asterisk02*CLI> sip debug SIP Debugging Enabled [Mar 19 13:39:19] Destroying call '17725-D1B9-12F6-C4E0-8F529350FC34@192.168.1.229' [Mar 19 13:39:21] <-- SIP read from 192.168.1.254:49554: INVITE sip:105@asterisk02 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.254:49554;rport;branch=z9hG4bKB252CA9A2D6D46DEAABE588661026F22 From: 101 ;tag=1562601555 To: Contact: Call-ID: B610EE9B-C484-4BA9-A2FE-0CA4BC82843A@192.168.1.254 CSeq: 46293 INVITE Max-Forwards: 70 Content-Type: application/sdp User-Agent: X-PRO build 1082 Content-Length: 251 v=0 o=101 687354250 687354250 IN IP4 192.168.1.254 s=X-PRO c=IN IP4 192.168.1.254 t=0 0 m=audio 49556 RTP/AVP 18 98 97 101 a=rtpmap:18 G729/8000 a=rtpmap:98 iLBC/8000 a=rtpmap:97 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 [Mar 19 13:39:21] --- (11 headers 11 lines)[Mar 19 13:39:21] --- [Mar 19 13:39:21] Using latest request as basis request [Mar 19 13:39:21] Sending to 192.168.1.254 : 49554 (non-NAT) [Mar 19 13:39:21] Reliably Transmitting (no NAT) to 192.168.1.254:49554: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.1.254:49554;branch=z9hG4bKB252CA9A2D6D46DEAABE588661026F22 From: 101 ;tag=1562601555 To: ;tag=as3cfcc735 Call-ID: B610EE9B-C484-4BA9-A2FE-0CA4BC82843A@192.168.1.254 CSeq: 46293 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Proxy-Authenticate: Digest realm="asterisk02.katherinebishop.com", nonce="5ce352ed" Content-Length: 0 --- [Mar 19 13:39:21] Scheduling destruction of call 'B610EE9B-C484-4BA9-A2FE-0CA4BC82843A@192.168.1.254' in 15000 ms [Mar 19 13:39:21] Found user '101' [Mar 19 13:39:22] <-- SIP read from 192.168.1.254:49554: ACK sip:105@asterisk02 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.254:49554;rport;branch=z9hG4bKB252CA9A2D6D46DEAABE588661026F22 From: 101 ;tag=1562601555 To: ;tag=as3cfcc735 Contact: Call-ID: B610EE9B-C484-4BA9-A2FE-0CA4BC82843A@192.168.1.254 CSeq: 46293 ACK Max-Forwards: 70 Content-Length: 0 [Mar 19 13:39:22] --- (9 headers 0 lines)[Mar 19 13:39:22] --- [Mar 19 13:39:22] <-- SIP read from 192.168.1.254:49554: INVITE sip:105@asterisk02 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.254:49554;rport;branch=z9hG4bKB06B4B247A384F1B8266E697E6E2D4B7 From: 101 ;tag=1562601555 To: Contact: Call-ID: B610EE9B-C484-4BA9-A2FE-0CA4BC82843A@192.168.1.254 CSeq: 46294 INVITE Proxy-Authorization: Digest username="101",realm="asterisk02.katherinebishop.com",nonce="5ce352ed",response="97091ab925f99c5882a89fe869e238de",uri="sip:105@asterisk02" Max-Forwards: 70 Content-Type: application/sdp User-Agent: X-PRO build 1082 Content-Length: 251 v=0 o=101 687355093 687355093 IN IP4 192.168.1.254 s=X-PRO c=IN IP4 192.168.1.254 t=0 0 m=audio 49556 RTP/AVP 18 98 97 101 a=rtpmap:18 G729/8000 a=rtpmap:98 iLBC/8000 a=rtpmap:97 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 [Mar 19 13:39:22] --- (12 headers 11 lines)[Mar 19 13:39:22] --- [Mar 19 13:39:22] Using latest request as basis request [Mar 19 13:39:22] Sending to 192.168.1.254 : 49554 (non-NAT) [Mar 19 13:39:22] Found user '101' [Mar 19 13:39:22] Found RTP audio format 18 [Mar 19 13:39:22] Found RTP audio format 98 [Mar 19 13:39:22] Found RTP audio format 97 [Mar 19 13:39:22] Found RTP audio format 101 [Mar 19 13:39:22] Peer audio RTP is at port 192.168.1.254:49556 [Mar 19 13:39:22] Found description format G729 [Mar 19 13:39:22] Found description format iLBC [Mar 19 13:39:22] Found description format speex [Mar 19 13:39:22] Found description format telephone-event [Mar 19 13:39:22] Capabilities: us - 0x80406 (gsm|ulaw|ilbc|h263), peer - audio=0x700 (g729|speex|ilbc)/video=0x0 (nothing), combined - 0x400 (ilbc) [Mar 19 13:39:22] Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) [Mar 19 13:39:22] Looking for 105 in ldaccess [Mar 19 13:39:22] list_route: hop: [Mar 19 13:39:22] Transmitting (no NAT) to 192.168.1.254:49554: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.254:49554;branch=z9hG4bKB06B4B247A384F1B8266E697E6E2D4B7 From: 101 ;tag=1562601555 To: ;tag=as1cb19647 Call-ID: B610EE9B-C484-4BA9-A2FE-0CA4BC82843A@192.168.1.254 CSeq: 46294 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 --- [Mar 19 13:39:22] -- Executing Macro("SIP/101-e373", "stdexten|105|SIP/105") in new stack [Mar 19 13:39:22] -- Executing Dial("SIP/101-e373", "SIP/105|20") in new stack [Mar 19 13:39:22] We're at 192.168.1.14 port 10220 [Mar 19 13:39:22] Video is at 192.168.1.14 port 12328 [Mar 19 13:39:22] Answering/Requesting with root capability 0x400 (ilbc) [Mar 19 13:39:22] Answering with preferred capability 0x2 (gsm) [Mar 19 13:39:22] Answering with preferred capability 0x4 (ulaw) [Mar 19 13:39:22] Answering with preferred capability 0x80000 (h263) [Mar 19 13:39:22] Answering with non-codec capability 0x1 (telephone-event) [Mar 19 13:39:22] 12 headers, 14 lines [Mar 19 13:39:22] Reliably Transmitting (no NAT) to 192.168.1.105:5060: INVITE sip:105@192.168.1.105:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.14:5060;branch=z9hG4bK3f2ce751;rport From: "Serge Vecher (videophone)" ;tag=as7bacaf5c To: Contact: Call-ID: 7dcb49ad0a383caa47fc49c85bfe4673@192.168.1.14 CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Sat, 19 Mar 2005 18:39:22 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 313 v=0 o=root 1082 1082 IN IP4 192.168.1.14 s=session c=IN IP4 192.168.1.14 t=0 0 m=audio 10220 RTP/AVP 97 3 0 101 a=rtpmap:97 iLBC/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - m=video 12328 RTP/AVP 34 a=rtpmap:34 H263/90000 --- [Mar 19 13:39:22] -- Called 105 [Mar 19 13:39:22] <-- SIP read from 192.168.1.105:50417: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.14:5060;branch=z9hG4bK3f2ce751;rport From: "Serge Vecher (videophone)" ;tag=as7bacaf5c To: Call-ID: 7dcb49ad0a383caa47fc49c85bfe4673@192.168.1.14 Date: Sat, 19 Mar 2005 18:39:24 GMT CSeq: 102 INVITE Server: CSCO/7 Contact: Content-Length: 0 [Mar 19 13:39:22] --- (10 headers 0 lines)[Mar 19 13:39:22] --- [Mar 19 13:39:22] <-- SIP read from 192.168.1.105:50417: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.1.14:5060;branch=z9hG4bK3f2ce751;rport From: "Serge Vecher (videophone)" ;tag=as7bacaf5c To: ;tag=001201dbe43d03be17fa9b41-1c506923 Call-ID: 7dcb49ad0a383caa47fc49c85bfe4673@192.168.1.14 Date: Sat, 19 Mar 2005 18:39:24 GMT CSeq: 102 INVITE Server: CSCO/7 Contact: Content-Length: 0 [Mar 19 13:39:22] --- (10 headers 0 lines)[Mar 19 13:39:22] --- [Mar 19 13:39:22] -- SIP/105-7ef5 is ringing [Mar 19 13:39:22] Transmitting (no NAT) to 192.168.1.254:49554: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.1.254:49554;branch=z9hG4bKB06B4B247A384F1B8266E697E6E2D4B7 From: 101 ;tag=1562601555 To: ;tag=as1cb19647 Call-ID: B610EE9B-C484-4BA9-A2FE-0CA4BC82843A@192.168.1.254 CSeq: 46294 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 [Mar 19 13:39:24] --- (11 headers 11 lines)[Mar 19 13:39:24] --- [Mar 19 13:39:24] Destroying call '320524af730fee470176a921472b1e1b@192.168.1.14' [Mar 19 13:39:29] <-- SIP read from 192.168.1.254:49554: CANCEL sip:105@asterisk02 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.254:49554;rport;branch=z9hG4bKB06B4B247A384F1B8266E697E6E2D4B7 From: 101 ;tag=1562601555 To: ;tag=as1cb19647 Contact: Call-ID: B610EE9B-C484-4BA9-A2FE-0CA4BC82843A@192.168.1.254 CSeq: 46294 CANCEL Proxy-Authorization: Digest username="101",realm="asterisk02.katherinebishop.com",nonce="5ce352ed",response="7063f607442d7e6cc46492751997caec",uri="sip:105@asterisk02" User-Agent: X-PRO build 1082 Content-Length: 0 [Mar 19 13:39:29] --- (10 headers 0 lines)[Mar 19 13:39:29] --- [Mar 19 13:39:29] Sending