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New Voice Gateway



I have been tasked with setting up a Cisco 4431 to replace our outdated 3825.  Instead of setting it up MGCP as the 3825 is, management wants it set up SIP. 


This is not my strongest area of expertise and I'm having a hard time wrapping my head around voice translations and dial peers.  If I understand the call flow correctly, inbound calls will be passed to CUCM and it will decide routing.  Outbound calls from CUCM will flow to the gateway where it will decide routing.  I'm stuck and need some guidance as to where to go from here.

So far this is what I have for dial peers:

dial-peer voice 2 voip
description incoming calls from CUCM
session protocol sipv2
incoming uri via 1
voice-class codec 1
dtmf-relay sip-kpml rtp-nte
dial-peer voice 10 voip
description outgoing calls to CUCM
dial-peer voice 11 pots
description outgoing calls to PSTN Local
dial-peer voice 1 voip
description incoming calls from PSTN
session target ipv4:
dial-peer voice 12 pots
description outgoing calls to PSTN LD


I'm having trouble filling in the blanks for the dial peers.  This is what I have for voice-translations (we dial 7 to get out).

voice translation-rule 10

rule 1 /^1\(.*\)/ /\1/


voice translation-rule 20


rule 1 /^7\(.*\)/ /\1/
I would appreciate any guidance that you can provide.  Attached is the entire show run.
Thank you,

Re: New Voice Gateway

Hello Tony,

I would strongly suggest to read the following guide that describes the dial-peers and the call flow, when using CUBE:


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Re: New Voice Gateway

Thanks George.  I read through the document but what I'm really hung up on is outgoing dial-peers to pots.  The outbound calls will be going out 5 PRIs, 2 local and 3 LD.  How do I tell the gateway which port to send the call out?

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Re: New Voice Gateway

With SIP and H323 call routing is performed on both, the GW and CUCM. Only with MGCP you depend on CUCM for all call routing.

Look at H323 configuration docs to get an idea of this, pretty much the same thing for call routing. 



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Re: New Voice Gateway

I've decided to just focus on making an outbound long distance call for starters as I have an unused LD PRI laying around.  I've set up  the SIP trunk in CUCM and it is in Full Service.  Set up the RG and RL and a route pattern pointing to that SIP trunk.  However, I'm getting a busy signal and can't figure out why.  I've attached the config.  Any insight would be greatly appreciated. 

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