Cheap is relative - so not sure of any recommendations could make here. Bandwidth.com has worked in the past but not sure if that falls into cheap or not.
FYI - using the SPs in the drop down does not require you do any testing, so would that not be a better option (unless they are not in the same geographical area as you) - they also have partner programs
I'm currently using Nexvortex SIP trunks for testing. MRC for the trunks are $30. Unlimited outbound concurrent calls (depends on data bandwidth), unlimited incoming, 1250 minutes (US continental only). Any additional number (any rate center) is $5/mo. I got outbound calls working right away but I'm currently having issues with incoming calls. I'm trying to figure out why iIncoming SIP calls aren't going to the AA.
Something to check is that the AA is accessible from within the system (extension dialing) and I am sure you checked this. If its not working inside then I would open http:10.1.10.1 and check the CUE setting for AA number.
...but also make sure that same extension that you use (where I use 401) is in a translation rule mapping the DID to the AA:
(In CCA 1.9.1, this is set in the Configure:Telephony:AA drawer and the parameter is called the AA PSTN Number)
voice translation-profile AA_Profile translate called 2001
(make sure there is no conflicting AA DID Mapping using that smae DID in the Configure:Telephone:Voice Drawer - Dial Plan Tab -> Incoming Call translations)
Something to consider that will be visible in the 'debug ccsip messages' output is the CODEC being used and negotiated. Our default configuration allows G711 and G729, prefering G711. If you changed this and the SP doesnt do whats in your list, could be a problem.
If the calls are coming into the system as G729, they will need to be transcoded to G711 to hit the CUE (both AA and Voice Mail work with G711 only).
I doubt there are resource (DSP) constraints that would prevent that, and I know there are dial-peer settings to insure we use the proper DTMF technique, but that would only manefest itself as being connected to the AA and not being able to tone anthing into it, so maybe not so likely a problem in your case. :-)
I have noticed that is a standard practice for SIP trunk providers to position SLA capable circuits instead of over the top, consumer grade access. How else can they be accountable for voice quality issues?
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