asterisk*CLI> Verbosity is at least 100 Core debug is at least 100 asterisk*CLI>  <--- SIP read from UDP:192.168.3.4:5060 ---> <-------------> asterisk*CLI> Reliably Transmitting (no NAT) to 192.168.3.3:2048: OPTIONS sip:104@192.168.3.3:2048;line=k5l8dvj0 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.1:5060;branch=z9hG4bK39283743 Max-Forwards: 70 From: "asterisk" ;tag=as55b1bfaa To: Contact: Call-ID: 338d03f14b4e358218f2ac1e183ee0d8@192.168.3.1:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.3-1digium1~squeeze Date: Tue, 12 Apr 2011 12:55:20 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- Reliably Transmitting (no NAT) to 192.168.1.158:5060: OPTIONS sip:zJZlTpTaQyFc4k0voKjL@192.168.1.158 SIP/2.0 Via: SIP/2.0/UDP asterisk-public-ip:5060;branch=z9hG4bK38054c86 Max-Forwards: 70 From: "asterisk" ;tag=as38ed27a8 To: Contact: Call-ID: 233f4e183468e8e12d3758bb1e272c8f@asterisk-public-ip:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.3-1digium1~squeeze Date: Tue, 12 Apr 2011 12:55:20 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- asterisk*CLI> Reliably Transmitting (no 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=z9hG4bK41cf5804 Max-Forwards: 70 From: "asterisk" ;tag=as791a375a To: Contact: Call-ID: 0fd2dda07f4947a26c728e6a38c1f457@192.168.3.1:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.3-1digium1~squeeze Date: Tue, 12 Apr 2011 12:55:20 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- asterisk*CLI>  <--- SIP read from UDP:192.168.3.3:2048 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.1:5060;branch=z9hG4bK39283743 From: "asterisk" ;tag=as55b1bfaa To: Call-ID: 338d03f14b4e358218f2ac1e183ee0d8@192.168.3.1:5060 CSeq: 102 OPTIONS Contact: ;reg-id=1 User-Agent: snom300/8.4.31 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 '338d03f14b4e358218f2ac1e183ee0d8@192.168.3.1:5060' Method: OPTIONS asterisk*CLI>  <--- SIP read from UDP:192.168.1.158:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP asterisk-public-ip:5060;branch=z9hG4bK38054c86 To: ;tag=dpprdhs05thc7d3comr9 From: "asterisk" ;tag=as38ed27a8 Call-ID: 233f4e183468e8e12d3758bb1e272c8f@asterisk-public-ip:5060 CSeq: 102 OPTIONS Accept: application/sdp Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Really destroying SIP dialog '233f4e183468e8e12d3758bb1e272c8f@asterisk-public-ip:5060' Method: OPTIONS asterisk*CLI>  <--- 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=z9hG4bK41cf5804 From: "asterisk" ;tag=as791a375a To: ;tag=2008538244 Call-ID: 0fd2dda07f4947a26c728e6a38c1f457@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 '0fd2dda07f4947a26c728e6a38c1f457@192.168.3.1:5060' Method: OPTIONS asterisk*CLI> Reliably Transmitting (no NAT) to 83.211.227.21:5060: OPTIONS sip:voip.eutelia.it SIP/2.0 Via: SIP/2.0/UDP asterisk-public-ip:5060;branch=z9hG4bK67bcb972 Max-Forwards: 70 From: "asterisk" ;tag=as39a5a0bd To: Contact: Call-ID: 32c63d97568b93f72983e4f950b7a4dd@asterisk-public-ip:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.3-1digium1~squeeze Date: Tue, 12 Apr 2011 12:55:20 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- asterisk*CLI>  <--- SIP read from UDP:192.168.3.2:5060 ---> REGISTER sip:192.168.3.