<p dir="ltr">Hi,</p>
<p dir="ltr">Think you need to specify a preference other than 0 under hunt group for this to work.</p>
<p dir="ltr">Kind regards, </p>
<p dir="ltr">Roshan</p>
<div class="gmail_quote">On 20 Jan 2015 20:31, "Kapil Jain" <<a href="mailto:kapiljain007@hotmail.com">kapiljain007@hotmail.com</a>> wrote:<br type="attribution"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div><div dir="ltr"><span style="color:rgb(68,68,68);font-size:15.1999998092651px;line-height:21.2999992370605px;background-color:rgb(255,255,255)">I am having issue configuring SRST with SIP ITSP.</span><div style="line-height:21.2999992370605px;color:rgb(68,68,68);font-size:15.1999998092651px;background-color:rgb(255,255,255)"><br style="line-height:21.2999992370605px"></div><div style="line-height:21.2999992370605px;color:rgb(68,68,68);font-size:15.1999998092651px;background-color:rgb(255,255,255)">I am using CME as SRST, huntgroup is taking preference over an Existing dial-peer going to CUCM in non SRST mode. Configuration looks like below</div><div style="line-height:21.2999992370605px;color:rgb(68,68,68);font-size:15.1999998092651px;background-color:rgb(255,255,255)"><br style="line-height:21.2999992370605px"></div><div style="line-height:21.2999992370605px;color:rgb(68,68,68);font-size:15.1999998092651px;background-color:rgb(255,255,255)">ITSP>>>GW>>>CUCM>>>Hunt-Pilot 5000>>List of DN's</div><div style="line-height:21.2999992370605px;color:rgb(68,68,68);font-size:15.1999998092651px;background-color:rgb(255,255,255)"><br style="line-height:21.2999992370605px"></div><div style="line-height:21.2999992370605px;color:rgb(68,68,68);font-size:15.1999998092651px;background-color:rgb(255,255,255)">Same call flow is required during SRST.</div><div style="line-height:21.2999992370605px;color:rgb(68,68,68);font-size:15.1999998092651px;background-color:rgb(255,255,255)"><br style="line-height:21.2999992370605px"></div><div style="line-height:21.2999992370605px;color:rgb(68,68,68);font-size:15.1999998092651px;background-color:rgb(255,255,255)">Below is the config in GW</div><div style="line-height:21.2999992370605px;color:rgb(68,68,68);font-size:15.1999998092651px;background-color:rgb(255,255,255)"><br></div><div style="line-height:21.2999992370605px;color:rgb(68,68,68);font-size:15.1999998092651px;background-color:rgb(255,255,255)"><span style="font-size:15.1999998092651px;line-height:21.2999992370605px"><u>ITSP to GW to CUCM :-</u></span></div><div style="line-height:21.2999992370605px;color:rgb(68,68,68);font-size:15.1999998092651px;background-color:rgb(255,255,255)"><div style="line-height:21.2999992370605px;font-size:15.1999998092651px"><br></div><div style="line-height:21.2999992370605px;font-size:15.1999998092651px"><div style="line-height:21.2999992370605px;font-size:15.1999998092651px">dial-peer voice 1 voip</div><div style="line-height:21.2999992370605px;font-size:15.1999998092651px"> description "Incoming calls from ITSP to Pilot# "</div><div style="line-height:21.2999992370605px;font-size:15.1999998092651px"> session protocol sipv2</div><div style="line-height:21.2999992370605px;font-size:15.1999998092651px"> session target sip-server</div><div style="line-height:21.2999992370605px;font-size:15.1999998092651px"> incoming called-number 2223335000</div><div style="line-height:21.2999992370605px;font-size:15.1999998092651px"> voice-class codec 1</div><div style="line-height:21.2999992370605px;font-size:15.1999998092651px"> dtmf-relay rtp-nte</div><div style="line-height:21.2999992370605px;font-size:15.1999998092651px"><br></div><div style="line-height:21.2999992370605px;font-size:15.1999998092651px">dial-peer voice 30 voip</div><div style="line-height:21.2999992370605px;font-size:15.1999998092651px"> description "calls to CUCM"</div><div style="line-height:21.2999992370605px;font-size:15.1999998092651px"> destination-pattern <span style="font-size:15.1999998092651px;line-height:21.2999992370605px">2223335000</span></div><div style="line-height:21.2999992370605px;font-size:15.1999998092651px"> session protocol sipv2</div><div style="line-height:21.2999992370605px;font-size:15.1999998092651px"> session target ipv4:10.10.8.50</div><div style="line-height:21.2999992370605px;font-size:15.1999998092651px"> voice-class codec 1</div><div style="line-height:21.2999992370605px;font-size:15.1999998092651px"> dtmf-relay rtp-nte</div><div style="line-height:21.2999992370605px;font-size:15.1999998092651px"><br></div><div><div style="line-height:21.2999992370605px;font-size:15.1999998092651px"><u>Hunt Group in SRST :-</u></div><div style="line-height:21.2999992370605px;font-size:15.1999998092651px"><br></div><div style="line-height:21.2999992370605px;font-size:15.1999998092651px"><div style="font-size:15.1999998092651px;line-height:21.2999992370605px">ephone-hunt 1 sequential</div><div style="font-size:15.1999998092651px;line-height:21.2999992370605px"> pilot <span style="font-size:15.1999998092651px;line-height:21.2999992370605px">2223335000</span></div><div style="font-size:15.1999998092651px;line-height:21.2999992370605px"> list 5001, 5002</div><div style="font-size:15.1999998092651px;line-height:21.2999992370605px"> final 5007</div><div style="font-size:15.1999998092651px;line-height:21.2999992370605px"> timeout 20, 20</div><div><br></div><div>As soon as I create ephone hunt , my calls fails due to dial-peer created by hunt group is taking preference. </div><div><br></div><div>Thxs!!</div><div>Kapil</div></div></div></div><div><br></div></div><div style="line-height:21.2999992370605px;color:rgb(68,68,68);font-size:15.1999998092651px;background-color:rgb(255,255,255)"><br></div><div style="line-height:21.2999992370605px;color:rgb(68,68,68);font-size:15.1999998092651px;background-color:rgb(255,255,255)"><br></div> </div></div>
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