Re: Voice: ~OT: terminating SIP service/trunk to CUCM7

In terms of just getting it working in a lab:
-You will probably still want to get a router to act as CUBE. CUBE
licensing is not enforced so it's just a matter of configuration.
This is largely to prevent a NAT requirement on your SIP/RTP traffic.
-The alternative is to use a firewall with NAT and make sure that the
SIP inspection works well enough to get calls through.
-If you do use the NAT approach without CUBE, you will want to find a
SIP trunk that does not require username/password/realm
authentication. CUCM doesn't send a register on behalf of it's
endpoints unlike CME/CUBE. Even with the CUBE approach it's not very
elegant for other reasons. It works fine with CME however.

There are approximately half a jillion SIP providers, so I don't think
it should be too bad. They may just not be very forthcoming with
their registration method before you give them money.

-nick

On Tue, Mar 16, 2010 at 1:26 PM, Ryan West <rwest_at_zyedge.com> wrote:
> Jason,
>
>> -----Original Message-----
>> Sent: Tuesday, March 16, 2010 2:14 PM
>> To: Cisco certification
>> Subject: Voice: ~OT: terminating SIP service/trunk to CUCM7
>>
>> forgive the 'lower level' quesiton on the list but... I'm confused and need
>> some basic direction
>>
>> Ok, i just got CM up and running in a VM here in my lab.
>>
>> first thing i want to do is get some kind of connection to the PSTN so i can
>> play with making outbound calls.
>>
>> I have no pots line so my only other option is some type of data circuit,
>>
>> first thought is a SIP service, i know some guys using callwithus.com with
>> CME and have successfully made calls through them with just an analog phone,
>> FXS card and a rtr...
>>
>> I assume CM is going to have to be setup as a UA (forgive the misuse of
>> terms if i've done so)
>>
>> I dont see any documentation on how to make CM act as a UA, it looks like
>> SIP realms will allow it to respond to a challenged from a UA but i'm
>> assuming thats not the same thing and wont work the same way
>>
>> sooo... can call manager not act as a UA, or am i just WAY off base and need
>> to read for a while?
>
> You need a router or CUBE to act as the UA and trunk your connection to the router, at that point you should be able to connect to any 3rd party SIP provider under the sun.
>
> https://supportforums.cisco.com/thread/300040;jsessionid=EB2CD650CB711EEAF9F092DC1F2B736C.node0?tstart=1
>
> Look for Nick's post, I believe he posts here on occasion and cisco-voip as well.
>
> -ryan
>
>
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Received on Wed Mar 17 2010 - 16:04:17 ART