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.2:5060;rport;branch=z9hG4bK90e567dc77 From: "wind8" ;tag=207747e0 To: "wind8" Call-ID: 14bbd42a37661c0810ba6da23f73af42@192.168.3.2 Contact: CSeq: 15702 REGISTER Max-Forwards: 70 Expires: 60 Allow: INVITE,CANCEL,ACK,BYE,NOTIFY,REFER,OPTIONS,INFO,MESSAGE,UPDATE Authorization: Digest username="wind8",realm="asterisk",nonce="3c3f5a79",response="9c6583e63049eeecd00f6e2eb1f82ce6",uri="sip:192.168.3.1",algorithm=MD5 User-Agent: Mv-37x (904290) Content-Length: 0 <-------------> --- (13 headers 0 lines) --- asterisk*CLI> Sending to 192.168.3.2:5060 (no NAT) <--- Transmitting (no NAT) to 192.168.3.2:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.3.2:5060;branch=z9hG4bK90e567dc77;received=192.168.3.2;rport=5060 From: "wind8" ;tag=207747e0 To: "wind8" Call-ID: 14bbd42a37661c0810ba6da23f73af42@192.168.3.2 CSeq: 15702 REGISTER Server: Asterisk PBX 1.8.3-1digium1~squeeze Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> <--- Transmitting (no NAT) to 192.168.3.2:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.3.2:5060;branch=z9hG4bK90e567dc77;received=192.168.3.2;rport=5060 From: "wind8" ;tag=207747e0 To: "wind8" ;tag=as77125c11 Call-ID: 14bbd42a37661c0810ba6da23f73af42@192.168.3.2 CSeq: 15702 REGISTER Server: Asterisk PBX 1.8.3-1digium1~squeeze Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="49231b2b" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '14bbd42a37661c0810ba6da23f73af42@192.168.3.2' in 32000 ms (Method: REGISTER) asterisk*CLI>  <--- SIP read from UDP:83.211.227.21:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP asterisk-public-ip:5060;branch=z9hG4bK67bcb972 From: "asterisk" ;tag=as39a5a0bd To: ;tag=c040a69dfc7733bdec8c921a7a9f2d3a.6040 Call-ID: 32c63d97568b93f72983e4f950b7a4dd@asterisk-public-ip:5060 CSeq: 102 OPTIONS Accept: */* Accept-Encoding: Accept-Language: en Supported: Server: SPS EUT RM GW 04 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- asterisk*CLI> Really destroying SIP dialog '32c63d97568b93f72983e4f950b7a4dd@asterisk-public-ip:5060' Method: OPTIONS asterisk*CLI>  <--- SIP read from UDP:192.168.3.2:5060 ---> REGISTER sip:192.168.3.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.2:5060;rport;branch=z9hG4bK64c8a62b15 From: "wind8" ;tag=207747e0 To: "wind8" Call-ID: 14bbd42a37661c0810ba6da23f73af42@192.168.3.2 Contact: CSeq: 15703 REGISTER Max-Forwards: 70 Expires: 60 Allow: INVITE,CANCEL,ACK,BYE,NOTIFY,REFER,OPTIONS,INFO,MESSAGE,UPDATE Authorization: Digest username="wind8",realm="asterisk",nonce="49231b2b",response="c560d9f9c7964452093c9a256fc9c5e5",uri="sip:192.168.3.1",algorithm=MD5 User-Agent: Mv-37x (904290) Content-Length: 0 <-------------> --- (13 headers 0 lines) --- asterisk*CLI> Sending to 192.168.3.2:5060 (no NAT) <--- Transmitting (no NAT) to 192.168.3.2:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.3.2:5060;branch=z9hG4bK64c8a62b15;received=192.168.3.2;rport=5060 From: "wind8" ;tag=207747e0 To: "wind8" Call-ID: 14bbd42a37661c0810ba6da23f73af42@192.168.3.2 CSeq: 15703 REGISTER Server: Asterisk PBX 1.8.3-1digium1~squeeze Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> asterisk*CLI> Reliably Transmitting (no NAT) to 192.168.3.2:5060: OPTIONS sip:wind8@192.168.3.2:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.1:5060;branch=z9hG4bK3678d86e