Sipgate.de incoming route problem

I’m having trouble configuring FreePBX to work with sipgate.de. I can get the outbound trunk to work, so the trunk connects. However, when I try the inbound route, the call hits my PBX, but it doesn’t ring my phone. Here is the log:

[2013-11-08 16:21:23] VERBOSE[2268][C-00005964] netsock2.c: == Using SIP RTP TOS bits 184
[2013-11-08 16:21:23] VERBOSE[2268][C-00005964] netsock2.c: == Using SIP RTP CoS mark 5
[2013-11-08 16:21:23] VERBOSE[13902][C-00005964] pbx.c: – Executing [[email protected]:1] NoOp(“SIP/SipgateTrunk-0000a9d3”, “No DID or CID Match”) in new stack
[2013-11-08 16:21:23] VERBOSE[13902][C-00005964] pbx.c: – Executing [[email protected]:2] Answer(“SIP/SipgateTrunk-0000a9d3”, “”) in new stack
[2013-11-08 16:21:24] VERBOSE[13902][C-00005964] pbx.c: – Executing [[email protected]:3] Wait(“SIP/SipgateTrunk-0000a9d3”, “2”) in new stack
[2013-11-08 16:21:26] VERBOSE[13902][C-00005964] pbx.c: – Executing [[email protected]:4] Playback(“SIP/SipgateTrunk-0000a9d3”, “ss-noservice”) in new stack
[2013-11-08 16:21:26] VERBOSE[13902][C-00005964] file.c: – <SIP/SipgateTrunk-0000a9d3> Playing ‘ss-noservice.ulaw’ (language ‘en’)
[2013-11-08 16:21:31] VERBOSE[13902][C-00005964] pbx.c: – Executing [[email protected]:5] SayAlpha(“SIP/SipgateTrunk-0000a9d3”, “”) in new stack
[2013-11-08 16:21:31] VERBOSE[13902][C-00005964] pbx.c: – Executing [[email protected]:6] Hangup(“SIP/SipgateTrunk-0000a9d3”, “”) in new stack
[2013-11-08 16:21:31] VERBOSE[13902][C-00005964] pbx.c: == Spawn extension (from-trunk, s, 6) exited non-zero on ‘SIP/SipgateTrunk-0000a9d3’
[2013-11-08 16:21:31] VERBOSE[13902][C-00005964] pbx.c: – Executing [[email protected]:1] Macro(“SIP/SipgateTrunk-0000a9d3”, “hangupcall,”) in new stack
[2013-11-08 16:21:31] VERBOSE[13902][C-00005964] pbx.c: – Executing [[email protected]:1] GotoIf(“SIP/SipgateTrunk-0000a9d3”, “1?theend”) in new stack
[2013-11-08 16:21:31] VERBOSE[13902][C-00005964] pbx.c: – Goto (macro-hangupcall,s,3)
[2013-11-08 16:21:31] VERBOSE[13902][C-00005964] pbx.c: – Executing [[email protected]:3] ExecIf(“SIP/SipgateTrunk-0000a9d3”, “0?Set(CDR(recordingfile)=)”) in new stack
[2013-11-08 16:21:31] VERBOSE[13902][C-00005964] pbx.c: – Executing [[email protected]:4] Hangup(“SIP/SipgateTrunk-0000a9d3”, “”) in new stack
[2013-11-08 16:21:31] VERBOSE[13902][C-00005964] app_macro.c: == Spawn extension (macro-hangupcall, s, 4) exited non-zero on ‘SIP/SipgateTrunk-0000a9d3’ in macro ‘hangupcall’
[2013-11-08 16:21:31] VERBOSE[13902][C-00005964] pbx.c: == Spawn extension (from-trunk, h, 1) exited non-zero on ‘SIP/SipgateTrunk-0000a9d3’

So that tells me that the call is hitting my PBX, but it doesn’t know what DID to put the call through to. I’ve tried putting the SIP-ID in the DID, and I’ve tried putting the phone number in the DID. So far, nothing works. Here is my trunk config:

PEER DETAILS

username=***
type=peer
secret=***
qualify=yes
nat=yes
insecure=very
host=sipgate.de
fromuser=***
dtmfmode=rfc2833
disallow=all
context=from-trunk
canreinvite=yes
authuser=***
allow=ulaw

USER CONTEXT: from-trunk

USER DETAILS:

[blank]

REGISTER STRING:
sipid:[email protected]/sipid

What am I doing wrong? I am using Asterisk version 11.2.1. Been trying to get this work for a good 5 hours now, and I feel that I’m really not far off. Any help would be much appreciated.

Thank you