I’ve got Asterisk @ Home 1.5 set up (with an el-cheapo FX100 card from eBay), and a DTA-310 flashed like so:
Application Code Version: SIP version 1.0 US (LDTK AR16O ) checksum e423
Downloader Code Version: 1.0 US (8x8 010116) checksum 3e91
I can get in to the DTA web config just fine and play with the SIP settings. I have them configured per these instructions:
From a X-Lite softphone on my Mac, I can dial 9xxxxxxx out through the FX100 to POTS, and I can dial “special” numbers on the Asterisk box like *60 for time, etc. The trunk and extension show the expected behavior on the Flash Operator Panel.
When I dial *60 on the phone connected to the DTA-310, I get a busy signal. If I dial 9xxxxxxx to call a POTS number, it crashes the DTA (need to power cycle to get a dialtone).
In the asterisk log, I’m seeing a lot of these, every minute or so:
Sep 13 03:07:03 DEBUG: Setting NAT on RTP to 0
Sep 13 03:07:03 DEBUG: Stopping retransmission on ‘email@example.com’ of Request 102: Found
Sep 13 03:07:03 VERBOSE: – Got SIP response 400 “Content To Short” back from 192.168.0.205
Sep 13 03:07:12 DEBUG: Manager received command 'Command’
Sep 13 03:07:12 DEBUG: Manager received command ‘Command’
Sep 13 03:07:14 DEBUG: Auto destroying call 'firstname.lastname@example.org*’
The SIP settings on the DTA and the extension config in Asterisk match (password, port, etc).
Any ideas? Thanks.