Max-Forwards: 70 From: "asterisk" ;tag=as6cfec669 To: Contact: Call-ID: 7212b4515c1268ea198c46102f8b06a8@192.168.3.1:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.3-1digium1~squeeze Date: Tue, 12 Apr 2011 12:55:20 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- <--- Transmitting (no NAT) to 192.168.3.2:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.2:5060;branch=z9hG4bK64c8a62b15;received=192.168.3.2;rport=5060 From: "wind8" ;tag=207747e0 To: "wind8" ;tag=as77125c11 Call-ID: 14bbd42a37661c0810ba6da23f73af42@192.168.3.2 CSeq: 15703 REGISTER Server: Asterisk PBX 1.8.3-1digium1~squeeze Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Expires: 60 Contact: ;expires=60 Date: Tue, 12 Apr 2011 12:55:20 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog '14bbd42a37661c0810ba6da23f73af42@192.168.3.2' in 32000 ms (Method: REGISTER) asterisk*CLI>  <--- SIP read from UDP:192.168.3.2:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.1:5060;branch=z9hG4bK3678d86e From: "asterisk" ;tag=as6cfec669 To: ;tag=05b8591f Call-ID: 7212b4515c1268ea198c46102f8b06a8@192.168.3.1:5060 CSeq: 102 OPTIONS Allow: INVITE,CANCEL,ACK,BYE,NOTIFY,REFER,OPTIONS,INFO,MESSAGE,UPDATE Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Really destroying SIP dialog '7212b4515c1268ea198c46102f8b06a8@192.168.3.1:5060' Method: OPTIONS asterisk*CLI>  <--- SIP read from UDP:192.168.1.151:5060 ---> <-------------> asterisk*CLI>  <--- SIP read from UDP:83.211.227.21:5060 ---> INVITE sip:my-eutelia-number@asterisk-public-ip:5060 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 83.211.227.21;branch=z9hG4bKfddf.de925497.0 Via: SIP/2.0/UDP 195.62.226.4:5060;rport=52650;received=195.62.226.4;x-route-tag="tgrp:Slot6";branch=z9hG4bK207DB19EC1 From: ;tag=DF91DECC-1407 To: Call-ID: FAE09B93-643A11E0-9AA5A52A-99A453FB@195.62.226.4 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: 417 P-hint: 2 Niente 2 v=0 o=CiscoSystemsSIP-GW-UserAgent 1979 1319 IN IP4 195.62.226.4 s=SIP Call c=IN IP4 83.211.227.14 t=0 0 m=audio 53784 RTP/AVP 18 8 0 4 3 125 101 c=IN IP4 83.211.227.14 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:4 G723/8000 a=fmtp:4 bitrate=5.3;annexa=no a=rtpmap:3 GSM/8000 a=rtpmap:125 X-CCD/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 <-------------> --- (16 headers 17 lines) --- asterisk*CLI>  == Using UDPTL CoS mark 5 Sending to 83.211.227.21:5060 (no NAT) Using INVITE request as basis request - FAE09B93-643A11E0-9AA5A52A-99A453FB@195.62.226.4 Found peer 'eutelia' for '195.62.226.4' from 83.211.227.21:5060 asterisk*CLI>  == Using SIP RTP CoS mark 5 asterisk*CLI> Found RTP audio format 18 Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 4 Found RTP audio format 3 Found RTP audio format 125 Found RTP audio format 101 Found audio description format G729 for ID 18 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G723 for ID 4 Found audio description format GSM for ID 3 Found audio description format X-CCD for ID 125 Found audio description format telephone-event for ID 101 Capabilities: us - 0x100 (g729), peer - audio=0x10f (g723|gsm|ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 83.211.227.14:53784 Looking for my-eutelia-number in eutelia-in (domain asterisk-public-ip:5060) asterisk*CLI> list_route: