Incoming SIP calls connect but end after being answered

Question

I'm using Asterisk 11, a Cisco SPA303 Phone, and Twilio.

I can make outgoing phone calls without any issue and the call quality is top notch. On an incoming call however, my extension (and phone) ring, however when I answer the phone, there is no audio on either end and 30 seconds later, both calls end. Using Twilio's PCAP log, it shows that my asterisk server sends a BYE when answered. Asterisk however does not log a single thing on incoming calls. (All SIP traffic is logged on outgoing calls). Does anyone have any idea what's going on?

Update: Turns out that I had my incoming extensions in the wrong context, once that was updated, incoming and outgoing calls worked without a hitch. I marked @arheops as the answer because the logging of unanswered calls lead to the diagnosing of the problem.

Answer 1

Very likly(but you not informed) your asterisk installation is after NAT.

IF so, you have configure asterisk as described in http://www.voip-info.org/wiki/index.php?page_id=410

Also may need change ports on router/disable SIP ALG on router etc.

Logging(you mean cdr,right?) is controled by /etc/asterisk/cdr.conf

Most likly you have ananswered=no in that file.