SIP Debugging Enabled for IP: 192.168.11.114 asterisk1*CLI> <-- SIP read from 192.168.11.114:5060: INVITE sip:*67@192.168.11.112;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.11.114:5060 Max-Forwards: 70 From: 63 ;tag=88d587b5a533701e To: Call-ID: 89ce9cf1c4819af0@192.168.11.114 CSeq: 1 INVITE Contact: 63 Supported: timer Session-Expires: 180 Content-Type: application/sdp Content-Length: 194 v=0 o=- 1 0 IN IP4 192.168.11.114 s=- c=IN IP4 192.168.11.114 t=0 0 m=audio 49152 RTP/AVP 18 0 8 2 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:2 G726/8000 --- (12 headers 10 lines) --- Using INVITE request as basis request - 89ce9cf1c4819af0@192.168.11.114 Sending to 192.168.11.114 : 5060 (non-NAT) Reliably Transmitting (NAT) to 192.168.11.114:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.11.114:5060;received=192.168.11.114 From: 63 ;tag=88d587b5a533701e To: ;tag=as7f369d7c Call-ID: 89ce9cf1c4819af0@192.168.11.114 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="657b2418" Content-Length: 0 --- Scheduling destruction of call '89ce9cf1c4819af0@192.168.11.114' in 15000 ms Found user '63' asterisk1*CLI> <-- SIP read from 192.168.11.114:5060: ACK sip:*67@192.168.11.112;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.11.114:5060 Max-Forwards: 70 From: 63 ;tag=88d587b5a533701e To: ;tag=as7f369d7c Call-ID: 89ce9cf1c4819af0@192.168.11.114 CSeq: 1 ACK Content-Length: 0 --- (8 headers 0 lines) --- asterisk1*CLI> <-- SIP read from 192.168.11.114:5060: INVITE sip:*67@192.168.11.112;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.11.114:5060 Max-Forwards: 70 From: 63 ;tag=88d587b5a533701e To: Call-ID: 89ce9cf1c4819af0@192.168.11.114 CSeq: 2 INVITE Contact: 63 Supported: timer Session-Expires: 180 Proxy-Authorization: Digest username="63", realm="asterisk", nonce="657b2418", opaque="", uri="sip:*67@192.168.11.112", response="1e6ffa4f0921b45b3aea1ae92b70918e" Content-Type: application/sdp Content-Length: 194 v=0 o=- 1 0 IN IP4 192.168.11.114 s=- c=IN IP4 192.168.11.114 t=0 0 m=audio 49152 RTP/AVP 18 0 8 2 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:2 G726/8000 --- (13 headers 10 lines) --- Using INVITE request as basis request - 89ce9cf1c4819af0@192.168.11.114 Sending to 192.168.11.114 : 5060 (NAT) Found user '63' Found RTP audio format 18 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 2 Peer audio RTP is at port 192.168.11.114:49152 Found description format G729 Found description format PCMU Found description format PCMA Found description format G726 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Looking for *67 in from-internal (domain 192.168.11.112;user=phone) Reliably Transmitting (NAT) to 192.168.11.114:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.11.114:5060;received=192.168.11.114 From: 63 ;tag=88d587b5a533701e To: ;tag=as7f369d7c Call-ID: 89ce9cf1c4819af0@192.168.11.114 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- asterisk1*CLI> <-- SIP read from 192.168.11.114:5060: ACK sip:*67@192.168.11.112;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.11.114:5060 Max-Forwards: 70 From: 63 ;tag=88d587b5a533701e To: ;tag=as7f369d7c Call-ID: 89ce9cf1c4819af0@192.168.11.114 CSeq: 2 ACK Content-Length: 0 --- (8 headers 0 lines) --- Destroying call '89ce9cf1c4819af0@192.168.11.114'