<--- SIP read from 172.24.8.130:50676 ---> INVITE sip:4802214606@172.24.8.36 SIP/2.0 Via: SIP/2.0/UDP 172.24.8.130:5060;branch=z9hG4bK5ce68a79 From: "101" ;tag=00115c430951001054cfaa5d-3550bcb0 To: Call-ID: 00115c43-09510011-137ea541-730a6ebc@172.24.8.130 CSeq: 101 INVITE User-Agent: CSCO/6 Contact: Expires: 180 Content-Type: application/sdp Content-Length: 246 Accept: application/sdp v=0 o=Cisco-SIPUA 24456 1744 IN IP4 172.24.8.130 s=SIP Call c=IN IP4 172.24.8.130 t=0 0 m=audio 20704 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> --- (12 headers 11 lines) --- Sending to 172.24.8.130 : 5060 (no NAT) Using INVITE request as basis request - 00115c43-09510011-137ea541-730a6ebc@172.24.8.130 Found no matching peer or user for '172.24.8.130:50676' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 172.24.8.130:20704 Found description format PCMU for ID 0 Found description format PCMA for ID 8 Found description format G729 for ID 18 Found description format telephone-event for ID 101 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 172.24.8.130:20704 Looking for 4802214606 in default (domain 172.24.8.36) <--- Reliably Transmitting (no NAT) to 172.24.8.130:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 172.24.8.130:5060;branch=z9hG4bK5ce68a79;received=172.24.8.130 From: "101" ;tag=00115c430951001054cfaa5d-3550bcb0 To: ;tag=as5b2a8bf7 Call-ID: 00115c43-09510011-137ea541-730a6ebc@172.24.8.130 CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> Scheduling destruction of SIP dialog '00115c43-09510011-137ea541-730a6ebc@172.24.8.130' in 32000 ms (Method: INVITE) localhost*CLI> <--- SIP read from 172.24.8.130:50685 ---> ACK sip:4802214606@172.24.8.36 SIP/2.0 Via: SIP/2.0/UDP 172.24.8.130:5060;branch=z9hG4bK5ce68a79 From: "101" ;tag=00115c430951001054cfaa5d-3550bcb0 To: ;tag=as5b2a8bf7 Call-ID: 00115c43-09510011-137ea541-730a6ebc@172.24.8.130 CSeq: 101 ACK Content-Length: 0