<div>Im sorry about that yes I am using CUBE with the configuration below</div>
<div> </div>
<div>I have also attached my capture.</div>
<div> </div>
<div><br>voice service voip<br> allow-connections h323 to h323<br> allow-connections h323 to sip<br> allow-connections sip to h323<br> allow-connections sip to sip<br> fax protocol pass-through g711ulaw<br> h323<br> emptycapability<br>
h225 id-passthru<br> h245 passthru tcsnonstd-passthru<br> sip<br>!<br>!<br>!<br>voice class codec 1<br> codec preference 2 g711ulaw<br>!<br>!<br>!<br>!<br>voice class h323 50<br> h225 timeout tcp establish 3<br>!<br>!<br>
!<br>!<br>!<br>!<br>!<br>!<br>!<br>!<br>!<br>voice translation-rule 1<br> rule 1 /.*/ /190/<br>!<br>voice translation-rule 2<br> rule 1 /.*/ /1&/<br>!<br>!<br>voice translation-profile aa<br> translate called 1<br>!<br>
voice translation-profile addone<br> translate called 2<br>!<br><br>!<br>!<br>dial-peer voice 100 voip<br> description AA Publisher<br> preference 1<br> destination-pattern 1..<br> voice-class h323 50<br> session target ipv4:10.1.80.10<br>
dtmf-relay h245-alphanumeric<br> codec g711ulaw<br> no vad<br>!<br>dial-peer voice 1000 voip<br> description incoming Call<br> translation-profile incoming aa<br> preference 1<br> rtp payload-type nse 101<br> rtp payload-type nte 100<br>
incoming called-number 6782282221<br> dtmf-relay rtp-nte<br> codec g711ulaw<br> no vad<br>!<br>dial-peer voice 101 voip<br> description AA Subscriber<br> preference 2<br> destination-pattern 1..<br> voice-class h323 50<br>
session target ipv4:10.1.80.11<br> dtmf-relay h245-alphanumeric<br> codec g711ulaw<br> no vad<br>!<br>dial-peer voice 2000 voip<br> description outbound<br> translation-profile outgoing addone<br> preference 1<br> destination-pattern .T<br>
progress_ind setup enable 3<br> progress_ind progress enable 8<br> rtp payload-type nse 101<br> rtp payload-type nte 100<br> voice-class codec 1<br> session protocol sipv2<br> session target dns:<a href="http://sipconnect.ipcomms.net">sipconnect.ipcomms.net</a><br>
dtmf-relay rtp-nte<br> no vad<br>!<br>dial-peer voice 150 voip<br> description incoming outbound<br> preference 1<br> voice-class h323 50<br> incoming called-number .T<br> dtmf-relay h245-alphanumeric<br> codec g711ulaw<br>
no vad<br>!<br>!<br>sip-ua<br> credentials username ********* password 7 ********* realm <a href="http://sipconnect.ipcomms.net">sipconnect.ipcomms.net</a><br> authentication username********* password 7 1********* <br>
authentication username ********* password 7 ********* realm <a href="http://sipconnect.ipcomms.net">sipconnect.ipcomms.net</a><br> registrar dns:<a href="http://sipconnect.ipcomms.net">sipconnect.ipcomms.net</a> expires 60<br>
sip-server dns:<a href="http://sipconnect.ipcomms.net:5060">sipconnect.ipcomms.net:5060</a><br>!<br>!<br></div>
<div class="gmail_quote">On Fri, Dec 4, 2009 at 11:24 AM, Ryan Ratliff <span dir="ltr"><<a href="mailto:rratliff@cisco.com">rratliff@cisco.com</a>></span> wrote:<br>
<blockquote style="BORDER-LEFT: #ccc 1px solid; MARGIN: 0px 0px 0px 0.8ex; PADDING-LEFT: 1ex" class="gmail_quote">
<div style="WORD-WRAP: break-word">The gateway screenshot appears to be an H.323 gateway, not a SIP trunk. Additionally, you have only the inbound call routing info included, not outbound.
<div><br></div>
<div>If you are using CUBE to go from H.323 to SIP that's rather important and something you will want to mention.</div>
<div><br></div>
<div>In order to use the external phone number mask you need to tick that check box on the route pattern.</div>
<div><br>
<div><span style="TEXT-TRANSFORM: none; TEXT-INDENT: 0px; BORDER-COLLAPSE: separate; FONT: medium Helvetica; WHITE-SPACE: normal; LETTER-SPACING: normal; COLOR: rgb(0,0,0); WORD-SPACING: 0px">
<div>-Ryan</div></span></div><br>
<div>
<div>
<div></div>
<div class="h5">
<div>On Dec 4, 2009, at 11:04 AM, Dane Newman wrote:</div><br>
<div>Hello,</div>
<div> </div>
<div>I am having an odd issue and I am wondering why it is happening. I have a sip trunk to the provider. I call out a phone and it passes the CLID of my internal extention rather then the clid im setting on my gateway under prefix dn and External Phone Number Mask.</div>
<div> </div>
<div>I took a packet capture and in the sip invite message its showing SIP >From <a href="mailto:101@sipconnect.ipcomms.net" target="_blank">101@sipconnect.ipcomms.net</a> which 101 is my internal extention.</div>
<div> </div>
<div>Is anyone able to comment?</div>
<div> </div>
<div>Dane</div></div></div><span><dn.jpg></span><span><ext.jpg></span>_______________________________________________<br>cisco-voip mailing list<br><a href="mailto:cisco-voip@puck.nether.net" target="_blank">cisco-voip@puck.nether.net</a><br>
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