asterisk users - Sep 2005 - asterisk@home inbound call problem to SIP trunk. (voipfone UK)

Hi all,

I have recently installed Asterisk@home and outbound calling is
working great. However I am strugglings with inbound calls. I have
setup a trunk for my provider, voipfone and in the inbound area on AMP
I have the following :-

user context name = 3011XXXX

context=from-pstn
dtmfmode=rfc2283
fromdomain=voipfone.co.uk
host=voipfone.co.uk
insecure=very
secret=XXXXXX
type=user
user=3011XXXX
username=3011XXXX

To be honest a lot of this is guesswork so could be wrong. I've tried
a lot of others settings sut still get no inbound calls. I also went
into inbound routing and created a default route with icoming calls
sent to my extention. That is all I have done.

If I call my PSTN number from the PSTN i get a log entry and it shows
the calling PSTN number so It looks to me as though the trunk must be
okay as the call is getting routed to my Asterisk, or am I mistaken
with this? Does anyone know what "Failed to authenticate user
"0792124000" ;tag=as16492b07" means? Is it something to do with
my
inbound context? "07921 24000" is the PSTN number.

here is the full log extract.

p 27 14:30:53 DEBUG[2618] chan_sip.c: Stopping retransmission on
'4f420dd13a83403c5b25f34a13deb6ad@127.0.0.1' of Request 129: Match
Found
Sep 27 14:30:53 DEBUG[2618] chan_sip.c: Registration successful
Sep 27 14:30:53 DEBUG[2618] chan_sip.c: Cancelling timeout 14095
Sep 27 14:31:03 DEBUG[2618] chan_iax2.c: Peer lastms 33, historicms
33, maxms 2000
Sep 27 14:31:09 DEBUG[2618] manager.c: Manager received command
'Command'
Sep 27 14:31:09 DEBUG[2618] manager.c: Manager received command
'Command'
Sep 27 14:31:09 DEBUG[2618] manager.c: Manager received command
'MailboxStatus'
Sep 27 14:31:25 DEBUG[2618] chan_sip.c: Auto destroying call
'4f420dd13a83403c5b25f34a13deb6ad@127.0.0.1'
Sep 27 14:31:27 DEBUG[2618] chan_sip.c: Setting NAT on RTP to 0
Sep 27 14:31:27 DEBUG[2618] chan_sip.c: Stopping retransmission on
'2146758775271ffe538e69644749ceaa@212.187.162.178' of Response 102:
Match Found
Sep 27 14:31:27 DEBUG[2618] chan_sip.c: Setting NAT on RTP to 0
Sep 27 14:31:27 NOTICE[2618] chan_sip.c: Failed to authenticate user
"07921249135" ;tag=as16492b07
Sep 27 14:31:27 DEBUG[2618] chan_sip.c: Stopping retransmission on
'2146758775271ffe538e69644749ceaa@212.187.162.178' of Response 103:
Match Found
Sep 27 14:31:38 DEBUG[2618] chan_sip.c: Scheduled a registration
timeout for voipfone.co.uk id #14103
Sep 27 14:31:38 DEBUG[2618] chan_sip.c: (Provisional) Stopping
retransmission (but retaining packet) on
'4f420dd13a83403c5b25f34a13deb6ad@127.0.0.1' Request 130: Found
Sep 27 14:31:38 DEBUG[2618] chan_sip.c: Stopping retransmission on
'4f420dd13a83403c5b25f34a13deb6ad@127.0.0.1' of Request 130: Match
Found
Sep 27 14:31:38 DEBUG[2618] chan_sip.c: (Provisional) Stopping
retransmission (but retaining packet) on
'4f420dd13a83403c5b25f34a13deb6ad@127.0.0.1' Request 131: Found
Sep 27 14:31:38 DEBUG[2618] chan_sip.c: Stopping retransmission on
'4f420dd13a83403c5b25f34a13deb6ad@127.0.0.1' of Request 131: Match
Found
Sep 27 14:31:38 DEBUG[2618] chan_sip.c: Registration successful
Sep 27 14:31:38 DEBUG[2618] chan_sip.c: Cancelling timeout 14103
Sep 27 14:31:38 DEBUG[2618] chan_sip.c: Cancelling timeout 14103

I've tried for a week now and could really use some help!

Thanks
Steve