asterisk users - Jul 2004 - Audio cuts off 10 minutes into calls

Hello list,

We run Asterisk CVS-HEAD-06/02/04-11:25:18 built by root@Gate01 on a i686
running Linux.

All works fine except Audio is lost 10minutes into the call. This happens
for every call
PSTN-SIP, SIP-PSTN, SIP-SIP

Example of one call setup using Snom200 and Grandstream 486:
-- Executing Macro("SIP/xxxx1251-d638",
"CFW|xxxx1251|SIP/xxxx1253") in new
stack
    -- Executing DBget("SIP/xxxx1251-d638",
"temp=CFIM/xxxx1253") in new
stack
    -- DBget: varname=temp, family=CFIM, key=xxxx1253
    -- DBget: Value not found in database.
    -- Executing Goto("SIP/xxxx1251-d638", "s|4") in new
stack
    -- Goto (macro-CFW,s,4)
    -- Executing Dial("SIP/xxxx1251-d638",
"SIP/xxxx1253|30|t") in new stack
    -- Called xxxx1253
    -- SIP/xxxx1253-c5dc is ringing
    -- SIP/xxxx1253-c5dc answered SIP/xxxx1251-d638
    -- Attempting native bridge of SIP/xxxx1251-d638 and SIP/xxxx1253-c5dc
    -- Attempting native bridge of SIP/xxxx1251-d638 and SIP/xxxx1253-c5dc

PSTN to SIP(Grandstream):
-- Executing Macro("Zap/15-1", "CFW|xxxx1253|SIP/xxxx1253")
in new stack
    -- Executing DBget("Zap/15-1", "temp=CFIM/xxxx1253") in
new stack
    -- DBget: varname=temp, family=CFIM, key=xxxx1253
    -- DBget: Value not found in database.
    -- Executing Goto("Zap/15-1", "s|4") in new stack
    -- Goto (macro-CFW,s,4)
    -- Executing Dial("Zap/15-1", "SIP/xxxx1253|30|t") in
new stack
    -- Called xxxx1253
    -- Accepting call from 'xxxx6857' to 'xxxx1253' on channel
0/15, span 1
    -- SIP/xxxx1253-bc29 is ringing
    -- SIP/xxxx1253-bc29 answered Zap/15-1

I have set verbose 5, and nothing else is reported when audio is lost, when
I hang up the call some time after audio is lost
this is reported:(For PSTN-SIP(Grandstream)

Spawn extension (macro-CFW, s, 4) exited non-zero on 'Zap/15-1' in macro
'CFW'
  == Spawn extension (default, xxxx1253, 1) exited non-zero on
'Zap/15-1'
    -- Hungup 'Zap/15-1'

The call is not hung up, just loss of audio.

I have searched the archives and google without any luck.

Could someone pls give me a pointer of what may be the cause of this
problem.

We use TE410 PRI card, and the SIP clients are: Grandstream HandyTone 486,
Snom 200, Zyxel P2000W


With regards
Roar

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