Asterisk certified/16.8-cert2, Copyright (C) 1999 - 2018, 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 certified/16.8-cert2 currently running on localhost (pid = 1375) localhost*CLI> localhost*CLI> localhost*CLI> sip set debug on SIP Debugging enabled <--- SIP read from UDP:192.168.0.124:5060 ---> INVITE sip:52345@192.168.0.115:5060 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 192.168.0.124:5060;branch=z9hG4bKb162.86c913a4.0 Via: SIP/2.0/UDP 192.168.0.127:54058;received=192.168.0.127;branch=z9hG4bK-524287-1---930e88ba39c4750c;rport=54058 Max-Forwards: 69 Contact: To: From: ;tag=3e7d720b Call-ID: 105535NTAwYjVmMjI5MjM4ZDU4MGE5MzczYmFlYmI2ZTc1NTI CSeq: 1 INVITE Allow: OPTIONS, SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO Content-Type: application/sdp Supported: replaces User-Agent: Bria release 6.3.0 stamp 105535 Content-Length: 212 v=0 o=- 13256847453026219 1 IN IP4 192.168.0.127 s=Bria release 6.3.0 stamp 105535 c=IN IP4 192.168.0.127 t=0 0 m=audio 49914 RTP/AVP 0 8 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv <-------------> --- (15 headers 9 lines) --- Sending to 192.168.0.124:5060 (no NAT) Sending to 192.168.0.124:5060 (no NAT) Using INVITE request as basis request - 105535NTAwYjVmMjI5MjM4ZDU4MGE5MzczYmFlYmI2ZTc1NTI Found peer 'opensips' for '10001' from 192.168.0.124:5060 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Found audio description format telephone-event for ID 101 Capabilities: us - (ulaw|alaw|gsm|h263), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 192.168.0.127:49914 Looking for 52345 in playmedia (domain 192.168.0.115) sip_route_dump: route/path hop: <--- Transmitting (no NAT) to 192.168.0.124:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.124:5060;branch=z9hG4bKb162.86c913a4.0;received=192.168.0.124 Via: SIP/2.0/UDP 192.168.0.127:54058;received=192.168.0.127;branch=z9hG4bK-524287-1---930e88ba39c4750c;rport=54058 Record-Route: From: ;tag=3e7d720b To: Call-ID: 105535NTAwYjVmMjI5MjM4ZDU4MGE5MzczYmFlYmI2ZTc1NTI CSeq: 1 INVITE Server: Asterisk PBX certified/16.8-cert2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Length: 0 <------------> Audio is at 14158 Adding codec ulaw to SDP Adding codec alaw to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 192.168.0.124:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.124:5060;branch=z9hG4bKb162.86c913a4.0;received=192.168.0.124 Via: SIP/2.0/UDP 192.168.0.127:54058;received=192.168.0.127;branch=z9hG4bK-524287-1---930e88ba39c4750c;rport=54058 Record-Route: From: ;tag=3e7d720b To: ;tag=as2c9cc2b1 Call-ID: 105535NTAwYjVmMjI5MjM4ZDU4MGE5MzczYmFlYmI2ZTc1NTI CSeq: 1 INVITE Server: Asterisk PBX certified/16.8-cert2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 279 v=0 o=root 1517673119 1517673119 IN IP4 192.168.0.115 s=Asterisk PBX certified/16.8-cert2 c=IN IP4 192.168.0.115 t=0 0 m=audio 14158 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <------------> Retransmitting #1 (no NAT) to 192.168.0.124:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.124:5060;branch=z9hG4bKb162.86c913a4.0;received=192.168.0.124 Via: SIP/2.0/UDP 192.168.0.127:54058;received=192.168.0.127;branch=z9hG4bK-524287-1---930e88ba39c4750c;rport=54058 Record-Route: From: ;tag=3e7d720b To: ;tag=as2c9cc2b1 Call-ID: 105535NTAwYjVmMjI5MjM4ZDU4MGE5MzczYmFlYmI2ZTc1NTI CSeq: 1 INVITE Server: Asterisk PBX certified/16.8-cert2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 279 v=0 o=root 