gateway*CLI> <--- SIP read from UDP:212.27.52.5:5060 ---> INVITE sip:33950714930@192.100.100.154:5060;transport=udp SIP/2.0 Call-ID: 18401-ST-0b9bf7c2-3c2bb68a4@freephonie.net Contact: Content-Type: application/sdp CSeq: 192142438 INVITE From: " 0033482535638" ;tag=18401-IH-0b9bf7c3-1f7efcd44 Max-Forwards: 26 Record-Route: To: Via: SIP/2.0/UDP 212.27.52.5:5060;branch=z9hG4bK-ELDG-0dcb67a0-0bd0ab63 Allow: UPDATE,REFER,INFO User-Agent: Cirpack/v4.42q (gw_sip) Content-Length: 183 v=0 o=cp10 146114395807 146114395807 IN IP4 172.25.1.35 s=SIP Call c=IN IP4 212.27.52.130 t=0 0 m=audio 31178 RTP/AVP 8 b=AS:75 a=rtpmap:8 PCMA/8000/1 a=ptime:30 a=sendrecv <-------------> --- (13 headers 10 lines) --- Sending to 212.27.52.5:5060 (NAT) Sending to 212.27.52.5:5060 (NAT) Using INVITE request as basis request - 18401-ST-0b9bf7c2-3c2bb68a4@freephonie.net Found peer 'sip-outside' for '0478434930' from 212.27.52.5:5060 == Using SIP RTP CoS mark 5 Found RTP audio format 8 Found audio description format PCMA for ID 8 Capabilities: us - (gsm|ulaw|alaw), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 212.27.52.130:31178 Looking for 33950714930 in pstn2ip (domain 192.100.100.154) list_route: hop: <--- Transmitting (NAT) to 212.27.52.5:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 212.27.52.5:5060;branch=z9hG4bK-ELDG-0dcb67a0-0bd0ab63;received=212.27.52.5;rport=5060 Record-Route: From: " 0033482535638" ;tag=18401-IH-0b9bf7c3-1f7efcd44 To: Call-ID: 18401-ST-0b9bf7c2-3c2bb68a4@freephonie.net CSeq: 192142438 INVITE Server: emerginov Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> -- Executing [33950714930@pstn2ip:1] Set("SIP/sip-outside-0000000c", "TIMEOUT(absolute)=3600") in new stack -- Channel will hangup at 2016-04-20 12:19:17.656 CEST. -- Executing [33950714930@pstn2ip:2] Set("SIP/sip-outside-0000000c", "CALLERID(all)=+330478434930<+330478434930>") in new stack -- Executing [33950714930@pstn2ip:3] Dial("SIP/sip-outside-0000000c", "sip/33950714930@emerginov.apitech.eu") in new stack == Using SIP RTP CoS mark 5 Audio is at 19868 Adding codec 100004 (alaw) to SDP Adding codec 100003 (ulaw) to SDP Adding codec 100002 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 78.194.160.103:5060: INVITE sip:33950714930@emerginov.apitech.eu SIP/2.0 Via: SIP/2.0/UDP 109.190.122.74:5060;branch=z9hG4bK4633697a;rport Max-Forwards: 70 From: "+330478434930" ;tag=as74b748a6 To: Contact: Call-ID: 25a9ffa8266a977e59e97bc278e9eb0b@109.190.122.74:5060 CSeq: 102 INVITE User-Agent: emerginov Date: Wed, 20 Apr 2016 09:19:17 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 284 v=0 o=root 775169970 775169970 IN IP4 109.190.122.74 s=Asterisk PBX 11.7.0 c=IN IP4 109.190.122.74 t=0 0 m=audio 19868 RTP/AVP 8 0 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- -- Called sip/33950714930@emerginov.apitech.eu <--- SIP read from UDP:192.168.0.2:5060 ---> SIP/2.0 100 Giving a try Via: SIP/2.0/UDP 109.190.122.74:5060;received=192.168.0.5;branch=z9hG4bK4633697a;rport=5060 From: "+330478434930" ;tag=as74b748a6 To: Call-ID: 25a9ffa8266a977e59e97bc278e9eb0b@109.190.122.74:5060 CSeq: 102 INVITE Content-Length: 0 <-------------> --- (7 