dev*CLI> <--- SIP read from UDP:192.168.0.2:5060 ---> INVITE sip:VOCALAPP@emerginov.apitech.eu SIP/2.0 Record-Route: Via: SIP/2.0/UDP 78.194.160.103;branch=z9hG4bK247d.5a951ba3.0 Via: SIP/2.0/UDP 109.190.122.74:5060;received=192.168.0.5;branch=z9hG4bK4633697a;rport=5060 Max-Forwards: 69 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 P-Lang: en P-CallerID: +330478434930 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 <-------------> --- (18 headers 13 lines) --- Sending to 192.168.0.2:5060 (NAT) Sending to 192.168.0.2:5060 (NAT) Using INVITE request as basis request - 25a9ffa8266a977e59e97bc278e9eb0b@109.190.122.74:5060 No matching peer for '+330478434930' from '192.168.0.2:5060' == Using SIP RTP CoS mark 5 Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 3 Found RTP audio format 101 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format GSM for ID 3 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 109.190.122.74:19868 Looking for VOCALAPP in default (domain emerginov.apitech.eu) list_route: hop: <--- Transmitting (NAT) to 192.168.0.2:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 78.194.160.103;branch=z9hG4bK247d.5a951ba3.0;received=192.168.0.2;rport=5060 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: 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-Length: 0 <------------> -- Executing [VOCALAPP@default:1] Set("SIP/109.190.122.74-00000015", "file=incoming.php") in new stack -- Executing [VOCALAPP@default:2] AGI("SIP/109.190.122.74-00000015", "agi://dev.emerginov.localnet/vocalApp/incoming.php") in new stack Audio is at 15092 Adding codec 100003 (ulaw) to SDP Adding codec 100002 (gsm) to SDP Adding codec 100004 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (NAT) to 192.168.0.2:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 78.194.160.103;branch=z9hG4bK247d.5a951ba3.0;received=192.168.0.2;rport=5060 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 <------------> Retransmitting #1 (NAT) to 192.168.0.2:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 78.194.160.103;branch=z9hG4bK247d.5a951ba3.0;received=192.168.0.2;rport=5060 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 --- -- Playing '/var/lib/asterisk/emerginov-projects/media-out/vocalApp/echo_test' (escape_digits=#) (sample_offset 0) Retransmitting #2 (NAT) to 192.168.0.2:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 78.194.160.103;branch=z9hG4bK247d.5a951ba3.0;received=192.168.0.2;rport=5060 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 --- Retransmitting #3 (NAT) to 192.168.0.2:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 78.194.160.103;branch=z9hG4bK247d.5a951ba3.0;received=192.168.0.2;rport=5060 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 --- Retransmitting #4 (NAT) to 192.168.0.2:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 78.194.160.103;branch=z9hG4bK247d.5a951ba3.0;received=192.168.0.2;rport=5060 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 --- Retransmitting #5 (NAT) to 192.168.0.2:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 78.194.160.103;branch=z9hG4bK247d.5a951ba3.0;received=192.168.0.2;rport=5060 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 --- -- AGI Script agi://dev.emerginov.localnet/vocalApp/incoming.php completed, returning 4 == Spawn extension (default, VOCALAPP, 2) exited non-zero on 'SIP/109.190.122.74-00000015' Scheduling destruction of SIP dialog '25a9ffa8266a977e59e97bc278e9eb0b@109.190.122.74:5060' in 32000 ms (Method: INVITE) Retransmitting #6 (NAT) to 192.168.0.2:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 78.194.160.103;branch=z9hG4bK247d.5a951ba3.0;received=192.168.0.2;rport=5060 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 --- Retransmitting #7 (NAT) to 192.168.0.2:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 78.194.160.103;branch=z9hG4bK247d.5a951ba3.0;received=192.168.0.2;rport=5060 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 --- Retransmitting #8 (NAT) to 192.168.0.2:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 78.194.160.103;branch=z9hG4bK247d.5a951ba3.0;received=192.168.0.2;rport=5060 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 --- Retransmitting #9 (NAT) to 192.168.0.2:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 78.194.160.103;branch=z9hG4bK247d.5a951ba3.0;received=192.168.0.2;rport=5060 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 --- Retransmitting #10 (NAT) to 192.168.0.2:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 78.194.160.103;branch=z9hG4bK247d.5a951ba3.0;received=192.168.0.2;rport=5060 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 --- [Apr 20 11:19:24] WARNING[1881]: chan_sip.c:4175 retrans_pkt: Retransmission timeout reached on transmission 25a9ffa8266a977e59e97bc278e9eb0b@109.190.122.74:5060 for seqno 102 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 32000ms with no response Really destroying SIP dialog '25a9ffa8266a977e59e97bc278e9eb0b@109.190.122.74:5060' Method: INVITE