After seeing a few posts on the site regarding ITSPs I decided to bite the bullet and try one out. I chose nexVoterx at random and must say I was impressed. They had a very automated setup that assisgned me my new phone numbers in minutes. I got three numbers and they were in my local area code, though I could have chosen a number from anywhere in the US.
I used the Cisco configuration assistant to configure the SIP trunk and it did a great Job with one noteable exception.
All Calls to my phones work fine, but and calls directed to the auto attendant or transferred to voicemail fail. I found there were three main settings to change to give me the functionallity I needed.
Step 1
Type the following to allow hairpinnig of calls to the Unity Express module.
config t
voice service voip
allow-connections sip to h323
Step 2
Change the DTMF Relay type to rtp-nte under the autoattendant's dial peer.
dial-peer voice 2001 voip
description ** cue auto attendant number **
translation-profile outgoing PSTN_CallForwarding
destination-pattern 498
session protocol sipv2
session target ipv4:10.1.10.1
dtmf-relay rtp-nte
codec g711ulaw
no vad
Step 3
Allow SIP through the firewall to the Unity Express Module
The Cisco Configuration Assistant does an OK job of letting SIP Traffic in (from the internet) to phones, but the firewall must be manually modified to allow SIP traffic from the internet to the auto attendant. If you have a private SIP connection on an internal interface you should not have to modify the firewall.
I found this Guide at Cisco's site for configuring SIP on the Router based firewall.