<div dir="ltr">I've been trying to make a standardized CUBE configuration using a lot of the newer features like dial-peer groups.<div><br></div><div>This is what I have now. It's an inbound dial-peer for CUCM matching the CUCM IP's via Via header. Then an inbound dial-peer for the ISP. Then an outbound dial-peer to CUCM and an outbound dial-peer to the ISP. If you have more IP's for the ISP or CUCM, you can easily add them. destination-pattern .T is not used at all due to using dial-peer group matching. This doesn't account for bindings that must be done per dial-peer. It also doesn't show translation-profiles/rules. </div><div><br></div><div>This gives you 4 total dial-peers to match any number.</div><div><br></div><div>If it comes in from CUCM, it will route to the SIP carrier. If it comes in from the SIP carrier, it will route to CUCM.<br><br>voice class uri ISP sip<br> host ipv4:8.8.8.8<br><br>voice class uri CUCM sip<br> host ipv4:192.168.100.100<br> host ipv4:192.168.100.200<br><br>voice class server-group 100<br> ipv4 8.8.8.8 port 5060<br><br>voice class server-group 200<br> ipv4 192.168.100.100 port 5060<br> ipv4 192.168.100.200 port 5060 preference 1<br></div><div><br></div><div>voice class dpg 100<br><br>voice class dpg 200<br></div><div><br></div><div>dial-peer voice 100 voip<br> description Incoming Dial-peer from ISP<br> translation-profile incoming ISPInbound<br> session protocol sipv2<br> session transport udp<br> destination dpg 200<br> incoming uri via ISP<br> voice-class codec 1<br> dtmf-relay rtp-nte sip-kpml<br> fax-relay ecm disable<br> fax rate 9600<br><br>dial-peer voice 200 voip<br> description Incoming Dial-peer from CUCM<br> session protocol sipv2<br> destination dpg 100<br> incoming uri via CUCM<br> voice-class codec 1<br> dtmf-relay rtp-nte sip-kpml<br> fax-relay ecm disable<br> fax rate 9600<br></div><div><br></div><div>dial-peer voice 300 voip<br> description Outbound to ISP<br> translation-profile outgoing ISPOutbound<br> destination-pattern .T<br> session protocol sipv2<br> session transport udp<br> session server-group 100<br> voice-class codec 1<br> dtmf-relay rtp-nte sip-kpml<br> fax-relay ecm disable<br> fax rate 9600<br><br>dial-peer voice 400 voip<br> description Outbound to CUCM<br> destination-pattern .T<br> session protocol sipv2<br> session server-group 200<br> voice-class codec 1<br> dtmf-relay rtp-nte sip-kpml<br> fax-relay ecm disable<br> fax rate 9600<br></div><div><br></div><div>voice class dpg 100<br> dial-peer 300<br><br>voice class dpg 200<br> dial-peer 400<br></div></div><br><div class="gmail_quote"><div dir="ltr" class="gmail_attr">On Fri, Jun 12, 2020 at 3:12 PM JASON BURWELL via cisco-voip <<a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>> wrote:<br></div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">
<div lang="EN-US">
<div class="gmail-m_1169134371059202191WordSection1">
<p class="MsoNormal">Does anyone have a good, straightforward reference doc to configuring CUBE dial peers? I have what I would have thought should be a fairly basic config but I’m having trouble getting everything to work properly. I’ve had some assistance
but it seems like a whole lot of configuration to do what little I really need to do. Basically, I just need to send whatever comes from CUCM- either 10, 11 or 3 digits in the SIP invite for outbound and for inbound calls the provider sends me 10 digits in
the invite, I just need to strip off the first 6 and send the last 4 to CUCM to route. I have a lot of adding and stripping digits going on between CUCM and CUBE to make this work. Just trying to find reference docs to see if any of this can be cleaned up.
Thanks<u></u><u></u></p>
</div>
</div>
_______________________________________________<br>
cisco-voip mailing list<br>
<a href="mailto:cisco-voip@puck.nether.net" target="_blank">cisco-voip@puck.nether.net</a><br>
<a href="https://puck.nether.net/mailman/listinfo/cisco-voip" rel="noreferrer" target="_blank">https://puck.nether.net/mailman/listinfo/cisco-voip</a><br>
</blockquote></div>