to 192.168.1.254 : 49554 (non-NAT) [Mar 19 13:39:29] Reliably Transmitting (no NAT) to 192.168.1.254:49554: SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 192.168.1.254:49554;branch=z9hG4bKB06B4B247A384F1B8266E697E6E2D4B7 From: 101 ;tag=1562601555 To: ;tag=as1cb19647 Call-ID: B610EE9B-C484-4BA9-A2FE-0CA4BC82843A@192.168.1.254 CSeq: 46294 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 --- [Mar 19 13:39:29] Transmitting (no NAT) to 192.168.1.254:49554: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.254:49554;branch=z9hG4bKB06B4B247A384F1B8266E697E6E2D4B7 From: 101 ;tag=1562601555 To: ;tag=as1cb19647 Call-ID: B610EE9B-C484-4BA9-A2FE-0CA4BC82843A@192.168.1.254 CSeq: 46294 CANCEL User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 --- [Mar 19 13:39:29] Reliably Transmitting (no NAT) to 192.168.1.105:5060: CANCEL sip:105@192.168.1.105:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.14:5060;branch=z9hG4bK3f2ce751;rport From: "Serge Vecher (videophone)" ;tag=as7bacaf5c To: Contact: Call-ID: 7dcb49ad0a383caa47fc49c85bfe4673@192.168.1.14 CSeq: 102 CANCEL User-Agent: Asterisk PBX Content-Length: 0 --- [Mar 19 13:39:29] Scheduling destruction of call '7dcb49ad0a383caa47fc49c85bfe4673@192.168.1.14' in 15000 ms [Mar 19 13:39:29] <-- SIP read from 192.168.1.254:49554: ACK sip:105@asterisk02 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.254:49554;rport;branch=z9hG4bKB06B4B247A384F1B8266E697E6E2D4B7 From: 101 ;tag=1562601555 To: ;tag=as1cb19647 Contact: Call-ID: B610EE9B-C484-4BA9-A2FE-0CA4BC82843A@192.168.1.254 CSeq: 46294 ACK Max-Forwards: 70 Content-Length: 0 [Mar 19 13:39:29] --- (9 headers 0 lines)[Mar 19 13:39:29] --- [Mar 19 13:39:29] == Spawn extension (macro-stdexten, s, 1) exited non-zero on 'SIP/101-e373' in macro 'stdexten' [Mar 19 13:39:29] == Spawn extension (ldaccess, 105, 1) exited non-zero on 'SIP/101-e373' [Mar 19 13:39:29] Destroying call 'B610EE9B-C484-4BA9-A2FE-0CA4BC82843A@192.168.1.254' [Mar 19 13:39:29] <-- SIP read from 192.168.1.254:49554: ACK sip:105@asterisk02 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.254:49554;rport;branch=z9hG4bKB06B4B247A384F1B8266E697E6E2D4B7 From: 101 ;tag=1562601555 To: ;tag=as1cb19647 Contact: Call-ID: B610EE9B-C484-4BA9-A2FE-0CA4BC82843A@192.168.1.254 CSeq: 46294 ACK Max-Forwards: 70 Content-Length: 0 [Mar 19 13:39:29] --- (9 headers 0 lines)[Mar 19 13:39:29] --- [Mar 19 13:39:29] Destroying call 'B610EE9B-C484-4BA9-A2FE-0CA4BC82843A@192.168.1.254' [Mar 19 13:39:29] <-- SIP read from 192.168.1.105:50417: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.14:5060;branch=z9hG4bK3f2ce751;rport From: "Serge Vecher (videophone)" ;tag=as7bacaf5c To: ;tag=001201dbe43d03be17fa9b41-1c506923 Call-ID: 7dcb49ad0a383caa47fc49c85bfe4673@192.168.1.14 Date: Sat, 19 Mar 2005 18:39:31 GMT CSeq: 102 CANCEL Server: CSCO/7 Content-Length: 0 [Mar 19 13:39:29] --- (9 headers 0 lines)[Mar 19 13:39:29] --- [Mar 19 13:39:29] <-- SIP read from 192.168.1.105:50417: SIP/2.0 487 Request Cancelled Via: SIP/2.0/UDP 192.168.1.14:5060;branch=z9hG4bK3f2ce751;rport From: "Serge Vecher (videophone)" ;tag=as7bacaf5c To: ;tag=001201dbe43d03be17fa9b41-1c506923 Call-ID: 7dcb49ad0a383caa47fc49c85bfe4673@192.168.1.14 Date: Sat, 19 Mar 2005 18:39:31 GMT CSeq: 102 INVITE Server: CSCO/7 Contact: Content-Length: 0 [Mar 19 13:39:29] --- (10 headers 0 lines)[Mar 19 13:39:29] --- [Mar 19 13:39:29] Transmitting (no NAT) to 192.168.1.105:5060: ACK sip:105@192.168.1.105:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.14:5060;branch=z9hG4bK3f2ce751;rport From: "Serge Vecher (videophone)" ;tag=as7bacaf5c To: ;tag=001201dbe43d03be17fa9b41-1c506923 Contact: Call-ID: 7dcb49ad0a383caa47fc49c85bfe4673@192.168.1.14 CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0