asterisk users - Jan 2004 - Strange problem with call hangup on Budgetone 102 Phones

Hi,

I've got Asterisk configured and working (sort of) with an Eicon Diva Server
2M ISDN card (connected to S0 bus of another PBX). This * box is on a
'live', non-nat IP address.
I also have a couple of budgetone phones, one behind NAT and one not. When I
place an outgoing call, I get the following messages:

-- Executing Dial("SIP/filbert-9876", "CAPI/288:333") in new
stack
    -- creating pipe for PLCI=-1
       > sent CONNECT_REQ MN =0x5
    -- Called 288:333
    -- Setting up echo canceller (PLCI=0x201, function=1, options=2, tail=64)
       > sent FACILITY_REQ (PLCI=0x201)
    -- CAPI[contr1/288]/0 answered SIP/filbert-9876
    -- Echo canceller successfully set up (PLCI=0x201)
WARNING[9226]: File chan_sip.c, Line 464 (retrans_pkt): Maximum retries exceeded
on call 03ed3583-2cba-cbfc-a86f-605f7f57f024@213.152.57.46 for seqno 102
(Request)
    -- CAPI Hangingup
       > sent DISCONNECT_B3_REQ NCCI=0xa0201
       > sent DISCONNECT_REQ PLCI=0x201
    -- removed pipe for PLCI = 0x201
  == Spawn extension (sip, 9333, 1) exited non-zero on
'SIP/filbert-9876'
WARNING[9226]: File chan_sip.c, Line 464 (retrans_pkt): Maximum retries exceeded
on call 03ed3583-2cba-cbfc-a86f-605f7f57f024@213.152.57.46 for seqno 103
(Request)

I can hear the voicemail service (extn. 333) answer correctly, but then after
about 5 seconds i'll get the WARNING message and the system will hangup.

Here's a snippet from my sip.conf file:

-------
[general]
port = 5060
bindaddr = 0.0.0.0
context = sip-incoming
srvlookup=no
qualify=yes
disallow=all
allow=alaw
allow=ulaw

[filbert]
type=friend
host=dynamic
dtmfmode=info
context=sip
callerid="Jon Fautley" <200>
nat=yes
pickupgroup=1
reinvite=no
canreinvite=no
disallow=all
allow=ulaw
--------


Any ideas?

Many thanks,

Jon
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