<html><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; ">Maybe try "incoming called-number 65T" under dial peer 20 with your translation-rule. May the incoming call is hitting a different dial peer.<div><br></div><div>Try looking at "debug voip dialpeer" to see what is happening with the call.</div><div><br></div><div>Kevin<br><div><div>On Feb 18, 2010, at 3:38 PM, Kin Wai wrote:</div><br class="Apple-interchange-newline"><blockquote type="cite"><span class="Apple-style-span" style="border-collapse: separate; color: rgb(0, 0, 0); font-family: Helvetica; font-size: medium; font-style: normal; font-variant: normal; font-weight: normal; letter-spacing: normal; line-height: normal; orphans: 2; text-align: auto; text-indent: 0px; text-transform: none; white-space: normal; widows: 2; word-spacing: 0px; -webkit-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: 0px; -webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0px; "><div lang="EN-US" link="blue" vlink="purple"><div class="Section1"><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 11pt; font-family: Calibri, sans-serif; ">Hi,<o:p></o:p></div><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 11pt; font-family: Calibri, sans-serif; ">I’m have a gateway registered to Gatekeeper.<o:p></o:p></div><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 11pt; font-family: Calibri, sans-serif; "><o:p> </o:p></div><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 11pt; font-family: Calibri, sans-serif; ">Currently,<o:p></o:p></div><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 11pt; font-family: Calibri, sans-serif; ">Local call to remote sites through the GK --- OK<o:p></o:p></div><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 11pt; font-family: Calibri, sans-serif; ">Remote sites to PSTN through local gateway ---- OK (using the tech-prefix)<o:p></o:p></div><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 11pt; font-family: Calibri, sans-serif; ">Remote sites to Local Phone ----- NOT OK (using the same tech-prefix)<br><br><o:p></o:p></div><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 11pt; font-family: Calibri, sans-serif; ">I have the following configuration :<o:p></o:p></div><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 11pt; font-family: Calibri, sans-serif; ">dial-peer voice 20 voip<o:p></o:p></div><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 11pt; font-family: Calibri, sans-serif; "> destination-pattern 6.T<o:p></o:p></div><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 11pt; font-family: Calibri, sans-serif; "> session target ras<o:p></o:p></div><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 11pt; font-family: Calibri, sans-serif; ">!<o:p></o:p></div><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 11pt; font-family: Calibri, sans-serif; ">dial-peer voice 21 pots<o:p></o:p></div><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 11pt; font-family: Calibri, sans-serif; "> destination-pattern 65.T<o:p></o:p></div><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 11pt; font-family: Calibri, sans-serif; "> port 0/1/0:15<o:p></o:p></div><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 11pt; font-family: Calibri, sans-serif; "><o:p> </o:p></div><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 11pt; font-family: Calibri, sans-serif; ">*<b>I have excluded the h323-gateway commands</b>*<o:p></o:p></div><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 11pt; font-family: Calibri, sans-serif; ">The tech prefix for this site is 65<o:p></o:p></div><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 11pt; font-family: Calibri, sans-serif; "><o:p> </o:p></div><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 11pt; font-family: Calibri, sans-serif; ">What do I need to do in order for the remote site to call the local phones? The gatekeeper is a 3<sup>rd</sup><span class="Apple-converted-space"> </span>party gatekeeper.<o:p></o:p></div><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 11pt; font-family: Calibri, sans-serif; ">I suppose there’s no issue with the communications as remote sites user can make outgoing call to PSTN. <o:p></o:p></div><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 11pt; font-family: Calibri, sans-serif; "><o:p> </o:p></div><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 11pt; font-family: Calibri, sans-serif; ">Do I need to enable “allow connections h323 to h323” under voice service voip ? Or I need to create an additional dial-peer to cater for the local phones?<o:p></o:p></div><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 11pt; font-family: Calibri, sans-serif; "><o:p> </o:p></div><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 11pt; font-family: Calibri, sans-serif; ">I tried to apply translation-rule to strip away the tech-prefix or even make it the full DID number under dial-peer 20, but it still won’t work.<o:p></o:p></div><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 11pt; font-family: Calibri, sans-serif; "><o:p> </o:p></div><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 11pt; font-family: Calibri, sans-serif; ">Anyone have any clue about this problem?<o:p></o:p></div><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 11pt; font-family: Calibri, sans-serif; "><o:p> </o:p></div><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 11pt; font-family: Calibri, sans-serif; ">Thanks in advance!<o:p></o:p></div><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 11pt; font-family: Calibri, sans-serif; "><o:p> </o:p></div><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 11pt; font-family: Calibri, sans-serif; ">Regards,<o:p></o:p></div><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 11pt; font-family: Calibri, sans-serif; ">Kin Wai<o:p></o:p></div></div>_______________________________________________<br>cisco-voip mailing list<br><a href="mailto:cisco-voip@puck.nether.net" style="color: blue; text-decoration: underline; ">cisco-voip@puck.nether.net</a><br><a href="https://puck.nether.net/mailman/listinfo/cisco-voip" style="color: blue; text-decoration: underline; ">https://puck.nether.net/mailman/listinfo/cisco-voip</a><br></div></span></blockquote></div><br></div></body></html>