*CLI> <--- SIP read from UDP://192.168.1.189:5060 ---> INVITE sip:HIDDEN@192.168.1.212;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 192.168.1.189:5060;branch=z9hG4bK-d8754z-11a3f9248194b120-1---d8754z- Max-Forwards: 70 Contact: To: From: "HIDDEN_CALLER";tag=79610b27 Call-ID: Y2ZlMjA3N2QxNTQ0MTEzMzZlN2JlZWQ4ZmU4MTIyNTI. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp User-Agent: Zoiper rev.5324 Content-Length: 232 v=0 o=Zoiper_user 0 0 IN IP4 192.168.1.189 s=Zoiper_session c=IN IP4 192.168.1.189 t=0 0 m=audio 10000 RTP/AVP 8 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv <-------------> --- (12 headers 11 lines) --- Sending to 192.168.1.189 : 5060 (no NAT) Using INVITE request as basis request - Y2ZlMjA3N2QxNTQ0MTEzMzZlN2JlZWQ4ZmU4MTIyNTI. Found peer 'HIDDEN_CALLER' for 'HIDDEN_CALLER' from 192.168.1.189:5060 *CLI> <--- Reliably Transmitting (no NAT) to 192.168.1.189:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.189:5060;branch=z9hG4bK-d8754z-11a3f9248194b120-1---d8754z-;received=192.168.1.189 From: "HIDDEN_CALLER";tag=79610b27 To: ;tag=as72292f9c Call-ID: Y2ZlMjA3N2QxNTQ0MTEzMzZlN2JlZWQ4ZmU4MTIyNTI. CSeq: 1 INVITE Server: Asterisk 1.6.1.6 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="hidden_realm", nonce="2d372cbb" Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'Y2ZlMjA3N2QxNTQ0MTEzMzZlN2JlZWQ4ZmU4MTIyNTI.' in 6400 ms (Method: INVITE) *CLI> <--- SIP read from UDP://192.168.1.189:5060 ---> ACK sip:HIDDEN@192.168.1.212;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 192.168.1.189:5060;branch=z9hG4bK-d8754z-11a3f9248194b120-1---d8754z- Max-Forwards: 70 To: ;tag=as72292f9c From: "HIDDEN_CALLER";tag=79610b27 Call-ID: Y2ZlMjA3N2QxNTQ0MTEzMzZlN2JlZWQ4ZmU4MTIyNTI. CSeq: 1 ACK Content-Length: 0 <-------------> --- (8 headers 0 lines) --- *CLI> <--- SIP read from UDP://192.168.1.189:5060 ---> INVITE sip:HIDDEN@192.168.1.212;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 192.168.1.189:5060;branch=z9hG4bK-d8754z-ff726869b6348a5a-1---d8754z- Max-Forwards: 70 Contact: To: From: "HIDDEN_CALLER";tag=79610b27 Call-ID: Y2ZlMjA3N2QxNTQ0MTEzMzZlN2JlZWQ4ZmU4MTIyNTI. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp User-Agent: Zoiper rev.5324 Authorization: Digest username="HIDDEN_CALLER",realm="hidden_realm",nonce="2d372cbb",uri="sip:HIDDEN@192.168.1.212;transport=UDP",response="0ff9aad6d71ceeb8bb959c754bf42282",algorithm=MD5 Content-Length: 232 v=0 o=Zoiper_user 0 0 IN IP4 192.168.1.189 s=Zoiper_session c=IN IP4 192.168.1.189 t=0 0 m=audio 10000 RTP/AVP 8 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv <-------------> --- (13 headers 11 lines) --- Sending to 192.168.1.189 : 5060 (no NAT) Using INVITE request as basis request - Y2ZlMjA3N2QxNTQ0MTEzMzZlN2JlZWQ4ZmU4MTIyNTI. Found peer 'HIDDEN_CALLER' for 'HIDDEN_CALLER' from 192.168.1.189:5060 Found RTP audio format 8 Found RTP audio format 3 Found RTP audio format 101 Peer audio RTP is at port 192.168.1.189:10000 Found audio description format PCMA for ID 8 Found audio description format GSM for ID 3 Found audio description format telephone-event for ID 101 Capabilities: us - 0xa (gsm|alaw), peer - audio=0xa (gsm|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xa (gsm|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.1.189:10000 Looking for HIDDEN in autenticato-single (domain 192.168.1.212) list_route: hop: *CLI> <--- Transmitting (no NAT) to 192.168.1.189:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.189:5060;branch=z9hG4bK-d8754z-ff726869b6348a5a-1---d8754z-;received=192.168.1.189 