headers 0 lines) --- <--- SIP read from UDP:192.168.0.2:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 109.190.122.74:5060;received=192.168.0.5;branch=z9hG4bK4633697a;rport=5060 Record-Route: From: "+330478434930" ;tag=as74b748a6 To: ;tag=as755771e0 Call-ID: 25a9ffa8266a977e59e97bc278e9eb0b@109.190.122.74:5060 CSeq: 102 INVITE Server: emerginov Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 280 v=0 o=root 1941930198 1941930198 IN IP4 192.168.0.3 s=Asterisk PBX 11.7.0 c=IN IP4 192.168.0.3 t=0 0 m=audio 15092 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <-------------> --- (15 headers 13 lines) --- Found RTP audio format 0 Found RTP audio format 3 Found RTP audio format 8 Found RTP audio format 101 Found audio description format PCMU for ID 0 Found audio description format GSM for ID 3 Found audio description format PCMA for ID 8 Found audio description format telephone-event for ID 101 Capabilities: us - (gsm|ulaw|alaw), peer - audio=(gsm|ulaw|alaw)/video=(nothing)/text=(nothing), combined - (gsm|ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 192.168.0.3:15092 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 78.194.160.103:5060 Transmitting (NAT) to 192.168.0.2:5060: ACK sip:VOCALAPP@192.168.0.3:6060 SIP/2.0 Via: SIP/2.0/UDP 109.190.122.74:5060;branch=z9hG4bK5a2f1a29;rport Route: Max-Forwards: 70 From: "+330478434930" ;tag=as74b748a6 To: ;tag=as755771e0 Contact: Call-ID: 25a9ffa8266a977e59e97bc278e9eb0b@109.190.122.74:5060 CSeq: 102 ACK User-Agent: emerginov Content-Length: 0 --- -- SIP/emerginov.apitech.eu-0000000d answered SIP/sip-outside-0000000c Audio is at 11626 Adding codec 100004 (alaw) to SDP <--- Reliably Transmitting (NAT) to 212.27.52.5:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 212.27.52.5:5060;branch=z9hG4bK-ELDG-0dcb67a0-0bd0ab63;received=212.27.52.5;rport=5060 Record-Route: From: " 0033482535638" ;tag=18401-IH-0b9bf7c3-1f7efcd44 To: ;tag=as50212dbd Call-ID: 18401-ST-0b9bf7c2-3c2bb68a4@freephonie.net CSeq: 192142438 INVITE Server: emerginov Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 181 v=0 o=root 658481360 658481360 IN IP4 109.190.122.74 s=Asterisk PBX 11.7.0 c=IN IP4 109.190.122.74 t=0 0 m=audio 11626 RTP/AVP 8 a=rtpmap:8 PCMA/8000 a=ptime:20 a=sendrecv <------------> -- Locally bridging SIP/sip-outside-0000000c and SIP/emerginov.apitech.eu-0000000d <--- SIP read from UDP:212.27.52.5:5060 ---> ACK sip:33950714930@192.100.100.154:5060 SIP/2.0 Call-ID: 18401-ST-0b9bf7c2-3c2bb68a4@freephonie.net Contact: CSeq: 192142438 ACK From: " 0033482535638" ;tag=18401-IH-0b9bf7c3-1f7efcd44 Max-Forwards: 26 To: ;tag=as50212dbd Via: SIP/2.0/UDP 212.27.52.5:5060;branch=z9hG4bK-WCNO-0dcb67a3-0c35fd0a User-Agent: Cirpack/v4.42q (gw_sip) Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from UDP:192.168.0.2:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 109.190.122.74:5060;received=192.168.0.5;branch=z9hG4bK4633697a;rport=5060 Record-Route: From: "+330478434930" ;tag=as74b748a6 To: ;tag=as755771e0 Call-ID: 25a9ffa8266a977e59e97bc278e9eb0b@109.190.122.74:5060 CSeq: 102 INVITE Server: emerginov Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 