<html><head><style type='text/css'>p { margin: 0; }</style></head><body><div style='font-family: Verdana; font-size: 10pt; color: #000000'><style>p { margin: 0; }</style><link href="/zimbra/css/msgview.css?v=081117022631" rel="stylesheet"><div style="font-family: Verdana; font-size: 10pt; color: rgb(0, 0, 0);">Every incoming call on my CME hasn't got the E164 format. The standard + isn't shown.<br>Instead of +31201234567 I get 31201234567.<br><br>Looking at my translation rules and dialpeers I don't see the issue. <br>Can somebody help/advise?<br><br>####<br>voice translation-rule 11<br> rule 1 /.*/ /1099/<br>!<br>voice translation-rule 12<br> rule 1 /.*/ /1099/<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 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>!<br>voice translation-profile SIP_Outgoing_Local<br> translate calling 22<br> translate called 23<br><br>###################<br><br>!<br>dial-peer voice 1 voip<br> description *** incoming ***<br> translation-profile incoming SIP_Incoming1<br> max-conn 5<br> voice-class codec 1<br> voice-class sip dtmf-relay force rtp-nte<br> session protocol sipv2<br> session target sip-server<br> incoming uri to INCOMINGNR<br> dtmf-relay rtp-nte<br> no call fallback<br> no vad<br>!<br>dial-peer voice 2 voip<br> description *** incoming ***<br> translation-profile incoming SIP_Incoming2<br> max-conn 5<br> voice-class codec 1<br> voice-class sip dtmf-relay force rtp-nte<br> session protocol sipv2<br> session target sip-server<br> incoming uri to INCOMINGNR2<br> dtmf-relay rtp-nte<br> no call fallback<br> no vad<br>!<br>dial-peer voice 11 voip<br>
description *** outgoing ***<br>
translation-profile outgoing SIP_Outgoing<br>
max-conn 10<br>
destination-pattern 0T<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>dial-peer voice 12 voip<br> description *** outgoing ***<br> translation-profile outgoing SIP_Outgoing_Local<br> max-conn 5<br> destination-pattern 0252T<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></div></div></body></html>