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

From: Ryan West <rwest_at_zyedge.com>
Date: Tue, 16 Mar 2010 18:26:02 +0000

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

Blogs and organic groups at http://www.ccie.net
Received on Tue Mar 16 2010 - 18:26:02 ART

This archive was generated by hypermail 2.2.0 : Thu Apr 01 2010 - 07:26:35 ART