Hi Adrian,

Thanks for your reply. Unfortunately, changing the Asterisk side is not an
option as some PBX'es are not under my control and others are embedded
devices and such. So I'm looking for a solution on the OpenSIPS side...

Thanks,
Remco.

On Tue, Dec 6, 2011 at 2:04 AM, Adrian Serafini <[email protected]>wrote:

>
>  Could someone point me in the right direction?
>>
>
> It can be done from asterisk, server side would be cleaner.  This is from
> the top of the sample extensions.conf.
>
> [from-ser1]
> exten => _.,1,Goto(from-pstn,${CUT(CUT(**SIP_HEADER(To),@,1),:,2)},1)
>
>
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