<html><head><style type='text/css'>p { margin: 0; }</style></head><body><div style='font-family: Verdana; font-size: 10pt; color: #000000'>Thanks,<br>I created rule 24 and attached it to dial-peer voice 11 voip.<br><br><br>!<br>voice translation-rule 11<br> rule 1 /.*/ /1000/<br>!<br>voice translation-rule 12<br> rule 1 /.*/ /1010/<br>!<br>voice translation-rule 21<br> rule 1 /^0/ //<br>!<br>voice translation-rule 22<br> rule 1 /.*/ /31252763010/<br>!<br>voice translation-rule 23<br> rule 1 /^0/ /252/<br>!<br>voice translation-rule 24<br> rule 1 /^\+31/ /0/<br>!<br>voice translation-rule 410<br> rule 1 /^0\(.*\)/ /\1/<br> rule 2 /^....$/ /31252763010/<br>!<br>!<br>voice translation-profile CUE_Voicemail<br> translate called 1<br>!<br>voice translation-profile SIP_CallForwarding<br> translate redirect-target 410<br> translate redirect-called 410<br>!<br>voice translation-profile SIP_Incoming1<br> translate called 11<br>!<br>voice translation-profile SIP_Incoming2<br> translate called 12<br>!<br>voice translation-profile SIP_Outgoing<br>translate calling 22<br> translate called 21<br> translate called 24<br>!<br>voice translation-profile SIP_Outgoing_Local<br> translate calling 22<br> translate called 23<br>!<br><br>!<br>dial-peer voice 11 voip <br> description *** outgoing on winitu communications ***<br> translation-profile outgoing SIP_Outgoing<br> max-conn 5<br> destination-pattern 00T<br> voice-class codec 1<br> voice-class sip dtmf-relay force rtp-nte<br> session protocol sipv2<br> session target sip-server<br> dtmf-relay rtp-nte<br> no vad<br> !<br><br><br>----- Oorspronkelijk bericht -----<br>Van: "Mike Thompson" <mthompson729@gmail.com><br>Aan: "Marty van de Veerdonk [Voice Identity]" <marty@voiceidentity.nl>, "cisco-voip" <cisco-voip@puck.nether.net><br>Verzonden: Dinsdag 5 januari 2010 18:42:15 GMT +01:00 Amsterdam / Berlijn / Bern / Rome / Stockholm / Wenen<br>Onderwerp: RE: [cisco-voip] CME translation rule for +31 behavior?<br><br>
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<p class="MsoNormal"><span style="font-size: 11pt; font-family: "Calibri","sans-serif"; color: rgb(31, 73, 125);">Marty,</span></p>
<p class="MsoNormal"><span style="font-size: 11pt; font-family: "Calibri","sans-serif"; color: rgb(31, 73, 125);"> Yes, the way to handle this is to create a translation
akin to /^\+31/ /0/ and it should swap the prefix +31 and replace it with a 0.
Syntax may be a little off…worked an overnight on a customer cut and I’m a
shade on the sleepy side.</span></p>
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<p class="MsoNormal"><b><span style="font-size: 10pt; font-family: "Tahoma","sans-serif";">From:</span></b><span style="font-size: 10pt; font-family: "Tahoma","sans-serif";">
cisco-voip-bounces@puck.nether.net [mailto:cisco-voip-bounces@puck.nether.net] <b>On
Behalf Of </b>Marty van de Veerdonk [Voice Identity]<br>
<b>Sent:</b> Tuesday, January 05, 2010 10:01 AM<br>
<b>To:</b> cisco-voip<br>
<b>Subject:</b> [cisco-voip] CME translation rule for +31 behavior?</span></p>
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<p class="MsoNormal"><span style="font-size: 10pt; font-family: "Verdana","sans-serif"; color: black;">After an incoming call, the number of the user is shown in the
display of the Cisco phone in a +31xxxxxxxxx format.<br>
Therefor it's not possible to call the person back because CME doesn't
understand this format (or aspects a 0 for an outside line).<br>
<br>
Is it possible to fix this with a (outbound) translation rule?<br>
<br>
reagrds, Marty</span></p>
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