280 v=0 o=root 1941930198 1941930198 IN IP4 192.168.0.3 s=Asterisk PBX 11.7.0 c=IN IP4 192.168.0.3 t=0 0 m=audio 15092 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <-------------> --- (15 headers 13 lines) --- set_destination: Parsing for address/port to send to set_destination: set destination to 78.194.160.103:5060 Transmitting (NAT) to 192.168.0.2:5060: ACK sip:VOCALAPP@192.168.0.3:6060 SIP/2.0 Via: SIP/2.0/UDP 109.190.122.74:5060;branch=z9hG4bK716b30d7;rport Route: Max-Forwards: 70 From: "+330478434930" ;tag=as74b748a6 To: ;tag=as755771e0 Contact: Call-ID: 25a9ffa8266a977e59e97bc278e9eb0b@109.190.122.74:5060 CSeq: 102 ACK User-Agent: emerginov Content-Length: 0 --- <--- SIP read from UDP:192.168.0.2:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 109.190.122.74:5060;received=192.168.0.5;branch=z9hG4bK4633697a;rport=5060 Record-Route: From: "+330478434930" ;tag=as74b748a6 To: ;tag=as755771e0 Call-ID: 25a9ffa8266a977e59e97bc278e9eb0b@109.190.122.74:5060 CSeq: 102 INVITE Server: emerginov Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 280 v=0 o=root 1941930198 1941930198 IN IP4 192.168.0.3 s=Asterisk PBX 11.7.0 c=IN IP4 192.168.0.3 t=0 0 m=audio 15092 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <-------------> --- (15 headers 13 lines) --- set_destination: Parsing for address/port to send to set_destination: set destination to 78.194.160.103:5060 Transmitting (NAT) to 192.168.0.2:5060: ACK sip:VOCALAPP@192.168.0.3:6060 SIP/2.0 Via: SIP/2.0/UDP 109.190.122.74:5060;branch=z9hG4bK455cdc51;rport Route: Max-Forwards: 70 From: "+330478434930" ;tag=as74b748a6 To: ;tag=as755771e0 Contact: Call-ID: 25a9ffa8266a977e59e97bc278e9eb0b@109.190.122.74:5060 CSeq: 102 ACK User-Agent: emerginov Content-Length: 0 --- Reliably Transmitting (NAT) to 212.27.52.5:5060: OPTIONS sip:freephonie.net SIP/2.0 Via: SIP/2.0/UDP 109.190.122.74:5060;branch=z9hG4bK27ea76a6;rport Max-Forwards: 70 From: "asterisk" ;tag=as4c74718b To: Contact: Call-ID: 2d89cd257535b951539f39ef1de2fd48@109.190.122.74:5060 CSeq: 102 OPTIONS User-Agent: emerginov Date: Wed, 20 Apr 2016 09:19:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- <--- SIP read from UDP:212.27.52.5:5060 ---> SIP/2.0 501 Not Implemented Yet Call-ID: 2d89cd257535b951539f39ef1de2fd48@109.190.122.74:5060 CSeq: 102 OPTIONS From: "asterisk" ;tag=as4c74718b To: ;tag=00-31353-15502180-1125d5660 Via: SIP/2.0/UDP 192.100.100.154:5060;received=109.190.122.74;rport=5060;branch=z9hG4bK27ea76a6 Content-Length: 0 <-------------> --- (7 headers 0 lines) --- Really destroying SIP dialog '2d89cd257535b951539f39ef1de2fd48@109.190.122.74:5060' Method: OPTIONS <--- SIP read from UDP:192.168.0.2:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 109.190.122.74:5060;received=192.168.0.5;branch=z9hG4bK4633697a;rport=5060 Record-Route: From: "+330478434930" ;tag=as74b748a6 To: ;tag=as755771e0 Call-ID: 25a9ffa8266a977e59e97bc278e9eb0b@109.190.122.74:5060 CSeq: 102 INVITE Server: emerginov Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 280 v=0 o=root 1941930198 1941930198 IN IP4 192.168.0.3 s=Asterisk PBX 11.7.0 c=IN IP4 192.168.0.3 t=0 0 m=audio 