Hi Adrian,<br><br>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...<br>
<br>Thanks,<br>Remco.<br><br><div class="gmail_quote">On Tue, Dec 6, 2011 at 2:04 AM, Adrian Serafini <span dir="ltr"><<a href="mailto:wealwildwon@wombit.com">wealwildwon@wombit.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
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<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Could someone point me in the right direction?<br>
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It can be done from asterisk, server side would be cleaner. This is from the top of the sample extensions.conf.<br>
<br>
[from-ser1]<br>
exten => _.,1,Goto(from-pstn,${CUT(CUT(<u></u>SIP_HEADER(To),@,1),:,2)},1)<br>
<br>
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