Asterisk 1.8.10.1~dfsg-1ubuntu1, 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 1.8.10.1~dfsg-1ubuntu1 currently running on asterisk (pid = 952) asterisk*CLI> <--- SIP read from UDP:192.168.1.203:5060 ---> OPTIONS sip:ping@publicIP.com SIP/2.0 Via: SIP/2.0/UDP 192.168.1.203:5060;branch=z9hG4bK-e2350c238022467ba723aed9c4df62b3;rport To: From: ;tag=7cf241a4b0a48cdf7e20b2b0 Call-ID: 681cb0bc72564f16a965ef8da5ab949e CSeq: 2 OPTIONS Max-Forwards: 0 Allow: INVITE,ACK,CANCEL,BYE,MESSAGE Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Looking for ping in unauthenticated (domain publicIP.com) <--- Transmitting (NAT) to 192.168.1.203:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.1.203:5060;branch=z9hG4bK-e2350c238022467ba723aed9c4df62b3;received=192.168.1.203;rport=5060 From: ;tag=7cf241a4b0a48cdf7e20b2b0 To: ;tag=as292c4920 Call-ID: 681cb0bc72564f16a965ef8da5ab949e CSeq: 2 OPTIONS Server: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Accept: application/sdp Content-Length: 0 <------------> Scheduling destruction of SIP dialog '681cb0bc72564f16a965ef8da5ab949e' in 32000 ms (Method: OPTIONS) asterisk*CLI> <--- SIP read from UDP:192.168.1.203:5060 ---> REGISTER sip:asterisk SIP/2.0 Via: SIP/2.0/UDP 192.168.1.203:5060;branch=z9hG4bK-9acbd71e59a248e39542473ade197368;rport To: From: ;tag=caca43bbb9bb4acf57c2cf87 Call-ID: 52a02caa1eed4ebf8e8585ccfe219bdd CSeq: 1 REGISTER Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,MESSAGE Route: Contact: ;expires=300 Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Sending to 192.168.1.203:5060 (NAT) <--- Transmitting (NAT) to 192.168.1.203:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.203:5060;branch=z9hG4bK-9acbd71e59a248e39542473ade197368;received=192.168.1.203;rport=5060 From: ;tag=caca43bbb9bb4acf57c2cf87 To: ;tag=as2521980f Call-ID: 52a02caa1eed4ebf8e8585ccfe219bdd CSeq: 1 REGISTER Server: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1f8f4b69" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '52a02caa1eed4ebf8e8585ccfe219bdd' in 32000 ms (Method: REGISTER) asterisk*CLI> <--- SIP read from UDP:192.168.1.203:5060 ---> REGISTER sip:asterisk SIP/2.0 Via: SIP/2.0/UDP 192.168.1.203:5060;branch=z9hG4bK-995d682e0c714286a1a9de742fbdf547;rport To: From: ;tag=caca43bbb9bb4acf57c2cf87 Call-ID: 52a02caa1eed4ebf8e8585ccfe219bdd CSeq: 3 REGISTER Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,MESSAGE Route: Contact: ;expires=300 Authorization: digest realm="asterisk",username="alice",nonce="1f8f4b69",uri="sip:asterisk",cnonce="2f63c9758257420aaeb4d2e57248a428",response="922d4c1af63d0e123f7637d55787a268" Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Sending to 192.168.1.203:5060 (NAT) <--- Transmitting (NAT) to 192.168.1.203:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.203:5060;branch=z9hG4bK-995d682e0c714286a1a9de742fbdf547;received=192.168.1.203;rport=5060 From: ;tag=caca43bbb9bb4acf57c2cf87 To: ;tag=as2521980f Call-ID: 52a02caa1eed4ebf8e8585ccfe219bdd CSeq: 3 REGISTER Server: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Expires: 300 Contact: ;expires=300 Date: Fri, 30 Aug 2013 10:21:36 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog '52a02caa1eed4ebf8e8585ccfe219bdd' in 32000 ms (Method: REGISTER) asterisk*CLI> <--- SIP read from UDP:192.168.1.203:5060 ---> INVITE sip:echo@192.168.1.179 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.203:5060;branch=z9hG4bK-c72ff3ec98ce41eda289e84a99f3ed37;rport To: From: "Alice Cooper" ;tag=214942ec99aa32713dde15a5 Call-ID: 25fb81a7e1044521aecf597b05505bb9 CSeq: 4 INVITE Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,MESSAGE Content-Type: application/sdp Contact: Content-Length: 180 v=0 o=- 56602975 1 IN IP4 192.168.1.203 s=SIP Call t=0 0 m=audio 12000 RTP/AVP 0 8 c=IN IP4 192.168.1.203 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=ptime:20 a=sendrecv <-------------> --- (11 headers 10 lines) --- Sending to 192.168.1.203:5060 (NAT) Using INVITE request as basis request - 25fb81a7e1044521aecf597b05505bb9 Found peer 'alice' for 'alice' from 192.168.1.203:5060 <--- Reliably Transmitting (NAT) to 192.168.1.203:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.203:5060;branch=z9hG4bK-c72ff3ec98ce41eda289e84a99f3ed37;received=192.168.1.203;rport=5060 From: "Alice Cooper" ;tag=214942ec99aa32713dde15a5 To: ;tag=as12313b8e Call-ID: 25fb81a7e1044521aecf597b05505bb9 CSeq: 4 INVITE Server: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4fa3af3c" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '25fb81a7e1044521aecf597b05505bb9' in 32000 ms (Method: INVITE) asterisk*CLI> <--- SIP read from UDP:192.168.1.203:5060 ---> ACK sip:echo@192.168.1.179 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.203:5060;branch=z9hG4bK-c72ff3ec98ce41eda289e84a99f3ed37;rport Call-ID: 25fb81a7e1044521aecf597b05505bb9 From: "Alice Cooper" ;tag=214942ec99aa32713dde15a5 To: ;tag=as12313b8e CSeq: 4 ACK Max-Forwards: 70 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- asterisk*CLI> <--- SIP read from UDP:192.168.1.203:5060 ---> INVITE sip:echo@192.168.1.179 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.203:5060;branch=z9hG4bK-4ec27fe959db4898b42e31d1f76fe2c7;rport To: From: "Alice Cooper" ;tag=214942ec99aa32713dde15a5 Call-ID: 25fb81a7e1044521aecf597b05505bb9 CSeq: 5 INVITE Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,MESSAGE Content-Type: application/sdp Authorization: digest realm="asterisk",username="alice",nonce="4fa3af3c",uri="sip:echo@192.168.1.179",cnonce="e2792e9c4d9943e6a76a26ed632dbe59",response="2971fa3ca76b5affa59f4acfbf58fafc" Contact: Content-Length: 180 v=0 o=- 56602975 1 IN IP4 192.168.1.203 s=SIP Call t=0 0 m=audio 12000 RTP/AVP 0 8 c=IN IP4 192.168.1.203 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=ptime:20 a=sendrecv <-------------> --- (12 headers 10 lines) --- Sending to 192.168.1.203:5060 (NAT) Using INVITE request as basis request - 25fb81a7e1044521aecf597b05505bb9 Found peer 'alice' for 'alice' from 192.168.1.203:5060 Found RTP audio format 0 Found RTP audio format 8 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 192.168.1.203:12000 Looking for echo in SwissTiming (domain 192.168.1.179) list_route: hop: <--- Transmitting (NAT) to 192.168.1.203:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.203:5060;branch=z9hG4bK-4ec27fe959db4898b42e31d1f76fe2c7;received=192.168.1.203;rport=5060 From: "Alice Cooper" ;tag=214942ec99aa32713dde15a5 To: Call-ID: 25fb81a7e1044521aecf597b05505bb9 CSeq: 5 INVITE Server: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> Audio is at 17216 Adding codec 0x8 (alaw) to SDP Adding codec 0x4 (ulaw) to SDP <--- Reliably Transmitting (NAT) to 192.168.1.203:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.203:5060;branch=z9hG4bK-4ec27fe959db4898b42e31d1f76fe2c7;received=192.168.1.203;rport=5060 From: "Alice Cooper" ;tag=214942ec99aa32713dde15a5 To: ;tag=as3175074b Call-ID: 25fb81a7e1044521aecf597b05505bb9 CSeq: 5 INVITE Server: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 219 v=0 o=root 903423697 903423697 IN IP4 192.168.1.179 s=Asterisk PBX 1.8.10.1~dfsg-1ubuntu1 c=IN IP4 192.168.1.179 t=0 0 m=audio 17216 RTP/AVP 8 0 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=ptime:20 a=sendrecv <------------> asterisk*CLI> <--- SIP read from UDP:192.168.1.203:5060 ---> ACK sip:echo@192.168.1.179:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.203:5060;branch=z9hG4bK-2761f61ec95643fb820abc84da4e3dd5;rport To: ;tag=as3175074b From: ;tag=214942ec99aa32713dde15a5 Call-ID: 25fb81a7e1044521aecf597b05505bb9 CSeq: 5 ACK Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,MESSAGE Contact: Authorization: digest