SIP UA Trunk through CME to CUCM

Unanswered Question
Mar 19th, 2008
User Badges:

I'm trying to get a scenario working, thus far i'm not sure what i'm doing wrong - hopefully there are some examples people can provide.

I have a CUCM on the inside of the network (10.x) with a standard 2811 ISR acting as a PSTN gateway (FXO) plus that ISR also has a WIC-1ADSL for internet connectivity.

I want to bring in a SIP trunk from a provider, they can't really give me any configuration advice unless i'm using Asterisk. It appears I need to register with their SIP UA, and then they forward the calls to me.

Can I somehow bring this trunk in from outside the firewall to the CUCM or do I need to terminate this at the gateway an then send it through?

Any config examples would be quite helpful.

Will I need IPIPGateway to do this? If I do, going to an IPIPGateway load will this kill my PPPoA feature I need for the ADSL? (For some reason the IPIPGW loads don't have feature lists on CCO)

I'm sure someone has completed what i'm trying to do. The provider is Link2Voip.


  • 1
  • 2
  • 3
  • 4
  • 5
Overall Rating: 4 (2 ratings)
paolo bevilacqua Wed, 03/19/2008 - 09:37
User Badges:
  • Super Gold, 25000 points or more
  • Hall of Fame,

    Founding Member

Hi, the IP-to-IP images does all what a regular iamges doesn, plus ip-to-ip calls and few more advanced voice things.

The cisco recommendation is that you use that to terminate SIp trunks from providers. Make sure you have an access-list to block port 5060 to everyone but the provider else you may e subkject to toll fraud.

The codec usually is specified as "transparent". Beside that once you've it in place, and have problems, just ask here.

hope this helps, please rate post if it does!

justincohen Wed, 03/19/2008 - 10:28
User Badges:

Thanks for the information... Do you have a sample config you could post for this?

paolo bevilacqua Wed, 03/19/2008 - 11:08
User Badges:
  • Super Gold, 25000 points or more
  • Hall of Fame,

    Founding Member

Hi, config is limited to "allow sip to sip" and voip DPs with proper address and numbers.

There are examples in the CUBE config guide, but I'm reluctant to cite them here, because they contain a lot of stuff that is not said to apply. So better to start minimally first.


This Discussion