[cisco-voip] CUBE Config Dial Peers
Anthony Holloway
avholloway+cisco-voip at gmail.com
Fri Jun 12 20:53:05 EDT 2020
All great points, thanks for taking the time to respond.
The only one I think that I need to reply to is the DPG and
destination-pattern one. I was actually troubleshooting a customer CUBE
wherein this exact scenario was in place and the only negative was getting
options to work. Otherwise, it was routing the call just fine. Granted, I
couldn't tell you what version that was, as it was like a year or so ago,
but either way, I always put a destination-pattern on because you need one
for options to function.
I guess I could reply to one more, and that has to do with tweaking retries
from 6 to 2 AND using options. Why stick to one, when you can do both?
Here's the one I use which I said was very similar to yours.
The first thing to note is the numeric structure of my tags.
1000 series numbers are the ITSP side
2000 series numbers are the CUCM side
I would expand this to 3000, 4000, etc., for additional integrations like
PRIs, FXO, second ITSP, second PBX, etc. Most I ever had was 6
integrations into a single CUBE i think.
The second digit is 1 for incoming and 2 for outgoing.
The 4rd and fourth digit are generally not used, unless I need additional
dial-peers for the same peer and direction, but doing something slightly
different. Most I ever used was like 15 i think. E.g., 2215 But that was
not using IP addresses in the matching and DPGs, that was using phone
number matching, and I was using steering codes.
This numbering structure allows me to do something like this:
show run | section 12..
Which would then show me the following all at once: URI, Server Group,
Profile and Dial Peers all pertaining to the outgoing ITSP leg.
Also, in this example, we pass +E164 through the gateway bi-directionally,
so no digit manip needed.
voice class uri 1100 sip
host ipv4:8.8.8.8
host ipv4:9.9.9.9
!
voice class server-group 1200
description ITSP Peers
ipv4 8.8.8.8 preference 1
ipv4 9.9.9.9 preference 2
!
voice class sip-options-keepalive 1200
description ITSP Peers (Intentionally Left Blank)
!
voice class uri 2100 sip
host ipv4:10.1.1.2
host ipv4:10.1.1.3
!
voice class server-group 2200
description CUCM Nodes
ipv4 10.1.1.2 preference 1
ipv4 10.1.1.3 preference 2
!
voice class sip-options-keepalive 2200
description CUCM Nodes (Intentionally Left Blank)
!
voice class dpg 1200
dial-peer 1200
!
voice class dpg 2200
dial-peer 2200
!
dial-peer voice 1100 voip
description Incoming ITSP Call Leg
session protocol sipv2
incoming uri via 1100
destination dpg 2200
dtmf-relay rtp-nte ; ITSP only supports one dtmf relay
codec g711ulaw ; ITSP only supports one codec
ip qos dscp cs3 signaling
no vad
!
dial-peer voice 1200 voip
description Outgoing ITSP Call Leg
destination-pattern ABC123
session protocol sipv2
session server-group 1200
voice-class sip options-keepalive profile 1200
dtmf-relay rtp-nte
codec g711ulaw
ip qos dscp cs3 signaling
no vad
!
dial-peer voice 2100 voip
description Incoming CUCM Call Leg
session protocol sipv2
incoming uri via 2100
destination dpg 1200
dtmf-relay sip-kpml rtp-nte ; we support both in- and out-of-band
internally and cube interworks dtmf
codec g711ulaw
ip qos dscp cs3 signaling
no vad
!
dial-peer voice 2200 voip
description Outgoing CUCM Call Leg
session protocol sipv2
session server-group 2200
destination-pattern ABC123
voice-class sip options-keepalive profile 2200
dtmf-relay sip-kpml rtp-nte
codec g711ulaw
ip qos dscp cs3 signaling
no vad
!
! a little something extra here at the end
alias exec attra show call active voice | in
PeerAddr|PeerId|RemoteS|RemoteM|Dtmf|Coder|VAD
alias exec attrh show call history voice | in
PeerAddr|PeerId|RemoteS|RemoteM|Dtmf|Coder|VAD
On Fri, Jun 12, 2020 at 6:36 PM Brian Meade <bmeade90 at vt.edu> wrote:
> Anthony,
>
> Thanks for the feedback. I'll definitely take a look at yours as well.
>
> Here's some answers on mine:
> 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.
> So I usually replace this with the name of the actual SIP carrier. This
> seems to make it easier for customers to understand but I understand so
> many other things are number tags only.
> 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
> Yea, I've never tried it without specifying the port. I've got a lot of
> SIP carriers with weird SIP ports so making it stand out in the template
> helps to know where to change this.
> 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
> That's a good idea. I actually exported this from a customer not
> realizing what it looks like when I use pref 0 and 1. Making it 1 and 2
> would make that look cleaner.
> 4. Why didn't you should your translation profiles and rules too?
> These seem to be so customer/SIP carrier specific that I didn't think it
> was worth it. My most recent one had 80 rules in it because the carrier
> really cares about 10-digit/11-digit calling for the local area code. So
> we actually had to split it up for different NPA-NXX whether or not we
> added a 1.
> 5. I don't specify UDP as the transport, since it's the default, but
> again, being explicit doesn't hurt anything
> I also make UDP my default but it is nice to have it called out in a
> template so people know where to change it if needed.
> 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
> Yea, I always add both to make sure we never have to pull in an MTP. I'm
> not aware of a way to do this globally but would be nice.
> 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.
> I'm always debugging faxing it seems like. Disabling ECM and reducing
> speed to 9600 has seemed to help a lot over the years. It's slower but
> seems to work more reliably with every source/destination fax device. And
> people don't expect their fax to send quickly anyways.
> 8. Since you're doing DPGs, you don't need the destination-pattern .T
> command on the outbound DPs.
> It seems like IOS-XE will show a dial-peer as down and skip it if there is
> no destination-pattern configured. This looks to be called out as
> explicitely required here even though it isn't used-
> https://cisco.com/c/en/us/td/docs/ios-xml/ios/voice/cube/configuration/cube-book/multiple-outbound-dial-peer.html
>
> Using something like ABC123 for the destination-pattern may make more
> sense to not confuse anyone. Good call.
> 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.
> I've never been a fan of SIP Options ping. I've always used SIP timers
> for failover instead. I guess I've had a few outages where waiting for
> Options Ping to come back up after we fixed the underlying issue added
> additional delay. For monitoring purposes though, it probably is a good
> idea to get back into doing that for customers where we're monitoring their
> CUBEs.
> 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 like that idea and referenced it in 8 above.
>
>
>
> On Fri, Jun 12, 2020 at 6:29 PM Anthony Holloway <
> avholloway+cisco-voip at gmail.com> wrote:
>
>> 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
>>>>
>>> _______________________________________________
>>> cisco-voip mailing list
>>> cisco-voip at puck.nether.net
>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>>
>>
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