15092 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <-------------> --- (15 headers 13 lines) --- set_destination: Parsing for address/port to send to set_destination: set destination to 78.194.160.103:5060 Transmitting (NAT) to 192.168.0.2:5060: ACK sip:VOCALAPP@192.168.0.3:6060 SIP/2.0 Via: SIP/2.0/UDP 109.190.122.74:5060;branch=z9hG4bK6a6728a1;rport Route: Max-Forwards: 70 From: "+330478434930" ;tag=as74b748a6 To: ;tag=as755771e0 Contact: Call-ID: 25a9ffa8266a977e59e97bc278e9eb0b@109.190.122.74:5060 CSeq: 102 ACK User-Agent: emerginov Content-Length: 0 --- <--- SIP read from UDP:192.168.0.2:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 109.190.122.74:5060;received=192.168.0.5;branch=z9hG4bK4633697a;rport=5060 Record-Route: From: "+330478434930" ;tag=as74b748a6 To: ;tag=as755771e0 Call-ID: 25a9ffa8266a977e59e97bc278e9eb0b@109.190.122.74:5060 CSeq: 102 INVITE Server: emerginov Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 280 v=0 o=root 1941930198 1941930198 IN IP4 192.168.0.3 s=Asterisk PBX 11.7.0 c=IN IP4 192.168.0.3 t=0 0 m=audio 15092 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <-------------> --- (15 headers 13 lines) --- set_destination: Parsing for address/port to send to set_destination: set destination to 78.194.160.103:5060 Transmitting (NAT) to 192.168.0.2:5060: ACK sip:VOCALAPP@192.168.0.3:6060 SIP/2.0 Via: SIP/2.0/UDP 109.190.122.74:5060;branch=z9hG4bK75922a80;rport Route: Max-Forwards: 70 From: "+330478434930" ;tag=as74b748a6 To: ;tag=as755771e0 Contact: Call-ID: 25a9ffa8266a977e59e97bc278e9eb0b@109.190.122.74:5060 CSeq: 102 ACK User-Agent: emerginov Content-Length: 0 --- <--- SIP read from UDP:192.168.0.2:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 109.190.122.74:5060;received=192.168.0.5;branch=z9hG4bK4633697a;rport=5060 Record-Route: From: "+330478434930" ;tag=as74b748a6 To: ;tag=as755771e0 Call-ID: 25a9ffa8266a977e59e97bc278e9eb0b@109.190.122.74:5060 CSeq: 102 INVITE Server: emerginov Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 280 v=0 o=root 1941930198 1941930198 IN IP4 192.168.0.3 s=Asterisk PBX 11.7.0 c=IN IP4 192.168.0.3 t=0 0 m=audio 15092 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <-------------> --- (15 headers 13 lines) --- set_destination: Parsing for address/port to send to set_destination: set destination to 78.194.160.103:5060 Transmitting (NAT) to 192.168.0.2:5060: ACK sip:VOCALAPP@192.168.0.3:6060 SIP/2.0 Via: SIP/2.0/UDP 109.190.122.74:5060;branch=z9hG4bK13b917bf;rport Route: Max-Forwards: 70 From: "+330478434930" ;tag=as74b748a6 To: ;tag=as755771e0 Contact: Call-ID: 25a9ffa8266a977e59e97bc278e9eb0b@109.190.122.74:5060 CSeq: 102 ACK User-Agent: emerginov Content-Length: 0 --- <--- SIP read from UDP:192.168.0.2:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 109.190.122.74:5060;received=192.168.0.5;branch=z9hG4bK4633697a;rport=5060 Record-Route: From: "+330478434930" ;tag=as74b748a6 To: ;tag=as755771e0 Call-ID: 25a9ffa8266a977e59e97bc278e9eb0b@109.190.122.74:5060 CSeq: 102 INVITE Server: emerginov Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 280 v=0 o=root 1941930198 1941930198 IN IP4 