[cisco-voip] CUBE Config Dial Peers

[Anthony Holloway]

Brian,

Nice and clean, I like it!  Very similar to what I do.  I'd like to
comment/question yours a bit.

1. While I like that you can give a uri profile a name like ISP, I much
prefer to stick with numbers, since for most things, you must use numbers
when naming, so this keeps it consistent.
2. I don't specify the port in my server groups, since 5060 is default, but
I can see how it might help be more explicit for some people
3. Speaking of being explicit though, if that is your intention, I would
recommend pref 1 and pref 2, instead of implied pref 0 and pref 1
4. Why didn't you should your translation profiles and rules too?
5. I don't specify UDP as the transport, since it's the default, but again,
being explicit doesn't hurt anything
6. I like the extra dtmf on there.  too many configs are limited to rtp-nte
only and mtps are being invoked for every call to UCCX as one example
7. Why do you drop your fax rate down from 14400 to 9600 as a standard?  I
might learn something here, as faxing is not my strongest area.
8. Since you're doing DPGs, you don't need the destination-pattern .T
command on the outbound DPs.
9a. Why are you not doing sip options ping?  I would, and in which
case you need
a voice class sip options-keepalive profile
<https://community.cisco.com/t5/telepresence-and-video/sip-options-ping-and-session-server-group-on-dial-peer/td-p/2994584>
since you're using server groups.
9b. Also, if you do end up turning on options, you do in fact need a
destination-pattern command, and in which case, since it's not being used
for call routing, I just use like ABC123 as the pattern to ensure it never
can be used, but also, mildly clear it's not supposed to be used

I'll post a config as well, in a bit, and please feel free to
comment/question mine.




On Fri, Jun 12, 2020 at 3:20 PM Brian Meade <bmeade90 at vt.edu> wrote:

> I've been trying to make a standardized CUBE configuration using a lot of
> the newer features like dial-peer groups.
>
> 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.
>
> This gives you 4 total dial-peers to match any number.
>
> 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.
>
> voice class uri ISP sip
>  host ipv4:8.8.8.8
>
> voice class uri CUCM sip
>  host ipv4:192.168.100.100
>  host ipv4:192.168.100.200
>
> voice class server-group 100
>  ipv4 8.8.8.8 port 5060
>
> voice class server-group 200
>  ipv4 192.168.100.100 port 5060
>  ipv4 192.168.100.200 port 5060 preference 1
>
> voice class dpg 100
>
> voice class dpg 200
>
> dial-peer voice 100 voip
>  description Incoming Dial-peer from ISP
>  translation-profile incoming ISPInbound
>  session protocol sipv2
>  session transport udp
>  destination dpg 200
>  incoming uri via ISP
>  voice-class codec 1
>  dtmf-relay rtp-nte sip-kpml
>  fax-relay ecm disable
>  fax rate 9600
>
> dial-peer voice 200 voip
>  description Incoming Dial-peer from CUCM
>  session protocol sipv2
>  destination dpg 100
>  incoming uri via CUCM
>  voice-class codec 1
>  dtmf-relay rtp-nte sip-kpml
>  fax-relay ecm disable
>  fax rate 9600
>
> dial-peer voice 300 voip
>  description Outbound to ISP
>  translation-profile outgoing ISPOutbound
>  destination-pattern .T
>  session protocol sipv2
>  session transport udp
>  session server-group 100
>  voice-class codec 1
>  dtmf-relay rtp-nte sip-kpml
>  fax-relay ecm disable
>  fax rate 9600
>
> dial-peer voice 400 voip
>  description Outbound to CUCM
>  destination-pattern .T
>  session protocol sipv2
>  session server-group 200
>  voice-class codec 1
>  dtmf-relay rtp-nte sip-kpml
>  fax-relay ecm disable
>  fax rate 9600
>
> voice class dpg 100
>  dial-peer 300
>
> voice class dpg 200
>  dial-peer 400
>
> On Fri, Jun 12, 2020 at 3:12 PM JASON BURWELL via cisco-voip <
> cisco-voip at puck.nether.net> wrote:
>
>> 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
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
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