<html><body><div style="color:#000; background-color:#fff; font-family:times new roman, new york, times, serif;font-size:12pt"><div style="RIGHT: auto"><SPAN style="RIGHT: auto"><SPAN style="RIGHT: auto" class=tab>Having an issue where calls coming in from ITSP ring and never fwd to voicemail. Internally voicemail works. either by dialing the vm number or by pushing the messages button which essentially just speed dials the 4000 extension used for voicemail. Can someone take a look at the included config and maybe point out what i'm missing to allow for calls to be fwd'ed to Cue after the specified time.</SPAN></SPAN></div>
<div style="RIGHT: auto"><SPAN style="RIGHT: auto"><SPAN style="RIGHT: auto" class=tab></SPAN></SPAN> </div>
<div style="RIGHT: auto"><SPAN style="RIGHT: auto"><SPAN style="RIGHT: auto" class=tab>voice service voip<BR> ip address trusted list<BR> ipv4 0.0.0.0 0.0.0.0<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> supplementary-service h450.12<BR> sip<BR> registrar server expires max 600 min 60<BR>!<BR>!<BR>!<BR>!<BR>voice translation-rule 1<BR> rule 1 /^9/ //<BR>!<BR>voice translation-rule 3<BR> rule 1 /4.../ /5856786019/<BR>!<BR>!<BR>voice translation-profile voip.ms<BR> translate calling 3<BR> translate called 1<BR></div></SPAN></SPAN>
<div style="RIGHT: auto"><SPAN style="RIGHT: auto"><SPAN style="RIGHT: auto" class=tab>dial-peer voice 1 voip<BR> description **SIP Trunk to newyork.voip.ms**<BR> translation-profile outgoing voip.ms<BR> destination-pattern 9[2-9].[2-9].......<BR> session protocol sipv2<BR> session target dns:newyork.voip.ms<BR> dtmf-relay rtp-nte sip-notify<BR> codec g711ulaw<BR> no vad<BR>!<BR>dial-peer voice 2 voip<BR> description **Incoming SIP Trunk - Voip.ms**<BR> translation-profile incoming voip.ms<BR> session protocol sipv2<BR> session target ipv4:10.50.1.2<BR> incoming called-number 5856786019<BR> dtmf-relay sip-notify<BR> codec g711ulaw<BR> no vad<BR>!<BR>dial-peer voice 20 pots<BR> destination-pattern 5.T<BR> direct-inward-dial<BR> no sip-register<BR>!<BR>dial-peer voice 3 voip<BR> description ** Voicemail **<BR> destination-pattern 4000<BR> session
protocol sipv2<BR> session target ipv4:1.1.1.2<BR> dtmf-relay sip-notify<BR> codec g711ulaw<BR> no vad<BR> </SPAN><BR class=yui-cursor>ephone-dn-template 1<BR> call-forward busy 4000<BR> call-forward noan 4000 timeout 18</div></SPAN>
<DIV></DIV>
<DIV style="RIGHT: auto"> </DIV>
<DIV style="RIGHT: auto">ephone-dn 1 dual-line<BR> number 4005 secondary 5856786019 no-reg<BR> label 4005<BR> name Jay Stants<BR> ephone-dn-template 1<VAR id=yui-ie-cursor></VAR><BR></DIV>
<DIV style="RIGHT: auto"> </DIV>
<DIV style="RIGHT: auto"> </DIV>
<div style="RIGHT: auto"><SPAN style="COLOR: rgb(0,96,191); RIGHT: auto">Regards,</SPAN><BR style="COLOR: rgb(0,96,191)"><SPAN style="COLOR: rgb(0,96,191)">Jay Stants</SPAN><BR><A style="COLOR: rgb(0,64,127)" href="mailto:jaystants@rogers.com" rel=nofollow target=_blank>jaystants@rogers.com</A><BR></div></div></body></html>