192.168.0.3 s=Asterisk PBX 11.7.0 c=IN IP4 192.168.0.3 t=0 0 m=audio 15092 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <-------------> --- (15 headers 13 lines) --- set_destination: Parsing for address/port to send to set_destination: set destination to 78.194.160.103:5060 Transmitting (NAT) to 192.168.0.2:5060: ACK sip:VOCALAPP@192.168.0.3:6060 SIP/2.0 Via: SIP/2.0/UDP 109.190.122.74:5060;branch=z9hG4bK1a5f8261;rport Route: Max-Forwards: 70 From: "+330478434930" ;tag=as74b748a6 To: ;tag=as755771e0 Contact: Call-ID: 25a9ffa8266a977e59e97bc278e9eb0b@109.190.122.74:5060 CSeq: 102 ACK User-Agent: emerginov Content-Length: 0 --- <--- SIP read from UDP:212.27.52.5:5060 ---> Cirpack KeepAlive Packet <-------------> <--- SIP read from UDP:212.27.52.5:5060 ---> Cirpack KeepAlive Packet <-------------> <--- SIP read from UDP:192.168.0.2:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 109.190.122.74:5060;received=192.168.0.5;branch=z9hG4bK4633697a;rport=5060 Record-Route: From: "+330478434930" ;tag=as74b748a6 To: ;tag=as755771e0 Call-ID: 25a9ffa8266a977e59e97bc278e9eb0b@109.190.122.74:5060 CSeq: 102 INVITE Server: emerginov Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 280 v=0 o=root 1941930198 1941930198 IN IP4 192.168.0.3 s=Asterisk PBX 11.7.0 c=IN IP4 192.168.0.3 t=0 0 m=audio 15092 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <-------------> --- (15 headers 13 lines) --- set_destination: Parsing for address/port to send to set_destination: set destination to 78.194.160.103:5060 Transmitting (NAT) to 192.168.0.2:5060: ACK sip:VOCALAPP@192.168.0.3:6060 SIP/2.0 Via: SIP/2.0/UDP 109.190.122.74:5060;branch=z9hG4bK1ac6f2ba;rport Route: Max-Forwards: 70 From: "+330478434930" ;tag=as74b748a6 To: ;tag=as755771e0 Contact: Call-ID: 25a9ffa8266a977e59e97bc278e9eb0b@109.190.122.74:5060 CSeq: 102 ACK User-Agent: emerginov Content-Length: 0 --- <--- SIP read from UDP:212.27.52.5:5060 ---> Cirpack KeepAlive Packet <-------------> <--- SIP read from UDP:212.27.52.5:5060 ---> Cirpack KeepAlive Packet <-------------> <--- SIP read from UDP:192.168.0.2:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 109.190.122.74:5060;received=192.168.0.5;branch=z9hG4bK4633697a;rport=5060 Record-Route: From: "+330478434930" ;tag=as74b748a6 To: ;tag=as755771e0 Call-ID: 25a9ffa8266a977e59e97bc278e9eb0b@109.190.122.74:5060 CSeq: 102 INVITE Server: emerginov Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 280 v=0 o=root 1941930198 1941930198 IN IP4 192.168.0.3 s=Asterisk PBX 11.7.0 c=IN IP4 192.168.0.3 t=0 0 m=audio 15092 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <-------------> --- (15 headers 13 lines) --- set_destination: Parsing for address/port to send to set_destination: set destination to 78.194.160.103:5060 Transmitting (NAT) to 192.168.0.2:5060: ACK sip:VOCALAPP@192.168.0.3:6060 SIP/2.0 Via: SIP/2.0/UDP 109.190.122.74:5060;branch=z9hG4bK21f7c6c8;rport Route: Max-Forwards: 70 From: "+330478434930" ;tag=as74b748a6 To: ;tag=as755771e0 Contact: Call-ID: 25a9ffa8266a977e59e97bc278e9eb0b@109.190.122.74:5060 CSeq: 102 ACK User-Agent: emerginov Content-Length: 0 --- <--- SIP read from