I had CUBE and did it there.<div><br></div><div><div>voice translation-rule 1</div><div> rule 1 /^24\(..\)$/ /80355524\1/ --> 24xx is extension number changing to 80355524xx</div><div><br></div><div><div>voice translation-profile Redirect-number </div>
<div> translate redirect-target 1 </div><div> translate redirect-called 1</div></div><div><br></div><div><div>dial-peer voice 1000 voip --> you would need to add to all of your outbound dial peers</div>
<div> translation-profile incoming Redirect-number</div><div> destination-pattern 1[2-9]..[2-9]......</div><div> voice-class codec 1</div><div> no voice-class sip localhost </div><div> no voice-class sip outbound-proxy </div>
<div> voice-class sip early-offer forced</div><div> session protocol sipv2</div><div> session target sip-server</div><div> incoming called-number ....</div><div> dtmf-relay rtp-nte</div><div> ip qos dscp cs3 signaling</div>
</div><div><br></div><div>Good debugs to see what is going on on the gw</div><div><br></div><div><div>debug voip ccapi inout </div><div>debug ccsip call</div><div>debug ccsip events</div><div>debug ccsip error</div><div>debug ccsip messag</div>
</div><div><br></div><div>Here's what I saw when couldn't CF to PSTN</div><div><br></div><div><div>cc_api_call_setup_ind_common:</div><div> cisco-username=8038070788</div><div> ----- ccCallInfo IE subfields -----</div>
<div> cisco-ani=<a href="mailto:sip%3A8035550788@192.168.25.21">sip:8035550788@192.168.25.21</a></div><div> cisco-anitype=0</div><div> cisco-aniplan=0</div><div> cisco-anipi=0</div><div> cisco-anisi=0</div><div>
dest=<a href="http://sip:5554780@192.168.25.30:5060">sip:5554780@192.168.25.30:5060</a></div><div> cisco-desttype=0</div><div> cisco-destplan=0</div><div> cisco-rdie=74</div><div> cisco-rdn=2455 --> extension # carrier is seeing, when they need to see 10-digit</div>
<div> cisco-rdntype=0</div><div> cisco-rdnplan=0</div><div> cisco-rdnpi=0</div><div> cisco-rdnsi=1</div><div> cisco-redirectreason=4 fwd_final_type =0</div><div> final_redirectNumber =</div><div> hunt_group_timeout =0</div>
</div><div><br></div><br><div class="gmail_quote">On Fri, Feb 4, 2011 at 11:05 AM, Bill Riley <span dir="ltr"><<a href="mailto:bill@hitechconnection.net">bill@hitechconnection.net</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
<div lang="EN-US" link="blue" vlink="purple"><div><p class="MsoNormal"> </p><p class="MsoNormal">I have a SIP trunk from Quest. They require the outbound caller-ID to be a 10-digit number assigned to that SIP trunk. When a user places an outbound call I have the external phone number mask set to their 10 digit DID. When the user sets their phone to CFA to a cell phone the call fails. We set the Calling party selection to first redirected number and check Redirecting Diversion Header Deliver on the CUCM SIP trunk. Now the gateway sees the four digit extension of the original caller. I would like to send the original caller 10-digit number as the caller-id but if I can’t do that how can I send the IP phones 10 digit number as the caller-id so the call completes correctly. </p>
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