 CSeq: 114 REGISTER WWW-Authenticate: Digest realm="sip.messagenet.it", nonce="T4NADE+DPuA6Q7cZ8tDACL4QHayEKhAWpo3yT5zLdlVvw4W8eadTDGPp4cOC", qop="auth" Server: sip.messagenet.it SIP Proxy Content-Length: 0 Warning: 392 212.97.59.76:5061 "Noisy feedback tells: pid=18954 req_src_ip=2.119.245.40 req_src_port=5060 in_uri=sip:sip.messagenet.it:5061 out_uri=sip:sip.messagenet.it:5061 via_cnt==1" <-------------> --- (10 headers 0 lines) --- Responding to challenge, registration to domain/host name sip.messagenet.it > doing dnsmgr_lookup for 'sip.messagenet.it' REGISTER 11 headers, 0 lines Reliably Transmitting (NAT) to 212.97.59.76:5061: REGISTER sip:sip.messagenet.it SIP/2.0 Via: SIP/2.0/UDP 2.119.245.40:5060;branch=z9hG4bK2ebdcff9;rport Max-Forwards: 70 From: ;tag=as2030b59c To: Call-ID: 6f897dd35b4304763e6009ed4e97ee1b@127.0.1.1 CSeq: 115 REGISTER User-Agent: Asterisk PBX 1.8.11.0-2darkbasic1~squeeze Authorization: Digest username="5260447", realm="sip.messagenet.it", algorithm=MD5, uri="sip:sip.messagenet.it:5061", nonce="T4NADE+DPuA6Q7cZ8tDACL4QHayEKhAWpo3yT5zLdlVvw4W8eadTDGPp4cOC", response="4010c0c9f47170e7a0c609c4a86189c6", qop=auth, cnonce="7f2b6446", nc=00000001 Expires: 120 Contact: Content-Length: 0 --- <--- SIP read from UDP:212.97.59.76:5061 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 2.119.245.40:5060;branch=z9hG4bK2ebdcff9;rport=5060 From: ;tag=as2030b59c To: ;tag=ac2f3091de46227321b1af258e10b75e-316c Call-ID: 6f897dd35b4304763e6009ed4e97ee1b@127.0.1.1 CSeq: 115 REGISTER Date: Mon, 09 Apr 2012 19:56:23 GMT Contact: ;q=0.5;expires=120 Server: sip.messagenet.it SIP Proxy Content-Length: 0 Warning: 392 212.97.59.76:5061 "Noisy feedback tells: pid=18961 req_src_ip=2.119.245.40 req_src_port=5060 in_uri=sip:sip.messagenet.it out_uri=sip:sip.messagenet.it via_cnt==1" <-------------> --- (11 headers 0 lines) --- Scheduling destruction of SIP dialog '6f897dd35b4304763e6009ed4e97ee1b@127.0.1.1' in 32000 ms (Method: REGISTER) [Apr 9 21:56:23] NOTICE[1782]: chan_sip.c:20756 handle_response_register: Outbound Registration: Expiry for sip.messagenet.it is 120 sec (Scheduling reregistration in 105 s) [Apr 9 21:56:23] NOTICE[1782]: chan_sip.c:13077 sip_reregister: -- Re-registration for 0719206703@voip.eutelia.it > doing dnsmgr_lookup for 'voip.eutelia.it' REGISTER 11 headers, 0 lines Reliably Transmitting (NAT) to 83.211.227.21:5060: REGISTER sip:voip.eutelia.it SIP/2.0 Via: SIP/2.0/UDP 2.119.245.40:5060;branch=z9hG4bK334654d8;rport Max-Forwards: 70 From: ;tag=as3088ec6a To: Call-ID: 633d7e9a301ca2c46fcad6932c87a762@127.0.1.1 CSeq: 120 REGISTER User-Agent: Asterisk PBX 1.8.11.0-2darkbasic1~squeeze Authorization: Digest username="0719206703", realm="voip.eutelia.it", algorithm=MD5, uri="sip:voip.eutelia.it", nonce="4f833eba726aaa2ed484069d36865cd056b32cc3", response="f2720da7f3e1957fd71d73f096fc0877", qop=auth, cnonce="61502905", nc=00000002 Expires: 120 Contact: Content-Length: 0 --- <--- SIP read from UDP:83.211.227.21:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 2.119.245.40:5060;branch=z9hG4bK334654d8;rport=5060 From: ;tag=as3088ec6a To: ;tag=c040a69dfc7733bdec8c921a7a9f2d3a.13cd Call-ID: 633d7e9a301ca2c46fcad6932c87a762@127.0.1.1 CSeq: 120 REGISTER WWW-Authenticate: Digest realm="voip.eutelia.it", nonce="4f833f2357e9a30188dfe78464a31db20108b903", qop="auth", stale=true