hop: <--- Transmitting (no NAT) to 83.211.227.21:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 83.211.227.21;branch=z9hG4bKfddf.de925497.0;received=83.211.227.21 Via: SIP/2.0/UDP 195.62.226.4:5060;rport=52650;received=195.62.226.4;x-route-tag="tgrp:Slot6";branch=z9hG4bK207DB19EC1 Record-Route: From: ;tag=DF91DECC-1407 To: Call-ID: FAE09B93-643A11E0-9AA5A52A-99A453FB@195.62.226.4 CSeq: 101 INVITE Server: Asterisk PBX 1.8.3-1digium1~squeeze Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> -- Executing [my-eutelia-number@eutelia-in:1] Answer("SIP/eutelia-00000027", "") in new stack Audio is at 5060 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 83.211.227.21:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 83.211.227.21;branch=z9hG4bKfddf.de925497.0;received=83.211.227.21 Via: SIP/2.0/UDP 195.62.226.4:5060;rport=52650;received=195.62.226.4;x-route-tag="tgrp:Slot6";branch=z9hG4bK207DB19EC1 Record-Route: From: ;tag=DF91DECC-1407 To: ;tag=as585d9aa8 Call-ID: FAE09B93-643A11E0-9AA5A52A-99A453FB@195.62.226.4 CSeq: 101 INVITE Server: Asterisk PBX 1.8.3-1digium1~squeeze Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 272 v=0 o=root 430365997 430365997 IN IP4 asterisk-public-ip s=Asterisk PBX 1.8.3-1digium1~squeeze c=IN IP4 asterisk-public-ip t=0 0 m=audio 16980 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <------------> asterisk*CLI> Retransmitting #1 (no NAT) to 83.211.227.21:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 83.211.227.21;branch=z9hG4bKfddf.de925497.0;received=83.211.227.21 Via: SIP/2.0/UDP 195.62.226.4:5060;rport=52650;received=195.62.226.4;x-route-tag="tgrp:Slot6";branch=z9hG4bK207DB19EC1 Record-Route: From: ;tag=DF91DECC-1407 To: ;tag=as585d9aa8 Call-ID: FAE09B93-643A11E0-9AA5A52A-99A453FB@195.62.226.4 CSeq: 101 INVITE Server: Asterisk PBX 1.8.3-1digium1~squeeze Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 272 v=0 o=root 430365997 430365997 IN IP4 asterisk-public-ip s=Asterisk PBX 1.8.3-1digium1~squeeze c=IN IP4 asterisk-public-ip t=0 0 m=audio 16980 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- asterisk*CLI>  <--- SIP read from UDP:83.211.227.21:5060 ---> ACK sip:my-eutelia-number@asterisk-public-ip:5060 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 83.211.227.21;branch=z9hG4bKfddf.de925497.2 Via: SIP/2.0/UDP 195.62.226.4:5060;rport=52650;received=195.62.226.4;x-route-tag="tgrp:Slot6";branch=z9hG4bK207DB1A17A6 From: ;tag=DF91DECC-1407 To: ;tag=as585d9aa8 Call-ID: FAE09B93-643A11E0-9AA5A52A-99A453FB@195.62.226.4 Max-Forwards: 9 CSeq: 101 ACK Content-Length: 0 P-hint: rr-enforced <-------------> --- (11 headers 0 lines) --- asterisk*CLI>  <--- SIP read from UDP:83.211.227.21:5060 ---> ACK sip:my-eutelia-number@asterisk-public-ip:5060 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 83.211.227.21;branch=z9hG4bKfddf.de925497.2 Via: SIP/2.0/UDP 195.62.226.4:5060;rport=52650;received=195.62.226.4;x-route-tag="tgrp:Slot6";branch=z9hG4bK207DB1A17A6 From: ;tag=DF91DECC-1407 To: ;tag=as585d9aa8 Call-ID: FAE09B93-643A11E0-9AA5A52A-99A453FB@195.62.226.4 Max-Forwards: 9 CSeq: 101 ACK Content-Length: 0 P-hint: rr-enforced <-------------> --- (11 headers 0 lines) --- asterisk*CLI> Reliably Transmitting (no NAT) to 192.168.1.151:5060: OPTIONS