realm="asterisk",username="alice",nonce="4fa3af3c",uri="sip:echo@192.168.1.179",cnonce="e2792e9c4d9943e6a76a26ed632dbe59",response="2971fa3ca76b5affa59f4acfbf58fafc" Content-Length: 0 <-------------> --- (11 headers 0 lines) --- asterisk*CLI> <--- SIP read from UDP:192.168.1.203:5060 ---> INVITE sip:echo@192.168.1.179:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.203:5060;branch=z9hG4bK-992cdaf966cf4a6cbe20ebd4c5169c53;rport To: ;tag=as3175074b From: ;tag=214942ec99aa32713dde15a5 Call-ID: 25fb81a7e1044521aecf597b05505bb9 CSeq: 6 INVITE Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,MESSAGE Contact: Content-Type: application/sdp Content-Length: 180 v=0 o=- 56602975 2 IN IP4 192.168.1.203 s=SIP Call t=0 0 m=audio 12000 RTP/AVP 0 8 c=IN IP4 192.168.1.203 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=ptime:20 a=recvonly <-------------> --- (11 headers 10 lines) --- Sending to 192.168.1.203:5060 (NAT) Found RTP audio format 0 Found RTP audio format 8 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 192.168.1.203:12000 <--- Transmitting (NAT) to 192.168.1.203:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.203:5060;branch=z9hG4bK-992cdaf966cf4a6cbe20ebd4c5169c53;received=192.168.1.203;rport=5060 From: ;tag=214942ec99aa32713dde15a5 To: ;tag=as3175074b Call-ID: 25fb81a7e1044521aecf597b05505bb9 CSeq: 6 INVITE Server: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> Audio is at 17216 Adding codec 0x8 (alaw) to SDP Adding codec 0x4 (ulaw) to SDP <--- Reliably Transmitting (NAT) to 192.168.1.203:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.203:5060;branch=z9hG4bK-992cdaf966cf4a6cbe20ebd4c5169c53;received=192.168.1.203;rport=5060 From: ;tag=214942ec99aa32713dde15a5 To: ;tag=as3175074b Call-ID: 25fb81a7e1044521aecf597b05505bb9 CSeq: 6 INVITE Server: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 219 v=0 o=root 903423697 903423698 IN IP4 192.168.1.179 s=Asterisk PBX 1.8.10.1~dfsg-1ubuntu1 c=IN IP4 192.168.1.179 t=0 0 m=audio 17216 RTP/AVP 8 0 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=ptime:20 a=sendrecv <------------> asterisk*CLI> <--- SIP read from UDP:192.168.1.203:5060 ---> ACK sip:echo@192.168.1.179:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.203:5060;branch=z9hG4bK-0aab046c09944157995150839cd78d96;rport To: ;tag=as3175074b From: ;tag=214942ec99aa32713dde15a5 Call-ID: 25fb81a7e1044521aecf597b05505bb9 CSeq: 6 ACK Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,MESSAGE Contact: Content-Length: 0 <-------------> --- (10 headers 0 lines) --- asterisk*CLI> <--- SIP read from UDP:192.168.1.203:5060 ---> BYE sip:echo@192.168.1.179:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.203:5060;branch=z9hG4bK-36360060f5ab452aba49f94830032d47;rport To: ;tag=as3175074b From: ;tag=214942ec99aa32713dde15a5 Call-ID: 25fb81a7e1044521aecf597b05505bb9 CSeq: 7 BYE Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,MESSAGE Contact: Reason: SIP;text="Invalid SDP answer, sdp stream no: 0 stream-mode must be 'recvonly' (RFC 3264 6.)." Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Sending to 192.168.1.203:5060 (NAT) Scheduling destruction of SIP dialog '25fb81a7e1044521aecf597b05505bb9' in 32000 ms (Method: BYE) <--- Transmitting (NAT) to 192.168.1.203:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.203:5060;branch=z9hG4bK-36360060f5ab452aba49f94830032d47;received=192.168.1.203;rport=5060 From: ;tag=214942ec99aa32713dde15a5 To: ;tag=as3175074b Call-ID: 25fb81a7e1044521aecf597b05505bb9 CSeq: 7 BYE Server: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> asterisk*CLI> <--- SIP read from UDP:192.168.1.203:5060 ---> <-------------> asterisk*CLI> <--- SIP read from UDP:192.168.1.203:5060 ---> <-------------> asterisk*CLI> Really destroying SIP dialog '681cb0bc72564f16a965ef8da5ab949e' Method: OPTIONS asterisk*CLI> Really destroying SIP dialog '52a02caa1eed4ebf8e8585ccfe219bdd' Method: REGISTER asterisk*CLI> exit