[Asterisk-Users] getting asterisk to reliably answer a voip line

[jnuoiqweahf kajhdsff]

I have a sipphone.com account, with asterisk set to
answer incoming calls, using the following settings
(phone number and password omitted) in the Peer
Details for the SIP Trunk: 

allow=ulaw 
context=from-pstn 
dtmfmode=rfc2833 
fromdomain=proxy01.sipphone.com 
fromuser=1747xxxxxxx 
host=proxy01.sipphone.com 
insecure=very 
secret=xxxxx 
type=peer 
username=1747xxxxxxx 

The Asterisk machine is behind a Linksys router (full
cone NAT). 

About 25% of the time, when I call that number (from
another sipphone account), asterisk answers the line,
but about 75% of the time, asterisk fails to answer,
and doesn't even indicate that any incoming call was
attempted, and sipphone times out after 15-20 seconds
and dumps the unanswered call to its voicemail system.
I don't see any pattern to the intermittent answering,
and sometimes I can try numerous times and get no
answer, and sometimes I can try several times in a row
and get an answer each time. It seems random. Outgoing
calls work 100%; only incoming are having problems. 
How can I diagnose whether the problem is with
Asterisk or with Sipphone, or whether one or both are
having problems because of NAT? Bypassing the NAT
router is not an option, even for testing. Is this a
known problem with Sipphone? How do the various voip
providers (Sipphone, FWD, Broadvoice, etc) compare
with regards to incoming call completion reliability
when the receiving device (Asterisk in this case) is
behind NAT? 

I'll eventually need to accept incoming PSTN calls via
voip and I'm willing to pay for reliable service from
any provider, but I do need Asterisk to actually
receive and answer all attempted incoming calls.

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