Server: SPS EUT RM GW 01 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Responding to challenge, registration to domain/host name voip.eutelia.it > doing dnsmgr_lookup for 'voip.eutelia.it' REGISTER 11 headers, 0 lines Reliably Transmitting (NAT) to 83.211.227.21:5060: REGISTER sip:voip.eutelia.it SIP/2.0 Via: SIP/2.0/UDP 2.119.245.40:5060;branch=z9hG4bK6fcbc0de;rport Max-Forwards: 70 From: ;tag=as0ac58e16 To: Call-ID: 633d7e9a301ca2c46fcad6932c87a762@127.0.1.1 CSeq: 121 REGISTER User-Agent: Asterisk PBX 1.8.11.0-2darkbasic1~squeeze Authorization: Digest username="0719206703", realm="voip.eutelia.it", algorithm=MD5, uri="sip:voip.eutelia.it", nonce="4f833f2357e9a30188dfe78464a31db20108b903", response="1c1b3bd8a4a54a06f32b942249bf91df", qop=auth, cnonce="29c14815", nc=00000001 Expires: 120 Contact: Content-Length: 0 --- [Apr 9 21:56:23] NOTICE[1782]: chan_sip.c:13077 sip_reregister: -- Re-registration for 0719206651@voip.eutelia.it > doing dnsmgr_lookup for 'voip.eutelia.it' REGISTER 11 headers, 0 lines Reliably Transmitting (NAT) to 83.211.227.21:5060: REGISTER sip:voip.eutelia.it SIP/2.0 Via: SIP/2.0/UDP 2.119.245.40:5060;branch=z9hG4bK5ab7db15;rport Max-Forwards: 70 From: ;tag=as7b4869d8 To: Call-ID: 39c7ba005f490a6e34668e0c27887cab@127.0.1.1 CSeq: 120 REGISTER User-Agent: Asterisk PBX 1.8.11.0-2darkbasic1~squeeze Authorization: Digest username="0719206651", realm="voip.eutelia.it", algorithm=MD5, uri="sip:voip.eutelia.it", nonce="4f833eba726aaa2ed484069d36865cd056b32cc3", response="0d30910338a415939a43b0ad7870450f", qop=auth, cnonce="7b411b1e", nc=00000002 Expires: 120 Contact: Content-Length: 0 --- <--- SIP read from UDP:83.211.227.21:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 2.119.245.40:5060;branch=z9hG4bK6fcbc0de;rport=5060 From: ;tag=as0ac58e16 To: ;tag=c040a69dfc7733bdec8c921a7a9f2d3a.379b Call-ID: 633d7e9a301ca2c46fcad6932c87a762@127.0.1.1 CSeq: 121 REGISTER Date: Mon, 09 Apr 2012 19:56:23 GMT Contact: ;q=0.5;expires=120 Server: SPS EUT RM GW 01 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Scheduling destruction of SIP dialog '633d7e9a301ca2c46fcad6932c87a762@127.0.1.1' in 32000 ms (Method: REGISTER) [Apr 9 21:56:23] NOTICE[1782]: chan_sip.c:20756 handle_response_register: Outbound Registration: Expiry for voip.eutelia.it is 120 sec (Scheduling reregistration in 105 s) <--- SIP read from UDP:83.211.227.21:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 2.119.245.40:5060;branch=z9hG4bK5ab7db15;rport=5060 From: ;tag=as7b4869d8 To: ;tag=c040a69dfc7733bdec8c921a7a9f2d3a.61d3 Call-ID: 39c7ba005f490a6e34668e0c27887cab@127.0.1.1 CSeq: 120 REGISTER WWW-Authenticate: Digest realm="voip.eutelia.it", nonce="4f833f2357e9a30188dfe78464a31db20108b903", qop="auth", stale=true Server: SPS EUT RM GW 01 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Responding to challenge, registration to domain/host name voip.eutelia.it > doing dnsmgr_lookup for 'voip.eutelia.it' REGISTER 11 headers, 0 lines Reliably Transmitting (NAT) to 83.211.227.21:5060: REGISTER sip:voip.eutelia.it SIP/2.0 Via: SIP/2.0/UDP 2.119.245.40:5060;branch=z9hG4bK2690472c;rport Max-Forwards: 70 From: ;tag=as214767c8 To: Call-ID: 39c7ba005f490a6e34668e0c27887cab@127.0.1.1 CSeq: 121 REGISTER User-Agent: Asterisk PBX 1.8.11.0-2darkbasic1~squeeze Authorization: Digest