sip:152@192.168.1.151:5060 SIP/2.0 Via: SIP/2.0/UDP asterisk-public-ip:5060;branch=z9hG4bK6e3ba41e Max-Forwards: 70 From: "asterisk" ;tag=as11df1310 To: Contact: Call-ID: 3c2977e540a021fd1c1384284b53f5db@asterisk-public-ip:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.3-1digium1~squeeze Date: Tue, 12 Apr 2011 12:55:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- asterisk*CLI>  -- Executing [my-eutelia-number@eutelia-in:2] Set("SIP/eutelia-00000027", "DYNAMIC_FEATURES=automixmon") in new stack -- Executing [my-eutelia-number@eutelia-in:3] Dial("SIP/eutelia-00000027", "SIP/159,,tx") in new stack asterisk*CLI>  == Using UDPTL CoS mark 5 asterisk*CLI>  == Using SIP RTP CoS mark 5 asterisk*CLI> Audio is at 5060 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.1.158:5060: INVITE sip:zJZlTpTaQyFc4k0voKjL@192.168.1.158 SIP/2.0 Via: SIP/2.0/UDP asterisk-public-ip:5060;branch=z9hG4bK4c1cb2fe Max-Forwards: 70 From: "asterisk" ;tag=as3fa1707a To: Contact: Call-ID: 5149a50c711ac8d01f6ec52b7739f86c@asterisk-public-ip:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.3-1digium1~squeeze Date: Tue, 12 Apr 2011 12:55:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 274 v=0 o=root 1550387730 1550387730 IN IP4 asterisk-public-ip s=Asterisk PBX 1.8.3-1digium1~squeeze c=IN IP4 asterisk-public-ip t=0 0 m=audio 17374 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- -- Called 159 asterisk*CLI>  <--- SIP read from UDP:192.168.1.151:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP asterisk-public-ip:5060;branch=z9hG4bK6e3ba41e From: "asterisk" ;tag=as11df1310 To: ;tag=1102267903 Call-ID: 3c2977e540a021fd1c1384284b53f5db@asterisk-public-ip: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 asterisk*CLI> <-------------> --- (12 headers 0 lines) --- Really destroying SIP dialog '3c2977e540a021fd1c1384284b53f5db@asterisk-public-ip:5060' Method: OPTIONS asterisk*CLI>  <--- SIP read from UDP:192.168.1.158:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP asterisk-public-ip:5060;branch=z9hG4bK4c1cb2fe To: From: "asterisk" ;tag=as3fa1707a Call-ID: 5149a50c711ac8d01f6ec52b7739f86c@asterisk-public-ip:5060 CSeq: 102 INVITE Content-Length: 0 <-------------> --- (7 headers 0 lines) --- asterisk*CLI> [Apr 12 14:55:31] NOTICE[7976]: res_rtp_asterisk.c:2190 ast_rtp_read: Unknown RTP codec 120 received from '(null)' asterisk*CLI>  <--- SIP read from UDP:192.168.1.158:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP asterisk-public-ip:5060;branch=z9hG4bK4c1cb2fe Contact: From: "asterisk" ;tag=as3fa1707a To: ;tag=1mhjum5not5nttbpivbhneb2oe58vk6q Supported: 100rel Call-ID: 5149a50c711ac8d01f6ec52b7739f86c@asterisk-public-ip:5060 CSeq: 102 INVITE Allow: INVITE,ACK,CANCEL,OPTIONS,BYE Content-Length: 0 <-------------> --- (10 headers 0 lines) --- -- SIP/159-00000028 is ringing asterisk*CLI> [Apr 12 14:55:31] WARNING[7976]: dsp.c:1353 ast_dsp_process: Inband DTMF is not supported on codec g729. Use RFC2833 asterisk*CLI>  <--- SIP read from UDP:192.168.3.4:5060 ---> <-------------> asterisk*CLI>  <--- SIP read from UDP:192.168.1.151:5060 ---> <-------------> asterisk*CLI> Really destroying SIP dialog '14bbd42a37661c0810ba6da23f73af42@192.168.3.2' Method: REGISTER asterisk*CLI> Disconnected from Asterisk server