1517673119 1517673119 IN IP4 192.168.0.115 s=Asterisk PBX certified/16.8-cert2 c=IN IP4 192.168.0.115 t=0 0 m=audio 14158 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- Retransmitting #2 (no NAT) to 192.168.0.124:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.124:5060;branch=z9hG4bKb162.86c913a4.0;received=192.168.0.124 Via: SIP/2.0/UDP 192.168.0.127:54058;received=192.168.0.127;branch=z9hG4bK-524287-1---930e88ba39c4750c;rport=54058 Record-Route: From: ;tag=3e7d720b To: ;tag=as2c9cc2b1 Call-ID: 105535NTAwYjVmMjI5MjM4ZDU4MGE5MzczYmFlYmI2ZTc1NTI CSeq: 1 INVITE Server: Asterisk PBX certified/16.8-cert2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 279 v=0 o=root 1517673119 1517673119 IN IP4 192.168.0.115 s=Asterisk PBX certified/16.8-cert2 c=IN IP4 192.168.0.115 t=0 0 m=audio 14158 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- Retransmitting #3 (no NAT) to 192.168.0.124:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.124:5060;branch=z9hG4bKb162.86c913a4.0;received=192.168.0.124 Via: SIP/2.0/UDP 192.168.0.127:54058;received=192.168.0.127;branch=z9hG4bK-524287-1---930e88ba39c4750c;rport=54058 Record-Route: From: ;tag=3e7d720b To: ;tag=as2c9cc2b1 Call-ID: 105535NTAwYjVmMjI5MjM4ZDU4MGE5MzczYmFlYmI2ZTc1NTI CSeq: 1 INVITE Server: Asterisk PBX certified/16.8-cert2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 279 v=0 o=root 1517673119 1517673119 IN IP4 192.168.0.115 s=Asterisk PBX certified/16.8-cert2 c=IN IP4 192.168.0.115 t=0 0 m=audio 14158 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- Scheduling destruction of SIP dialog '105535NTAwYjVmMjI5MjM4ZDU4MGE5MzczYmFlYmI2ZTc1NTI' in 32000 ms (Method: INVITE) Retransmitting #4 (no NAT) to 192.168.0.124:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.124:5060;branch=z9hG4bKb162.86c913a4.0;received=192.168.0.124 Via: SIP/2.0/UDP 192.168.0.127:54058;received=192.168.0.127;branch=z9hG4bK-524287-1---930e88ba39c4750c;rport=54058 Record-Route: From: ;tag=3e7d720b To: ;tag=as2c9cc2b1 Call-ID: 105535NTAwYjVmMjI5MjM4ZDU4MGE5MzczYmFlYmI2ZTc1NTI CSeq: 1 INVITE Server: Asterisk PBX certified/16.8-cert2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 279 v=0 o=root 1517673119 1517673119 IN IP4 192.168.0.115 s=Asterisk PBX certified/16.8-cert2 c=IN IP4 192.168.0.115 t=0 0 m=audio 14158 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- Retransmitting #5 (no NAT) to 192.168.0.124:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.124:5060;branch=z9hG4bKb162.86c913a4.0;received=192.168.0.124 Via: SIP/2.0/UDP 192.168.0.127:54058;received=192.168.0.127;branch=z9hG4bK-524287-1---930e88ba39c4750c;rport=54058 Record-Route: From: ;tag=3e7d720b To: ;tag=as2c9cc2b1 Call-ID: 105535NTAwYjVmMjI5MjM4ZDU4MGE5MzczYmFlYmI2ZTc1NTI CSeq: 1 INVITE Server: Asterisk PBX certified/16.8-cert2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 279 v=0 o=root 1517673119 1517673119 IN IP4 192.168.0.115 s=Asterisk PBX certified/16.8-cert2 c=IN IP4 192.168.0.115 t=0 0 m=audio 14158 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- Retransmitting #6 (no NAT) to 192.168.0.124:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.124:5060;branch=z9hG4bKb162.86c913a4.0;received=192.168.0.124 Via: SIP/2.0/UDP 192.168.0.127:54058;received=192.168.0.127;branch=z9hG4bK-524287-1---930e88ba39c4750c;rport=54058 Record-Route: From: ;tag=3e7d720b To: ;tag=as2c9cc2b1 Call-ID: 105535NTAwYjVmMjI5MjM4ZDU4MGE5MzczYmFlYmI2ZTc1NTI CSeq: 1 INVITE