UDP:192.168.0.2:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 109.190.122.74:5060;received=192.168.0.5;branch=z9hG4bK4633697a;rport=5060 Record-Route: From: "+330478434930" ;tag=as74b748a6 To: ;tag=as755771e0 Call-ID: 25a9ffa8266a977e59e97bc278e9eb0b@109.190.122.74:5060 CSeq: 102 INVITE Server: emerginov Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 280 v=0 o=root 1941930198 1941930198 IN IP4 192.168.0.3 s=Asterisk PBX 11.7.0 c=IN IP4 192.168.0.3 t=0 0 m=audio 15092 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <-------------> --- (15 headers 13 lines) --- set_destination: Parsing for address/port to send to set_destination: set destination to 78.194.160.103:5060 Transmitting (NAT) to 192.168.0.2:5060: ACK sip:VOCALAPP@192.168.0.3:6060 SIP/2.0 Via: SIP/2.0/UDP 109.190.122.74:5060;branch=z9hG4bK11adc9c0;rport Route: Max-Forwards: 70 From: "+330478434930" ;tag=as74b748a6 To: ;tag=as755771e0 Contact: Call-ID: 25a9ffa8266a977e59e97bc278e9eb0b@109.190.122.74:5060 CSeq: 102 ACK User-Agent: emerginov Content-Length: 0 --- <--- SIP read from UDP:192.168.0.2:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 109.190.122.74:5060;received=192.168.0.5;branch=z9hG4bK4633697a;rport=5060 Record-Route: From: "+330478434930" ;tag=as74b748a6 To: ;tag=as755771e0 Call-ID: 25a9ffa8266a977e59e97bc278e9eb0b@109.190.122.74:5060 CSeq: 102 INVITE Server: emerginov Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 280 v=0 o=root 1941930198 1941930198 IN IP4 192.168.0.3 s=Asterisk PBX 11.7.0 c=IN IP4 192.168.0.3 t=0 0 m=audio 15092 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <-------------> --- (15 headers 13 lines) --- set_destination: Parsing for address/port to send to set_destination: set destination to 78.194.160.103:5060 Transmitting (NAT) to 192.168.0.2:5060: ACK sip:VOCALAPP@192.168.0.3:6060 SIP/2.0 Via: SIP/2.0/UDP 109.190.122.74:5060;branch=z9hG4bK299ddf65;rport Route: Max-Forwards: 70 From: "+330478434930" ;tag=as74b748a6 To: ;tag=as755771e0 Contact: Call-ID: 25a9ffa8266a977e59e97bc278e9eb0b@109.190.122.74:5060 CSeq: 102 ACK User-Agent: emerginov Content-Length: 0 --- <--- SIP read from UDP:212.27.52.5:5060 ---> Cirpack KeepAlive Packet <-------------> <--- SIP read from UDP:212.27.52.5:5060 ---> Cirpack KeepAlive Packet <-------------> <--- SIP read from UDP:212.27.52.5:5060 ---> Cirpack KeepAlive Packet <-------------> <--- SIP read from UDP:212.27.52.5:5060 ---> Cirpack KeepAlive Packet <-------------> <--- SIP read from UDP:212.27.52.5:5060 ---> BYE sip:33950714930@192.100.100.154:5060 SIP/2.0 Call-ID: 18401-ST-0b9bf7c2-3c2bb68a4@freephonie.net CSeq: 192142439 BYE From: " 0033482535638" ;tag=18401-IH-0b9bf7c3-1f7efcd44 Max-Forwards: 26 Record-Route: To: ;tag=as50212dbd Via: SIP/2.0/UDP 212.27.52.5:5060;branch=z9hG4bK-CCNS-0dcb6b60-3a50fb09 Reason: q.850;cause=16 User-Agent: Cirpack/v4.42q (gw_sip) Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Sending to 212.27.52.5:5060 (NAT) Scheduling destruction of SIP dialog '18401-ST-0b9bf7c2-3c2bb68a4@freephonie.net' in 6400 ms (Method: BYE) <--- Transmitting (NAT) to 212.27.52.5:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 