Quintum AFM400 trunk only half working

Hi

I am having a problem routing call out via my quintum AFM400
Any assistance woul be greatly appreciated I have included

When calling a SIP extension on the asterisk server from a handset on the quintum

-- Executing [s@macro-dial:7] Dial("SIP/quintum-000001fe", "SIP/118||TtrWw") in new stack
    -- Called 118
    -- SIP/118-000001ff is ringing

This works correctly I get ringing and can answer the call

When calling the number for the pstn line that is pluged into the first FXO port on the quintum I get

- Executing [s@from-sip-external:1] GotoIf("SIP/quintum-00000203", "1?from-trunk||1") in new stack
    -- Goto (from-trunk,s,1)
    -- Executing [s@from-trunk:1] Set("SIP/quintum-00000203", "__FROM_DID=s") in new stack
    -- Executing [s@from-trunk:2] Gosub("SIP/quintum-00000203", "app-blacklist-check|s|1") in new stack
    -- Executing [s@app-blacklist-check:1] LookupBlacklist("SIP/quintum-00000203", "") in new stack
    -- Executing [s@app-blacklist-check:2] GotoIf("SIP/quintum-00000203", "0?blacklisted") in new stack
    -- Executing [s@app-blacklist-check:3] Return("SIP/quintum-00000203", "") in new stack
    -- Executing [s@from-trunk:3] ExecIf("SIP/quintum-00000203", "1 |Set|CALLERID(name)=anonymous") in new stack
    -- Executing [s@from-trunk:4] Set("SIP/quintum-00000203", "__CALLINGPRES_SV=allowed_not_screened") in new stack
    -- Executing [s@from-trunk:5] SetCallerPres("SIP/quintum-00000203", "allowed_not_screened") in new stack
    -- Executing [s@from-trunk:6] Goto("SIP/quintum-00000203", "app-announcement-1|s|1") in new stack                                       

some other stuff

-- Executing [s@macro-dial:7] Dial("SIP/quintum-00000203", "SIP/10|45|TtrWwM(auto-blkvm)") in new stack
    -- Called 10
    -- SIP/10-00000205 is ringing
    -- SIP/10-00000205 is ringing

ok it works

I have configured a trunk to the quintum "SIP/quintum"

host=10.168.253.101
type=peer
allow=all

, but it never registers

 NOTICE[3542] chan_sip.c:    -- Registration for '4111@10.168.253.101' timed out, trying again (Attempt #31)
 NOTICE[3542] chan_sip.c:    -- Registration for '4111@10.168.253.101' timed out, trying again (Attempt #32)
 NOTICE[3542] chan_sip.c:    -- Registration for '4111@10.168.253.101' timed out, trying again (Attempt #33)
 NOTICE[3542] chan_sip.c:    -- Registration for '4111@10.168.253.101' timed out, trying again (Attempt #34)

and when I route calls over it that originate from the asterisk VoIP extensions It fails.

 NOTICE[9146] app_dial.c: Hey! chan DAHDI/3-1's context='macro-dialout-trunk', and exten='s'
 VERBOSE[9146] logger.c:     -- Called quintum/0738611808
 VERBOSE[3542] logger.c:     -- Got SIP response 503 "Service Unavailable" back from 10.168.253.101
 VERBOSE[9146] logger.c:     -- SIP/quintum-0000021e is circuit-busy
 VERBOSE[9146] logger.c:   == Everyone is busy/congested at this time (1:0/1/0)

Could somebody please assist me with getting this working.
I have tried several configurations that ppl have claimed to have had working for them, but I am not having any joy.

regards