Cannot get incoming calls to route DIDforSale FreePBX 2.10.0.4

I have a fresh install of FreePBX 2.10.0.4 running. I had my DIDs configured with my previous Tr*xbox setup and all was working. I copied all my configs for the trunks and installed FreePBX and pasted all my trunk configs into the new installation. They didn’t work. Caller hears “The number is not in service.” Asterisk Logs show the following “unknown peer” messages and the call is dumped.

[2012-04-17 14:50:38] VERBOSE[3231] netsock2.c: == Using SIP RTP TOS bits 184 [2012-04-17 14:50:38] VERBOSE[3231] netsock2.c: == Using SIP RTP CoS mark 5 [2012-04-17 14:50:38] VERBOSE[12221] pbx.c: -- Executing [[email protected]:1] NoOp("SIP/65.98.234.222-0000002a", "Received incoming SIP connection from unknown peer to 1925XXXXX66") in new stack [2012-04-17 14:50:38] VERBOSE[12221] pbx.c: -- Executing [[email protected]:2] Set("SIP/65.98.234.222-0000002a", "DID=1925XXXXX66") in new stack [2012-04-17 14:50:38] VERBOSE[12221] pbx.c: -- Executing [[email protected]:3] Goto("SIP/65.98.234.222-0000002a", "s,1") in new stack [2012-04-17 14:50:38] VERBOSE[12221] pbx.c: -- Goto (from-sip-external,s,1) [2012-04-17 14:50:38] VERBOSE[12221] pbx.c: -- Executing [[email protected]:1] GotoIf("SIP/65.98.234.222-0000002a", "0?checklang:noanonymous") in new stack [2012-04-17 14:50:38] VERBOSE[12221] pbx.c: -- Goto (from-sip-external,s,5) [2012-04-17 14:50:38] VERBOSE[12221] pbx.c: -- Executing [[email protected]:5] Set("SIP/65.98.234.222-0000002a", "TIMEOUT(absolute)=15") in new stack [2012-04-17 14:50:38] VERBOSE[12221] func_timeout.c: Channel will hangup at 2012-04-17 14:50:53.342 PDT. [2012-04-17 14:50:38] VERBOSE[12221] pbx.c: -- Executing [[email protected]:6] Answer("SIP/65.98.234.222-0000002a", "") in new stack [2012-04-17 14:50:38] VERBOSE[12221] pbx.c: -- Executing [[email protected]:7] Wait("SIP/65.98.234.222-0000002a", "2") in new stack [2012-04-17 14:50:40] VERBOSE[12221] pbx.c: -- Executing [[email protected]:8] Playback("SIP/65.98.234.222-0000002a", "ss-noservice") in new stack [2012-04-17 14:50:40] VERBOSE[12221] file.c: -- <SIP/65.98.234.222-0000002a> Playing 'ss-noservice.ulaw' (language 'en') [2012-04-17 14:50:46] VERBOSE[12221] pbx.c: -- Executing [[email protected]:9] PlayTones("SIP/65.98.234.222-0000002a", "congestion") in new stack [2012-04-17 14:50:46] VERBOSE[12221] pbx.c: -- Executing [[email protected]:10] Congestion("SIP/65.98.234.222-0000002a", "5") in new stack [2012-04-17 14:50:51] VERBOSE[12221] pbx.c: == Spawn extension (from-sip-external, s, 10) exited non-zero on 'SIP/65.98.234.222-0000002a' [2012-04-17 14:50:51] VERBOSE[12221] pbx.c: -- Executing [[email protected]:1] Hangup("SIP/65.98.234.222-0000002a", "") in new stack [2012-04-17 14:50:51] VERBOSE[12221] pbx.c: == Spawn extension (from-sip-external, h, 1) exited non-zero on 'SIP/65.98.234.222-0000002a'

The config look like this:

type=peer
host=209.216.2.211&209.216.15.70
nat=yes
canreinvite=no
disallow=all
allow=ulaw&alaw
dtmfmode=rfc2833
insecure=very
context=from-didforsale

I also have an inbound route setup with the DID specified. (Also tried it without DID.)

Also, can someone enlighten me on what the “SIP/XX.XX.XX.XX-222…” means? When I did a revered IP lookup it came back to the IP address of some ViOP provider (Probably DIDforSale) however this is not the IP addresses they provide for their registration. I tried using that IP address on the config and had no affect.

Also: the FreePBX box is located behind a Zentyal System acting as Firewall/Infrastructure. The built in asterisk module on Zentyal is disabled. I have Packet Filtering set to send all VoIP to the box and port forward sending 5060 and 10000-20000 to the box. I intend to shrink the media port range once I get everything back up and running.

It’s quiet in here… I have tried the recommendations of just about every post regarding DIDforSale and nothing works. Is there a problem with this release of FreePBX?

Even the DIDforSale support staff cannot figure out how to make it work.
This worked fine in FreePBX 1.6 and Trixbox. I’m starting to wonder if this is a bug somewhere. If it worked in previous versions and all of a sudden breaks…

DIDforSale mentioned that the IP address is showing up on the FreePBX box as the IP of THEIR vendor and not DIDforSale. They can’t figure out how my FreePBX is even seeing that IP. Since they are an IP based authentication provider this presents problems with authenticating the incoming SIP call.

It has to be an Asterisk 1.8 issue with reinvite behavior.

More carriers invite the media stream off their networks to the downstream carrier.