[cisco-voip] Can a CUBE be used to connect CUCM and Asterisk to a SIP Trunk

Matthew Saskin msaskin at gmail.com
Fri Dec 7 22:54:05 EST 2012


Direct CUBE to CUCM is easy.  The following is a snippet from my asterisk
lab sip.conf - very simple, no authentication happening, but sets up
trunking to both nodes in the lab cluster and dumps inbound calls to a
common context for routing.  On the CUCM side, it's nothing but a pretty
standard SIP trunk config with nothing changed from the defaults.

[cucm9pub]
type=friend
context=from-callmanager
host=192.168.11.50
disallow=all
allow=ulaw
nat=no
qualify=yes

[cucm9sub]
type=friend
context=from-callmanager
host=192.168.11.51
disallow=all
allow=ulaw
nat=no
qualify=yes


Matthew Saskin
msaskin at gmail.com
203-253-9571



On Sun, Dec 2, 2012 at 11:39 AM, Nick Matthews <matthnick at gmail.com> wrote:

> Realistically you could probably trunk them together without CUBE. CUCM
> has a lot more capability in altering the inbound/outbound SIP settings
> that CUBE was used for previously. CUBE would make it potentially easier
> and still makes the media exchange simpler, but my guess is with Asterisk
> for 50 phones this design aspect isn't significant.
>
> I haven't seen any guides for either CUCM or CUBE connectivity to
> Asterisk, but they're probably out there on the interwebs somewhere.
>
> -nick
>
>
>
> On Sun, Dec 2, 2012 at 10:33 AM, Matthew Saskin <msaskin at gmail.com> wrote:
>
>> This would not be a problem at all.  The short answer is you'd end up
>> having dial peers on the CUBE that route (via destination pattern, etc.)
>> specific calls/DID's to the asterisk IP address and specific calls/DID's to
>> the be6k address.
>>
>> -matthew
>>
>> Matthew Saskin
>> msaskin at gmail.com
>> 203-253-9571
>>
>>
>>
>> On Sun, Dec 2, 2012 at 12:40 AM, Dieter Jansen <justtemprament at gmail.com>wrote:
>>
>>> Hi All,
>>>
>>> We are looking to replace an existing Asterisk PBX with about 40 SIP
>>> handsets with a CUCM UC solution (BE6K) with about 50 new CP-8945s.
>>>
>>> The current installation uses a SIP trunk and we plan to keep using this
>>> to connect to the PSTN.  We have a 100 DID block for our incoming calls.
>>>
>>> As we have comparatively little experience with the Cisco solution we're
>>> looking at a soft introduction and will probably have both sets of handsets
>>> in place in parallel for a while.
>>>
>>> I've got all the edge switches and VLANing working - independent data,
>>> voice and legacy voice VLANs/subnets and all required routing between.
>>>  We're at the point where test phones can place video calls to each other
>>> so we now need to get routing and PSTN access happening
>>>
>>> Originally I was thinking I'd set up a CUCM-behind-Asterisk
>>> or Asterisk behind-CUCM situation whilst testing but on reading the Cisco
>>> documents and books I've noted the suggested use of a CUBE SBC.  As we have
>>> a spare router that can probably do CUBE I was wondering if I should set
>>> that up to register with our VSP.
>>>
>>> In the first instance I'd get the Asterisk box to pass calls to the CUBE
>>> and have the CUBE forward all incoming calls to the Asterisk.
>>>
>>> What I'm hoping is that I could then also have the CUCM pass calls to
>>> the same CUBE and route a subset of the inward DIDs to the CUCM whilst
>>> testing - allowing me to get all the setup sorted whilst easily flipping
>>> incoming calls to specific DID between Asterisk and CUCM.
>>>
>>> I've not seen an example of the CUBE being used in this way - is it
>>> possible?
>>>
>>> Regards, JustTemprament.
>>>
>>>
>>>
>>> _______________________________________________
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>>> cisco-voip at puck.nether.net
>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>>
>>>
>>
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>
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