I would look at modifying the diversion header in the gateway if you can and your provider supports authentication on the diversion header. This way the calling ID of the forwarded call will be retained. The other method will overwrite the calling ID.<div>
<br></div><div><div><br></div><div>voice class sip-profiles 1</div></div><div>request INVITE sip-header Diversion modify “<sip:(.*)@(.*)>” “<<a href="mailto:sip%3A722555000@5.5.5.5">sip:722555000@5.5.5.5</a>>”</div>
<div><br></div><div>!replace with number/ip</div><div><br></div><div>dial-peer voice XXX voip</div><div><div>dial-peer voice 100 voip</div><div> description OUTBOUND to VzB</div><div> translation-profile outgoing DIGITSTRIP-9</div>
<div> preference 1</div><div> destination-pattern 9T</div><div><div>session protocol sipv2</div><div> session target sip-server</div><div> voice-class codec 1 offer-all</div></div><div>voice-class sip profiles 1</div><div>
<div> dtmf-relay rtp-nte</div><div> no vad</div></div><div><br></div><div><br clear="all">Best Regards,<br><br>Mike Lydick<br><br><br>
<br><br><div class="gmail_quote">On Tue, May 31, 2011 at 8:13 AM, Rynard Coetzee <span dir="ltr"><<a href="mailto:Rynard.Coetzee@bytes.co.za">Rynard.Coetzee@bytes.co.za</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
<div>
<div style="direction:ltr;font-family:Tahoma;color:#000000;font-size:10pt">
<p>Hi There</p>
<p>I have an issue with forwarding of calls from a DN on CUCM to a PSTN number that is routed across a SIP trunk ,i can call the PSTN number directly but when i forward the DN to the number the call fails when dialled from a different phone. If i dial only
the extension of the DN the call proceeds though ,it only happens when dialling the DID number.</p>
<p>Any ideas ?</p>
<p> </p>
</div>
</div>
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<br></blockquote></div><br></div></div>