--- (13 headers 9 lines)--- Using INVITE request as basis request - MDJkZjEzMjRlZDU3NmFjOTViNDMzOWQxZTNkNmEy NGU. Sending to 192.168.1.2 : 56540 (NAT) Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 0.0.0.0:59108 Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (noth ing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event) , combined - 0x1 (telephone-event) -- Music class default requested but no musiconhold loaded. We're at 192.168.1.9 port 12622 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 192.168.1.2:56540: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.2:56540;branch=z9hG4bK-d87543-600e1c27d2070c6a-1--d87 543-;received=192.168.1.2;rport=56540 From: "voip";tag=0e14124f To: "98187827803";tag=as2d928095 Call-ID: MDJkZjEzMjRlZDU3NmFjOTViNDMzOWQxZTNkNmEyNGU. CSeq: 5 INVITE User-Agent: Linksys/SPA2102-3.3.5(a) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 236 v=0 o=root 2762 2765 IN IP4 192.168.1.9 s=session c=IN IP4 192.168.1.9 t=0 0 m=audio 12622 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=silenceSupp:off - - - - --- asterisk1*CLI> <-- SIP read from 192.168.1.2:56540: ACK sip:98187827803@192.168.1.9 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.2:56540;branch=z9hG4bK-d87543-4e3aa421064ec167-1--d87 543-;rport Max-Forwards: 70 Contact: To: "98187827803";tag=as2d928095 From: "voip";tag=0e14124f Call-ID: MDJkZjEzMjRlZDU3NmFjOTViNDMzOWQxZTNkNmEyNGU. CSeq: 5 ACK Proxy-Authorization: Digest username="2000",realm="asterisk",nonce="376d9f9b",ur i="sip:98187827803@192.168.1.9",response="334f1d73e73aac5db30ebf30794c36f7",algo rithm=MD5 User-Agent: X-Lite release 1006e stamp 34025 Content-Length: 0