Connect you CME based VoIP network to an external VoIP provider

When you are building a converged network you are integrating Data, Voice and Video. The voice network has to be connected the telephony network of your Telco. The most common way how to do it, is just to use some telephony interfaces (FXO, E&M, T1/E1, ...) on your ISR router which will be connected directly to a Telco's endpoint. By using this way you have to have additional cards in your router, the management may be more difficult and the costs are usually higher.
Other approach how to connect your VoIP based telephony network to a Telco is directly with VoIP trunk. Why should you get one Internet connection and other different telephony connections if you can integrate all those services also on your network edge, just by reusing your data (Internet) connection. In this way you can save your company some money by lovering costs for different network accesses, your network is more easily managed, and you can implement all of your required converged network functionality in one device - in an ISR router with Unified Communications Manager Express.

What you will need is a Telco operator which provides VoIP services. Nowadays almost every Telco has also VoIP services in his list.
You have to get some information about the provided VoIP service by your Telco - which signalizing protocol they use - usually SIP and in some cases it may be H323. Then you should ask about supported codecs - only uncompressed G.711 or also a better choice for a WAN link - G.729.
The next step is to configure your CME service with an external SIP trunk. This can be accomplished in many different ways.
Depending on your Telco - he may provide a "real" SIP trunk which is a predefined voice route to the IP address of your router. For this kind of service you must have a public static IP address on the WAN side. If your operator provide this kind of service, you should get it. It's the best choice. Then the configuration of CME is really simple - consist just in creating a VoIP dial peer to PSTN numbers directing to your Telco's VoIP server.

! this configuration uses by default the G.729 codec. If your telco has no support for this codec, you must configure also a codec parameter for this dial peer
dial-peer voice 1 voip
 destination-pattern 9.
 session protocol sipv2
 session target dns:myvoicesp.dot

A bit more complicated is a situation when your Telco provides the VoIP service in a "true" SIP mode - in which your device has to register itself to the VoIP server of Telco. Usually the Telco assign your an username and password which is then used in the registration process. The advantage of this solution is that you may have a dynamic IP address on the WAN side, or even in some situations you may have a private IP address behind NAT. The CME's configuration for this solution may a a bit tricky and sometimes you may spend few minutes with a debug lines. Anyway, after configuring it works with the same quality like the previous solution.

sip-ua
authentication username VoIPUSERNAME password VoIPPASSWORD
registrar dns:myvoicesp.dot expires 300
sip-server dns:myvoicesp.dot

If you are deploying this kind of telephony network, please don't forget to configure the QoS parameters for your Internet connection. Otherwise if someone in your network will download a movie from the Internet, his data transfer will badly affect the telephone calls and it can even lead to call disconnections.

For more information about this topic please refer to:
Real configuration example with the registration solution
Securing SIP Gateways
Authentication in dial-peers and sip-ua