username="0719206651", realm="voip.eutelia.it", algorithm=MD5, uri="sip:voip.eutelia.it", nonce="4f833f2357e9a30188dfe78464a31db20108b903", response="92205a5dcecdfa7f9e58ed0984809c90", qop=auth, cnonce="46726db9", nc=00000001 Expires: 120 Contact: Content-Length: 0 --- <--- SIP read from UDP:83.211.227.21:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 2.119.245.40:5060;branch=z9hG4bK2690472c;rport=5060 From: ;tag=as214767c8 To: ;tag=c040a69dfc7733bdec8c921a7a9f2d3a.4cb3 Call-ID: 39c7ba005f490a6e34668e0c27887cab@127.0.1.1 CSeq: 121 REGISTER Date: Mon, 09 Apr 2012 19:56:23 GMT Contact: ;q=0.5;expires=120 Server: SPS EUT RM GW 01 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Scheduling destruction of SIP dialog '39c7ba005f490a6e34668e0c27887cab@127.0.1.1' in 32000 ms (Method: REGISTER) [Apr 9 21:56:23] NOTICE[1782]: chan_sip.c:20756 handle_response_register: Outbound Registration: Expiry for voip.eutelia.it is 120 sec (Scheduling reregistration in 105 s) <--- SIP read from UDP:192.168.3.10:5060 ---> <-------------> Reliably Transmitting (NAT) to 192.168.3.12:2048: OPTIONS sip:155@192.168.3.12:2048;line=qr8mjt5c SIP/2.0 Via: SIP/2.0/UDP 192.168.3.1:5060;branch=z9hG4bK6ed9951a;rport Max-Forwards: 70 From: "asterisk" ;tag=as42b8667e To: Contact: Call-ID: 64fe47ef2af3d303516303cf313884b9@192.168.3.1:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.11.0-2darkbasic1~squeeze Date: Mon, 09 Apr 2012 19:56:29 GMT llow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- Retransmitting #1 (NAT) to 192.168.3.12:2048: OPTIONS sip:155@192.168.3.12:2048;line=qr8mjt5c SIP/2.0 Via: SIP/2.0/UDP 192.168.3.1:5060;branch=z9hG4bK6ed9951a;rport Max-Forwards: 70 From: "asterisk" ;tag=as42b8667e To: Contact: Call-ID: 64fe47ef2af3d303516303cf313884b9@192.168.3.1:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.11.0-2darkbasic1~squeeze Date: Mon, 09 Apr 2012 19:56:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- Retransmitting #2 (NAT) to 192.168.3.12:2048: OPTIONS sip:155@192.168.3.12:2048;line=qr8mjt5c SIP/2.0 Via: SIP/2.0/UDP 192.168.3.1:5060;branch=z9hG4bK6ed9951a;rport Max-Forwards: 70 From: "asterisk" ;tag=as42b8667e To: Contact: Call-ID: 64fe47ef2af3d303516303cf313884b9@192.168.3.1:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.11.0-2darkbasic1~squeeze Date: Mon, 09 Apr 2012 19:56:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- Retransmitting #3 (NAT) to 192.168.3.12:2048: OPTIONS sip:155@192.168.3.12:2048;line=qr8mjt5c SIP/2.0 Via: SIP/2.0/UDP 192.168.3.1:5060;branch=z9hG4bK6ed9951a;rport Max-Forwards: 70 From: "asterisk" ;tag=as42b8667e To: Contact: Call-ID: 64fe47ef2af3d303516303cf313884b9@192.168.3.1:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.11.0-2darkbasic1~squeeze Date: Mon, 09 Apr 2012 19:56:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- <--- SIP read from UDP:192.168.3.4:5060 ---> <-------------> Retransmitting #4 (NAT) to 192.168.3.12:2048: OPTIONS sip:155@192.168.3.12:2048;line=qr8mjt5c SIP/2.0 Via: SIP/2.0/UDP 192.168.3.1:5060;branch=z9hG4bK6ed9951a;rport Max-Forwards: 70 From: "asterisk" ;tag=as42b8667e To: Contact: Call-ID: 64fe47ef2af3d303516303cf313884b9@192.168.3.1:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.11.0-2darkbasic1~squeeze Date: Mon, 09 Apr 2012 19:56:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- Really