We're at 192.168.1.101 port 19536
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 216.18.125.7:5065:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 216.18.125.7:5065;branch=z9hG4bK-d87543-961e467b16872d607e64-1-cHAyMGRlNWY2OWRmMDY1ZjUwYWEwMg..-d87543-;received=216.18.125.7
From: "anonymous"<sip:anonymous@anonymous.invalid>;tag=d3aeaa05
To: <sip:18198502523@216.18.125.3;user=phone>;tag=as66f3f883
Call-ID: 2f4ff91a273a0f436b8d017e26c89133
CSeq: 185851410 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:s@192.168.1.101>
Content-Type: application/sdp
Content-Length: 216