From: "HIDDEN_CALLER";tag=79610b27 To: Call-ID: Y2ZlMjA3N2QxNTQ0MTEzMzZlN2JlZWQ4ZmU4MTIyNTI. CSeq: 2 INVITE Server: Asterisk 1.6.1.6 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Length: 0 <------------> >> (HIDDEN) # Started new outgoing call from HIDDEN_CALLER at 2010-03-22 17:16:44 Audio is at 192.168.1.212 port 14956 Adding codec 0x8 (alaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.1.55:5060: INVITE sip:HIDDEN@192.168.1.55 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.212:5060;branch=z9hG4bK416d8e66;rport Max-Forwards: 70 From: "HIDDEN_CALLER" ;tag=as7b3856e7 To: Contact: Call-ID: 2d4b90cf792395094d506a760a201773@192.168.1.212 CSeq: 102 INVITE User-Agent: Asterisk 1.6.1.6 Date: Mon, 22 Mar 2010 16:16:44 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 286 v=0 o=root 861334254 861334254 IN IP4 192.168.1.212 s=Asterisk PBX 1.6.1.6 c=IN IP4 192.168.1.212 t=0 0 m=audio 14956 RTP/AVP 8 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- *CLI> <--- SIP read from UDP://192.168.1.55:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.212:5060;branch=z9hG4bK416d8e66;rport=5060;received=192.168.1.212 From: "HIDDEN_CALLER" ;tag=as7b3856e7 To: Call-ID: 2d4b90cf792395094d506a760a201773@192.168.1.212 CSeq: 102 INVITE Server: Patton SN4960 1E30V UI 00A0BA03264F R5.4 2009-07-20 H323 RBS SIP M5T SIP Stack/4.0.28.28 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- *CLI> <--- SIP read from UDP://192.168.1.55:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.1.212:5060;branch=z9hG4bK416d8e66;rport=5060;received=192.168.1.212 From: "HIDDEN_CALLER" ;tag=as7b3856e7 To: ;tag=2280262559 Call-ID: 2d4b90cf792395094d506a760a201773@192.168.1.212 CSeq: 102 INVITE Contact: Server: Patton SN4960 1E30V UI 00A0BA03264F R5.4 2009-07-20 H323 RBS SIP M5T SIP Stack/4.0.28.28 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- *CLI> <--- Transmitting (no NAT) to 192.168.1.189:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.1.189:5060;branch=z9hG4bK-d8754z-ff726869b6348a5a-1---d8754z-;received=192.168.1.189 From: "HIDDEN_CALLER";tag=79610b27 To: ;tag=as0d8fc30c Call-ID: Y2ZlMjA3N2QxNTQ0MTEzMzZlN2JlZWQ4ZmU4MTIyNTI. CSeq: 2 INVITE Server: Asterisk 1.6.1.6 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Length: 0 <------------> *CLI> <--- SIP read from UDP://192.168.1.55:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.212:5060;branch=z9hG4bK416d8e66;rport=5060;received=192.168.1.212 From: "HIDDEN_CALLER" ;tag=as7b3856e7 To: ;tag=2280262559 Call-ID: 2d4b90cf792395094d506a760a201773@192.168.1.212 CSeq: 102 INVITE Contact: Server: Patton SN4960 1E30V UI 00A0BA03264F R5.4 2009-07-20 H323 RBS SIP M5T SIP Stack/4.0.28.28 Supported: replaces Content-Type: application/sdp Content-Length: 141 v=0 o=MxSIP 0 280 IN IP4 192.168.1.55 s=SIP Call c=IN IP4 192.168.1.55 t=0 0 m=audio 10474 RTP/AVP 8 a=rtpmap:8 PCMA/8000 a=sendrecv <-------------> --- (11 headers 8 lines) --- Found RTP audio format 8 Peer audio RTP is at port 192.168.1.55:10474 Found audio description format PCMA for ID 8 Capabilities: us - 0xa (gsm|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 192.168.1.55:10474 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.1.55, port 5060 Transmitting (no NAT) to 192.168.1.55:5060: ACK sip:HIDDEN@192.168.1.55:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.212:5060;branch=z9hG4bK36e20eb1;rport Max-Forwards: 70 From: "HIDDEN_CALLER" ;tag=as7b3856e7 To: ;tag=2280262559 