destroying SIP dialog '64fe47ef2af3d303516303cf313884b9@192.168.3.1:5060' Method: OPTIONS <--- SIP read from UDP:192.168.3.18:1024 ---> SUBSCRIBE sip:NikoCasa@192.168.3.1:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.18:1024;branch=z9hG4bK-qac84nncae8w;rport From: ;tag=hgajldfr16 To: ;tag=as06b0e352 Call-ID: 3c267b2ea481-etdtv5vc0swf CSeq: 6 SUBSCRIBE Max-Forwards: 70 Contact: ;reg-id=1 Event: dialog Accept: application/dialog-info+xml User-Agent: snom370/8.4.34 Expires: 3600 Content-Length: 0 <-------------> --- (13 headers 0 lines) --- <--- Transmitting (NAT) to 192.168.3.18:1024 ---> SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP 192.168.3.18:1024;branch=z9hG4bK-qac84nncae8w;received=192.168.3.18;rport=1024 From: ;tag=hgajldfr16 To: ;tag=as06b0e352 Call-ID: 3c267b2ea481-etdtv5vc0swf CSeq: 6 SUBSCRIBE Server: Asterisk PBX 1.8.11.0-2darkbasic1~squeeze Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> Scheduling destruction of SIP dialog '3c267b2ea481-etdtv5vc0swf' in 32000 ms (Method: SUBSCRIBE) Reliably Transmitting (NAT) to 192.168.3.14:2048: OPTIONS sip:154@192.168.3.14:2048;line=d0iogtg6 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.1:5060;branch=z9hG4bK05a3830a;rport Max-Forwards: 70 From: "asterisk" ;tag=as1f5ddc6c To: Contact: Call-ID: 703a3ead4874e46d7568420354b51369@192.168.3.1:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.11.0-2darkbasic1~squeeze Date: Mon, 09 Apr 2012 19:56:37 GMT llow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- Reliably Transmitting (NAT) to 192.168.3.16:2048: OPTIONS sip:153@192.168.3.16:2048;line=dd5h6b64 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.1:5060;branch=z9hG4bK5dcfe0ff;rport Max-Forwards: 70 From: "asterisk" ;tag=as52edd09c To: Contact: Call-ID: 4e31690965f8ee8f0fa6b274044d91a6@192.168.3.1:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.11.0-2darkbasic1~squeeze Date: Mon, 09 Apr 2012 19:56:37 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- Reliably Transmitting (NAT) to 192.168.3.18:1024: OPTIONS sip:156@192.168.3.18:1024 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.1:5060;branch=z9hG4bK03c9ec21;rport Max-Forwards: 70 From: "asterisk" ;tag=as753bb0bd To: Contact: Call-ID: 5e6b89fd45e46d786e068a5a3ee18936@192.168.3.1:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.11.0-2darkbasic1~squeeze Date: Mon, 09 Apr 2012 19:56:37 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- <--- SIP read from UDP:192.168.3.14:2048 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.1:5060;branch=z9hG4bK05a3830a;rport=5060 From: "asterisk" ;tag=as1f5ddc6c To: Call-ID: 703a3ead4874e46d7568420354b51369@192.168.3.1:5060 CSeq: 102 OPTIONS Contact: ;reg-id=1 User-Agent: snom300/8.4.34 Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Content-Length: 0 <-------------> --- (14 headers 0 lines) --- Really destroying SIP dialog '703a3ead4874e46d7568420354b51369@192.168.3.1:5060' Method: OPTIONS <--- SIP read from UDP:192.168.3.16:2048 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.1:5060;branch=z9hG4bK5dcfe0ff;rport=5060 From: "asterisk" ;tag=as52edd09c To: Call-ID: 4e31690965f8ee8f0fa6b274044d91a6@192.168.3.1:5060 CSeq: 102 OPTIONS Contact: ;reg-id=1 User-Agent: snom300/8.4.34 Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Content-Length: 0 <-------------> --- (14 headers 0 lines) --- Really destroying SIP dialog '4e31690965f8ee8f0fa6b274044d91a6@192.168.3.1:5060' Method: OPTIONS <--- SIP read from UDP:192.168.3.18:1024 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.1:5060;branch=z9hG4bK03c9ec21;rport=5060 From: "asterisk" ;tag=as753bb0bd To: Call-ID: 5e6b89fd45e46d786e068a5a3ee18936@192.168.3.1:5060 CSeq: 102 OPTIONS Contact: ;reg-id=1 User-Agent: snom370/8.4.34 Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Content-Length: 0 <-------------> --- (14 headers 0 lines) --- Really destroying SIP dialog '5e6b89fd45e46d786e068a5a3ee18936@192.168.3.1:5060' Method: OPTIONS Reliably Transmitting (NAT) to 192.168.3.22:2053: OPTIONS sip:104@192.168.3.22:2053;line=tk81f7o0 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.1:5060;branch=z9hG4bK32020bfe;rport Max-Forwards: 70 From: "asterisk" ;tag=as6e909ee9 To: Contact: Call-ID: 16cec015100def8e6968e53f0b238ebe@192.168.3.1:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.11.0-2darkbasic1~squeeze Date: Mon, 09 Apr 2012 19:56:37 GMT llow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- <--- SIP read from UDP:192.168.3.22:2053 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.1:5060;branch=z9hG4bK32020bfe;rport=5060 From: "asterisk" ;tag=as6e909ee9 To: Call-ID: 16cec015100def8e6968e53f0b238ebe@192.168.3.1:5060 CSeq: 102 OPTIONS Contact: ;reg-id=1 User-Agent: snom300/8.4.34 Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Content-Length: 0 <-------------> --- (14 headers 0 lines) --- Really destroying SIP dialog '16cec015100def8e6968e53f0b238ebe@192.168.3.1:5060' Method: OPTIONS Reliably Transmitting (NAT) to 83.211.227.21:5060: OPTIONS sip:voip.eutelia.it SIP/2.0 Via: SIP/2.0/UDP 2.119.245.40:5060;branch=z9hG4bK7ebdcd23;rport Max-Forwards: 70 From: "asterisk" ;tag=as64fb1ecd To: Contact: Call-ID: 49727c3318441fb73956e6890daadd6d@2.119.245.40:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.11.0-2darkbasic1~squeeze Date: Mon, 09 Apr 2012 19:56:38 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- <--- SIP read from UDP:83.211.227.21:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 2.119.245.40:5060;branch=z9hG4bK7ebdcd23;rport=5060 From: "asterisk" ;tag=as64fb1ecd To: ;tag=c040a69dfc7733bdec8c921a7a9f2d3a.70ff Call-ID: 49727c3318441fb73956e6890daadd6d@2.119.245.40:5060 CSeq: 102 OPTIONS Accept: */* Accept-Encoding: Accept-Language: en Supported: Server: SPS EUT RM GW 01 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Really destroying SIP dialog '49727c3318441fb73956e6890daadd6d@2.119.245.40:5060' Method: OPTIONS Reliably Transmitting (NAT) to 212.97.59.76:5061: OPTIONS sip:sip.messagenet.it SIP/2.0 Via: SIP/2.0/UDP 2.119.245.40:5060;branch=z9hG4bK19069f24;rport Max-Forwards: 70 From: "asterisk" ;tag=as5f1f3c02 To: Contact: Call-ID: 49c2d603770428de350334da0cc2024e@2.119.245.40:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.11.0-2darkbasic1~squeeze Date: Mon, 09 Apr 2012 19:56:38 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- <--- SIP read from UDP:212.97.59.76:5061 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 2.119.245.40:5060;branch=z9hG4bK19069f24;rport=5060 From: "asterisk" ;tag=as5f1f3c02 To: ;tag=98df93dff07c6bc0cd4e22344f9aa5a7.182c Call-ID: 49c2d603770428de350334da0cc2024e@2.119.245.40:5060 CSeq: 