212.27.52.5:5060;branch=z9hG4bK-CCNS-0dcb6b60-3a50fb09;received=212.27.52.5;rport=5060 Record-Route: From: " 0033482535638" ;tag=18401-IH-0b9bf7c3-1f7efcd44 To: ;tag=as50212dbd Call-ID: 18401-ST-0b9bf7c2-3c2bb68a4@freephonie.net CSeq: 192142439 BYE Server: emerginov Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> Scheduling destruction of SIP dialog '25a9ffa8266a977e59e97bc278e9eb0b@109.190.122.74:5060' in 32000 ms (Method: INVITE) set_destination: Parsing for address/port to send to set_destination: set destination to 78.194.160.103:5060 Reliably Transmitting (NAT) to 192.168.0.2:5060: BYE sip:VOCALAPP@192.168.0.3:6060 SIP/2.0 Via: SIP/2.0/UDP 109.190.122.74:5060;branch=z9hG4bK50b8e845;rport Route: Max-Forwards: 70 From: "+330478434930" ;tag=as74b748a6 To: ;tag=as755771e0 Call-ID: 25a9ffa8266a977e59e97bc278e9eb0b@109.190.122.74:5060 CSeq: 103 BYE User-Agent: emerginov X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- == Spawn extension (pstn2ip, 33950714930, 3) exited non-zero on 'SIP/sip-outside-0000000c' Retransmitting #1 (NAT) to 192.168.0.2:5060: BYE sip:VOCALAPP@192.168.0.3:6060 SIP/2.0 Via: SIP/2.0/UDP 109.190.122.74:5060;branch=z9hG4bK50b8e845;rport Route: Max-Forwards: 70 From: "+330478434930" ;tag=as74b748a6 To: ;tag=as755771e0 Call-ID: 25a9ffa8266a977e59e97bc278e9eb0b@109.190.122.74:5060 CSeq: 103 BYE User-Agent: emerginov X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- Retransmitting #2 (NAT) to 192.168.0.2:5060: BYE sip:VOCALAPP@192.168.0.3:6060 SIP/2.0 Via: SIP/2.0/UDP 109.190.122.74:5060;branch=z9hG4bK50b8e845;rport Route: Max-Forwards: 70 From: "+330478434930" ;tag=as74b748a6 To: ;tag=as755771e0 Call-ID: 25a9ffa8266a977e59e97bc278e9eb0b@109.190.122.74:5060 CSeq: 103 BYE User-Agent: emerginov X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- Retransmitting #3 (NAT) to 192.168.0.2:5060: BYE sip:VOCALAPP@192.168.0.3:6060 SIP/2.0 Via: SIP/2.0/UDP 109.190.122.74:5060;branch=z9hG4bK50b8e845;rport Route: Max-Forwards: 70 From: "+330478434930" ;tag=as74b748a6 To: ;tag=as755771e0 Call-ID: 25a9ffa8266a977e59e97bc278e9eb0b@109.190.122.74:5060 CSeq: 103 BYE User-Agent: emerginov X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- Really destroying SIP dialog '18401-ST-0b9bf7c2-3c2bb68a4@freephonie.net' Method: BYE Retransmitting #4 (NAT) to 192.168.0.2:5060: BYE sip:VOCALAPP@192.168.0.3:6060 SIP/2.0 Via: SIP/2.0/UDP 109.190.122.74:5060;branch=z9hG4bK50b8e845;rport Route: Max-Forwards: 70 From: "+330478434930" ;tag=as74b748a6 To: ;tag=as755771e0 Call-ID: 25a9ffa8266a977e59e97bc278e9eb0b@109.190.122.74:5060 CSeq: 103 BYE User-Agent: emerginov X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- Reliably Transmitting (NAT) to 212.27.52.5:5060: OPTIONS sip:freephonie.net SIP/2.0 Via: SIP/2.0/UDP 109.190.122.74:5060;branch=z9hG4bK34350b11;rport Max-Forwards: 70 From: "asterisk" ;tag=as03055bb2 To: Contact: Call-ID: 408504c86c103eea156c85040767f1c2@109.190.122.74:5060 CSeq: 102 OPTIONS User-Agent: emerginov Date: Wed, 20 Apr 2016 09:20:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- <--- SIP read from UDP:212.27.52.5:5060 ---> SIP/2.0 501 Not Implemented Yet