Contact: Call-ID: 2d4b90cf792395094d506a760a201773@192.168.1.212 CSeq: 102 ACK User-Agent: Asterisk 1.6.1.6 Content-Length: 0 --- Audio is at 192.168.1.212 port 13640 Adding codec 0x8 (alaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 192.168.1.189:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.189:5060;branch=z9hG4bK-d8754z-ff726869b6348a5a-1---d8754z-;received=192.168.1.189 From: "HIDDEN_CALLER";tag=79610b27 To: ;tag=as0d8fc30c Call-ID: Y2ZlMjA3N2QxNTQ0MTEzMzZlN2JlZWQ4ZmU4MTIyNTI. CSeq: 2 INVITE Server: Asterisk 1.6.1.6 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 286 v=0 o=root 572729483 572729483 IN IP4 192.168.1.212 s=Asterisk PBX 1.6.1.6 c=IN IP4 192.168.1.212 t=0 0 m=audio 13640 RTP/AVP 8 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> *CLI> <--- SIP read from UDP://192.168.1.189:5060 ---> ACK sip:HIDDEN@192.168.1.212 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.189:5060;branch=z9hG4bK-d8754z-b75224c8035faec7-1---d8754z- Max-Forwards: 70 Contact: To: ;tag=as0d8fc30c From: "HIDDEN_CALLER";tag=79610b27 Call-ID: Y2ZlMjA3N2QxNTQ0MTEzMzZlN2JlZWQ4ZmU4MTIyNTI. CSeq: 2 ACK User-Agent: Zoiper rev.5324 Authorization: Digest username="HIDDEN_CALLER",realm="hidden_realm",nonce="2d372cbb",uri="sip:HIDDEN@192.168.1.212;transport=UDP",response="0ff9aad6d71ceeb8bb959c754bf42282",algorithm=MD5 Content-Length: 0 <-------------> --- (11 headers 0 lines) --- *CLI> <--- SIP read from UDP://192.168.1.189:5060 ---> BYE sip:HIDDEN@192.168.1.212 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.189:5060;branch=z9hG4bK-d8754z-6b250a3b2f231008-1---d8754z- Max-Forwards: 70 Contact: To: ;tag=as0d8fc30c From: "HIDDEN_CALLER";tag=79610b27 Call-ID: Y2ZlMjA3N2QxNTQ0MTEzMzZlN2JlZWQ4ZmU4MTIyNTI. CSeq: 3 BYE User-Agent: Zoiper rev.5324 Authorization: Digest username="HIDDEN_CALLER",realm="hidden_realm",nonce="2d372cbb",uri="sip:HIDDEN@192.168.1.212",response="e5e4342c233d2b28d36f7ec947322538",algorithm=MD5 Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Sending to 192.168.1.189 : 5060 (no NAT) *CLI> <--- Transmitting (no NAT) to 192.168.1.189:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.189:5060;branch=z9hG4bK-d8754z-6b250a3b2f231008-1---d8754z-;received=192.168.1.189 From: "HIDDEN_CALLER";tag=79610b27 To: ;tag=as0d8fc30c Call-ID: Y2ZlMjA3N2QxNTQ0MTEzMzZlN2JlZWQ4ZmU4MTIyNTI. CSeq: 3 BYE Server: Asterisk 1.6.1.6 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> >> (HIDDEN) # Call from HIDDEN_CALLER ended at 22/03/2010 17:17:01. Duration(sec): 16. Scheduling destruction of SIP dialog '2d4b90cf792395094d506a760a201773@192.168.1.212' in 6400 ms (Method: INVITE) set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.1.55, port 5060 Reliably Transmitting (no NAT) to 192.168.1.55:5060: BYE sip:HIDDEN@192.168.1.55:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.212:5060;branch=z9hG4bK50c450f0;rport Max-Forwards: 70 From: "HIDDEN_CALLER" ;tag=as7b3856e7 To: ;tag=2280262559 Call-ID: 2d4b90cf792395094d506a760a201773@192.168.1.212 CSeq: 103 BYE User-Agent: Asterisk 1.6.1.6 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- *CLI> <--- SIP read from UDP://192.168.1.55:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.212:5060;branch=z9hG4bK50c450f0;rport=5060;received=192.168.1.212 From: "HIDDEN_CALLER" ;tag=as7b3856e7 To: ;tag=2280262559 Call-ID: 2d4b90cf792395094d506a760a201773@192.168.1.212 CSeq: 103 BYE Server: Patton SN4960 1E30V UI 00A0BA03264F R5.4 2009-07-20 H323 RBS SIP M5T SIP Stack/4.0.28.28 Content-Length: 0