102 OPTIONS Accept: */* Accept-Encoding: Accept-Language: en Supported: Server: sip.messagenet.it SIP Proxy Content-Length: 0 Warning: 392 212.97.59.76:5061 "Noisy feedback tells: pid=18954 req_src_ip=2.119.245.40 req_src_port=5060 in_uri=sip:sip.messagenet.it out_uri=sip:sip.messagenet.it via_cnt==1" <-------------> --- (13 headers 0 lines) --- Really destroying SIP dialog '49c2d603770428de350334da0cc2024e@2.119.245.40:5060' Method: OPTIONS <--- SIP read from UDP:192.168.3.10:5060 ---> REGISTER sip:192.168.3.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.10:5060;branch=z9hG4bK844b92bd5239aa6779f30b7980c12e2e;rport From: "152" ;tag=314340917 To: "152" Call-ID: 2486947262@192_168_3_10 CSeq: 51 REGISTER Contact: "152" Max-Forwards: 70 User-Agent: C450 IP010720000000 Expires: 180 Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Sending to 192.168.3.10:5060 (NAT) <--- Transmitting (NAT) to 192.168.3.10:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.3.10:5060;branch=z9hG4bK844b92bd5239aa6779f30b7980c12e2e;received=192.168.3.10;rport=5060 From: "152" ;tag=314340917 To: "152" ;tag=as07819bb4 Call-ID: 2486947262@192_168_3_10 CSeq: 51 REGISTER Server: Asterisk PBX 1.8.11.0-2darkbasic1~squeeze Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk.linuxsystems.it", nonce="272eafad" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '2486947262@192_168_3_10' in 32000 ms (Method: REGISTER) <--- SIP read from UDP:192.168.3.10:5060 ---> REGISTER sip:192.168.3.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.10:5060;branch=z9hG4bKf92d46de34bf00dfa8b15ae396592fd;rport From: "152" ;tag=314340917 To: "152" Call-ID: 2486947262@192_168_3_10 CSeq: 52 REGISTER Contact: "152" Authorization: Digest username="152", realm="asterisk.linuxsystems.it", algorithm=MD5, uri="sip:192.168.3.1", nonce="272eafad", response="c0e38000e3c7ae55874f84b5a09c35d6" Max-Forwards: 70 User-Agent: C450 IP010720000000 Expires: 180 Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO Content-Length: 0 <-------------> --- (13 headers 0 lines) --- Sending to 192.168.3.10:5060 (NAT) Reliably Transmitting (NAT) to 192.168.3.10:5060: OPTIONS sip:152@192.168.3.10:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.1:5060;branch=z9hG4bK4ed75cdc;rport Max-Forwards: 70 From: "asterisk" ;tag=as34089db2 To: Contact: Call-ID: 22c2b00d07670bb94d39cf4e735e3192@192.168.3.1:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.11.0-2darkbasic1~squeeze Date: Mon, 09 Apr 2012 19:56:38 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- <--- Transmitting (NAT) to 192.168.3.10:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.10:5060;branch=z9hG4bKf92d46de34bf00dfa8b15ae396592fd;received=192.168.3.10;rport=5060 From: "152" ;tag=314340917 To: "152" ;tag=as07819bb4 Call-ID: 2486947262@192_168_3_10 CSeq: 52 REGISTER Server: Asterisk PBX 1.8.11.0-2darkbasic1~squeeze Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Expires: 180 Contact: ;expires=180 Date: Mon, 09 Apr 2012 19:56:38 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog '2486947262@192_168_3_10' in 32000 ms (Method: REGISTER) <--- SIP read from UDP:192.168.3.10:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.1:5060;branch=z9hG4bK4ed75cdc;rport=5060 From: "asterisk" ;tag=as34089db2 To: ;tag=2778731840 Call-ID: 22c2b00d07670bb94d39cf4e735e3192@192.168.3.1:5060 