Call-ID: 408504c86c103eea156c85040767f1c2@109.190.122.74:5060 CSeq: 102 OPTIONS From: "asterisk" ;tag=as03055bb2 To: ;tag=00-31863-1550298c-355fc3ec4 Via: SIP/2.0/UDP 192.100.100.154:5060;received=109.190.122.74;rport=5060;branch=z9hG4bK34350b11 Content-Length: 0 <-------------> --- (7 headers 0 lines) --- Really destroying SIP dialog '408504c86c103eea156c85040767f1c2@109.190.122.74:5060' Method: OPTIONS Retransmitting #5 (NAT) to 192.168.0.2:5060: BYE sip:VOCALAPP@192.168.0.3:6060 SIP/2.0 Via: SIP/2.0/UDP 109.190.122.74:5060;branch=z9hG4bK50b8e845;rport Route: Max-Forwards: 70 From: "+330478434930" ;tag=as74b748a6 To: ;tag=as755771e0 Call-ID: 25a9ffa8266a977e59e97bc278e9eb0b@109.190.122.74:5060 CSeq: 103 BYE User-Agent: emerginov X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- Retransmitting #6 (NAT) to 192.168.0.2:5060: BYE sip:VOCALAPP@192.168.0.3:6060 SIP/2.0 Via: SIP/2.0/UDP 109.190.122.74:5060;branch=z9hG4bK50b8e845;rport Route: Max-Forwards: 70 From: "+330478434930" ;tag=as74b748a6 To: ;tag=as755771e0 Call-ID: 25a9ffa8266a977e59e97bc278e9eb0b@109.190.122.74:5060 CSeq: 103 BYE User-Agent: emerginov X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- Retransmitting #7 (NAT) to 192.168.0.2:5060: BYE sip:VOCALAPP@192.168.0.3:6060 SIP/2.0 Via: SIP/2.0/UDP 109.190.122.74:5060;branch=z9hG4bK50b8e845;rport Route: Max-Forwards: 70 From: "+330478434930" ;tag=as74b748a6 To: ;tag=as755771e0 Call-ID: 25a9ffa8266a977e59e97bc278e9eb0b@109.190.122.74:5060 CSeq: 103 BYE User-Agent: emerginov X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- Retransmitting #8 (NAT) to 192.168.0.2:5060: BYE sip:VOCALAPP@192.168.0.3:6060 SIP/2.0 Via: SIP/2.0/UDP 109.190.122.74:5060;branch=z9hG4bK50b8e845;rport Route: Max-Forwards: 70 From: "+330478434930" ;tag=as74b748a6 To: ;tag=as755771e0 Call-ID: 25a9ffa8266a977e59e97bc278e9eb0b@109.190.122.74:5060 CSeq: 103 BYE User-Agent: emerginov X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- <--- SIP read from UDP:212.27.52.5:5060 ---> Cirpack KeepAlive Packet <-------------> <--- SIP read from UDP:212.27.52.5:5060 ---> Cirpack KeepAlive Packet <-------------> Retransmitting #9 (NAT) to 192.168.0.2:5060: BYE sip:VOCALAPP@192.168.0.3:6060 SIP/2.0 Via: SIP/2.0/UDP 109.190.122.74:5060;branch=z9hG4bK50b8e845;rport Route: Max-Forwards: 70 From: "+330478434930" ;tag=as74b748a6 To: ;tag=as755771e0 Call-ID: 25a9ffa8266a977e59e97bc278e9eb0b@109.190.122.74:5060 CSeq: 103 BYE User-Agent: emerginov X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- <--- SIP read from UDP:212.27.52.5:5060 ---> Cirpack KeepAlive Packet <-------------> <--- SIP read from UDP:212.27.52.5:5060 ---> Cirpack KeepAlive Packet <-------------> <--- SIP read from UDP:192.168.0.2:5060 ---> SIP/2.0 408 Request Timeout Via: SIP/2.0/UDP 109.190.122.74:5060;received=192.168.0.5;branch=z9hG4bK50b8e845;rport=5060 From: "+330478434930" ;tag=as74b748a6 To: ;tag=as755771e0 Call-ID: 25a9ffa8266a977e59e97bc278e9eb0b@109.190.122.74:5060 CSeq: 103 BYE Content-Length: 0 <-------------> --- (7 headers 0 lines) --- Really destroying SIP dialog '25a9ffa8266a977e59e97bc278e9eb0b@109.190.122.74:5060' Method: INVITE