[Asterisk-Users] Strange problem with call hangup on Budgetone 102 Phones

[Jon Fautley]

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 at 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 at 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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