asterisk users - Mar 2006 - Ringduration problem when calling out via Sip

Hello,

when I try to call someone via Sip, the called phone just rings about 25
seconds.

Here's my Outgoing-Context:
<snip>
exten => _X.,1,Dial(SIP/XXX${EXTEN}@freenet,120)
exten => s,1,Answer()
exten => s,2,Playback(invalid)
exten => s,3,Hangup()
exten => h,1,Hangup()
</snip>

And here's a log that shows the problem. (I call from 11 to 12 via SIP. 12
is also a number for which my asterisk is responsible.)

<snip>
    -- Accepting overlap voice call from '11' to '12' on channel
0/1, span 1
    -- Starting simple switch on 'Zap/1-1'
    -- Executing Dial("Zap/1-1", "SIP/XXX12@freenet|120") in
new stack
    -- Called XXX12@freenet
    -- Accepting voice call from '' to '12' on channel 0/1, span
2
    -- Executing Dial("Zap/4-1", "Zap/g1/12|120|rt") in new
stack
    -- Requested transfer capability: 0x00 - SPEECH
    -- Called g1/12
    -- Zap/2-1 is proceeding passing it to Zap/4-1
    -- SIP/freenet-56c5 is making progress passing it to Zap/1-1
    -- Zap/2-1 is ringing
    -- Channel 0/1, span 1 got hangup request
  == Spawn extension (extern, 12, 1) exited non-zero on 'Zap/1-1'
    -- Executing Hangup("Zap/1-1", "") in new stack
  == Spawn extension (extern, h, 1) exited non-zero on 'Zap/1-1'
    -- Hungup 'Zap/1-1'
    -- Channel 0/1, span 2 got hangup, cause 16
    -- Hungup 'Zap/2-1'
  == Spawn extension (default, 12, 1) exited non-zero on 'Zap/4-1'
    -- Executing Hangup("Zap/4-1", "") in new stack
  == Spawn extension (default, h, 1) exited non-zero on 'Zap/4-1'
    -- Hungup 'Zap/4-1'
</snip>

I'm using Asterisk Version 1.2.4-BRIstuffed-0.3.0-PRE-1k with spandsp
0.0.2pre25.

When I call 12 via my outgoing zap-interface the called phone rings about 2
minutes. It makes no difference if I call a number somewhere else or on my
system.

When the call is established within these 25 seconds everything works as is
should. 

Can someone please give me a hint?

Philipp