CSeq: 102 OPTIONS Contact: "152" Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO Accept: application/sdp,application/dtmf-relay Accept-Encoding: identity Accept-Language: en Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Really destroying SIP dialog '22c2b00d07670bb94d39cf4e735e3192@192.168.3.1:5060' Method: OPTIONS Scheduling destruction of SIP dialog '020d85de7f95ac9073059f40754d5e1e@2.119.245.40:5060' in 6400 ms (Method: ACK) set_destination: Parsing for address/port to send to set_destination: set destination to 83.211.227.21:5060 Reliably Transmitting (NAT) to 83.211.227.21:5060: BYE sip:494071914032@83.211.2.220:5060 SIP/2.0 Via: SIP/2.0/UDP 2.119.245.40:5060;branch=z9hG4bK3cf2c593;rport Route: Max-Forwards: 70 From: "asterisk" ;tag=as5b3a0f26 To: ;tag=2FE5A038-2207 Call-ID: 020d85de7f95ac9073059f40754d5e1e@2.119.245.40:5060 CSeq: 104 BYE User-Agent: Asterisk PBX 1.8.11.0-2darkbasic1~squeeze Proxy-Authorization: Digest username="0719206651", realm="voip.eutelia.it", algorithm=MD5, uri="sip:494071914032@83.211.2.220:5060", nonce="4f833f0567b572b037a33dd74f8b73cfb7f8d5d2", response="3d68d625a30459ea8ae5e29c19c7ef13", qop=auth, cnonce="03c50a5e", nc=00000002 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- == Spawn extension (faxes, 071914032, 2) exited non-zero on 'DAHDI/11-1' -- Hanging up on 'DAHDI/11-1' -- Hungup 'DAHDI/11-1' <--- SIP read from UDP:83.211.227.21:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 2.119.245.40:5060;received=2.119.245.40;branch=z9hG4bK3cf2c593;rport=5060 From: "asterisk" ;tag=as5b3a0f26 To: ;tag=2FE5A038-2207 Date: Mon, 09 Apr 2012 19:56:42 GMT Call-ID: 020d85de7f95ac9073059f40754d5e1e@2.119.245.40:5060 Server: Cisco-SIPGateway/IOS-12.x Content-Length: 0 CSeq: 104 BYE <-------------> --- (9 headers 0 lines) --- SIP Response message for INCOMING dialog BYE arrived Really destroying SIP dialog '020d85de7f95ac9073059f40754d5e1e@2.119.245.40:5060' Method: ACK Reliably Transmitting (NAT) to 192.168.3.12:2048: OPTIONS sip:155@192.168.3.12:2048;line=qr8mjt5c SIP/2.0 Via: SIP/2.0/UDP 192.168.3.1:5060;branch=z9hG4bK52240e71;rport Max-Forwards: 70 From: "asterisk" ;tag=as07e7ddc9 To: Contact: Call-ID: 41fc66af0551645b22c67cdb7fc07cfb@192.168.3.1:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.11.0-2darkbasic1~squeeze Date: Mon, 09 Apr 2012 19:56:43 GMT llow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- Retransmitting #1 (NAT) to 192.168.3.12:2048: OPTIONS sip:155@192.168.3.12:2048;line=qr8mjt5c SIP/2.0 Via: SIP/2.0/UDP 192.168.3.1:5060;branch=z9hG4bK52240e71;rport Max-Forwards: 70 From: "asterisk" ;tag=as07e7ddc9 To: Contact: Call-ID: 41fc66af0551645b22c67cdb7fc07cfb@192.168.3.1:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.11.0-2darkbasic1~squeeze Date: Mon, 09 Apr 2012 19:56:43 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- Retransmitting #2 (NAT) to 192.168.3.12:2048: OPTIONS sip:155@192.168.3.12:2048;line=qr8mjt5c SIP/2.0 Via: SIP/2.0/UDP 192.168.3.1:5060;branch=z9hG4bK52240e71;rport Max-Forwards: 70 From: "asterisk" ;tag=as07e7ddc9 To: Contact: Call-ID: 41fc66af0551645b22c67cdb7fc07cfb@192.168.3.1:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.11.0-2darkbasic1~squeeze Date: Mon, 09 Apr 2012 19:56:43 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- asterisk*CLI>