2 CME sites, to save on long distance toll

[Y W]

There are two CME sites, site a and site b, connected by a lease line / mpls / vpn. The two sites are able to ping each other.

The end goal is for site a to dial a prefix, let's say 0099+site b local number, so that a call will be forwarded to site b via destination-pattern, that way when the call is forwarded to site b and comes out, it will be a local call to site b cme, instead of paying long distance from site a.

My question is, what configuration is needed on the site b end so the call can be fetched from the dial-peer and forward it out to a connected t1 line?

I know that I will use a translation profile to strip off the 0099, but other than that, how do I get this done?

Thanks in advance.

[Jaime Valencia]

Nothing besides your usual config, a dial peer on the far end router, you can have an inbound dial peer in the receiving router if you want to use a certain config, and any digit manipulation you might need for calling/called number, based on the dialed digits and the dial-peers you already have.

When the call arrives from the other CME, it will just try to match something in your dial plan to route.

[Y W]

Thank you for the response Jaime,

so assuming 6 is the number you dial to get outside:

Site A

dial-peer voice 10 voip

destination-pattern 00996...........

session target: ipv4:SiteBIP

Site B

dial-peer voice 10 voip

incoming called-number 00996..........

translation-profile incoming stripe0099

port 0/0:23

that will work directly? or i need to have another dial-peer on B side that is

diak-peer voice 20 pots

destination-pattern 6...........

port 0/0:23

Do i need the command direct-dial-inward? should I ask Site A to dial 6 after the prefix 0099 like siteB normally dial to get out?

[Mohammed al Baqari]

Hi, site B dialpeer 10 won't access port command cuz its voip dialpeer. and yes you need to configure dialpeer 20 to break the call out to pstn.

This much more config on the CMEs including codecs, binding, dtmf-relay, bind, etc.

http://www.cisco.com/c/en/us/support/docs/voice-unified-communications/unified-communications-manager-express/91535-cme-sip-trunking-config.html

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