Server: Asterisk PBX certified/16.8-cert2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 279 v=0 o=root 1517673119 1517673119 IN IP4 192.168.0.115 s=Asterisk PBX certified/16.8-cert2 c=IN IP4 192.168.0.115 t=0 0 m=audio 14158 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- Retransmitting #7 (no NAT) to 192.168.0.124:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.124:5060;branch=z9hG4bKb162.86c913a4.0;received=192.168.0.124 Via: SIP/2.0/UDP 192.168.0.127:54058;received=192.168.0.127;branch=z9hG4bK-524287-1---930e88ba39c4750c;rport=54058 Record-Route: From: ;tag=3e7d720b To: ;tag=as2c9cc2b1 Call-ID: 105535NTAwYjVmMjI5MjM4ZDU4MGE5MzczYmFlYmI2ZTc1NTI CSeq: 1 INVITE Server: Asterisk PBX certified/16.8-cert2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 279 v=0 o=root 1517673119 1517673119 IN IP4 192.168.0.115 s=Asterisk PBX certified/16.8-cert2 c=IN IP4 192.168.0.115 t=0 0 m=audio 14158 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- Retransmitting #8 (no NAT) to 192.168.0.124:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.124:5060;branch=z9hG4bKb162.86c913a4.0;received=192.168.0.124 Via: SIP/2.0/UDP 192.168.0.127:54058;received=192.168.0.127;branch=z9hG4bK-524287-1---930e88ba39c4750c;rport=54058 Record-Route: From: ;tag=3e7d720b To: ;tag=as2c9cc2b1 Call-ID: 105535NTAwYjVmMjI5MjM4ZDU4MGE5MzczYmFlYmI2ZTc1NTI CSeq: 1 INVITE Server: Asterisk PBX certified/16.8-cert2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 279 v=0 o=root 1517673119 1517673119 IN IP4 192.168.0.115 s=Asterisk PBX certified/16.8-cert2 c=IN IP4 192.168.0.115 t=0 0 m=audio 14158 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- Retransmitting #9 (no NAT) to 192.168.0.124:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.124:5060;branch=z9hG4bKb162.86c913a4.0;received=192.168.0.124 Via: SIP/2.0/UDP 192.168.0.127:54058;received=192.168.0.127;branch=z9hG4bK-524287-1---930e88ba39c4750c;rport=54058 Record-Route: From: ;tag=3e7d720b To: ;tag=as2c9cc2b1 Call-ID: 105535NTAwYjVmMjI5MjM4ZDU4MGE5MzczYmFlYmI2ZTc1NTI CSeq: 1 INVITE Server: Asterisk PBX certified/16.8-cert2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 279 v=0 o=root 1517673119 1517673119 IN IP4 192.168.0.115 s=Asterisk PBX certified/16.8-cert2 c=IN IP4 192.168.0.115 t=0 0 m=audio 14158 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- Retransmitting #10 (no NAT) to 192.168.0.124:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.124:5060;branch=z9hG4bKb162.86c913a4.0;received=192.168.0.124 Via: SIP/2.0/UDP 192.168.0.127:54058;received=192.168.0.127;branch=z9hG4bK-524287-1---930e88ba39c4750c;rport=54058 Record-Route: From: ;tag=3e7d720b To: ;tag=as2c9cc2b1 Call-ID: 105535NTAwYjVmMjI5MjM4ZDU4MGE5MzczYmFlYmI2ZTc1NTI CSeq: 1 INVITE Server: Asterisk PBX certified/16.8-cert2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 279 v=0 o=root 1517673119 1517673119 IN IP4 192.168.0.115 s=Asterisk PBX certified/16.8-cert2 c=IN IP4 192.168.0.115 t=0 0 m=audio 14158 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Feb 3 23:08:03] WARNING[1457]: chan_sip.c:4127 retrans_pkt: Retransmission timeout reached on transmission 105535NTAwYjVmMjI5MjM4ZDU4MGE5MzczYmFlYmI2ZTc1NTI for seqno 1 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 32001ms with no response Really destroying SIP dialog '105535NTAwYjVmMjI5MjM4ZDU4MGE5MzczYmFlYmI2ZTc1NTI' Method: INVITE localhost*CLI> sip set debug off SIP Debugging Disabled localhost*CLI>