asterisk users - Jan 2010 - Inserting a wait in a sip dial

Hi All,

After searching and didnt found it, im just sending my situation here,
maybe someone knows where i should look.
Im using Asterisk 1.6.1.10

Internally the user with a sip phone dials a number for instance 0623456789
It goes fine to the specific dial rule:
which is: exten => _0[6].,2,Dial(SIP/0${EXTEN:1}#@xs4all-out,60,tTwWkK)

This works fine without a charm, but the situation is that i want to hide
the phonenumber going out, this is done in the netherlands by dialling
*31# and then the phonenumber you want to call.
so i modified it to:
exten => _0[6].,2,Dial(SIP/*31#0${EXTEN:1}@xs4all-out,60,tTwWkK)

Only then it doesnt work, since i prolly need to wait before dialling the
number.

so after searching i saw several posts and sites which stated that i need
to use 'w' in the dial command.

So i changed it to:
exten => _0[6].,2,Dial(SIP/*31#w0${EXTEN:1}@xs4all-out,60,tTwWkK)

But then the other peer says:

    -- Called *31#w06123456789 at xs4all-out
    -- SIP/xs4all-out-00000234 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
    -- Auto fallthrough, channel 'SIP/evert-00000233' status is
'CONGESTION'


Anyone an idea where i should look, or how i should change it, so that i
do get a wait before sending the rest of the number to the sip peer.

Thanks in advance,

Regards,

Evert

<snip>
But then the other peer says:

    -- Called *31#w06123456789 at xs4all-out
    -- SIP/xs4all-out-00000234 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
    -- Auto fallthrough, channel 'SIP/evert-00000233' status is
'CONGESTION'


Anyone an idea where i should look, or how i should change it, so that i
do get a wait before sending the rest of the number to the sip peer.
</snip>

I don't have an answer for this but am holding my breath that *someone*
does. I ran into a similar situation (dial a number, then wait, then dial an
extension via SIP to PSTN) a few weeks ago and never figured out a resolution...

My THOUGHT is that you would have to manually inject the DTMF into the stream
somehow after the SIP provider connects the call...

Thanks
Dave