[Asterisk-Users] Asterisk as a PSTN gateway for SER

[jerk face]

First off, here is what I want to do:
SIP Clients -> SER -> Asterisk -> VoIP provider

Where SER will handle communications between SIP
clients (since I would prefer that my SIP clients not
use all of my bandwidth)
Asterisk will handle calls to a VoIP provider

I have read that people have similar setups working,
but I have not seen any documentation of these setups.

So far, SIP Clients can talk to each other.
I can also start a call through Asterisk to a VoIP
provider, but there is a problem after the first ring:

Here is the output:
-- Executing Dial("SIP/-08114560",
"SIP/13239381067 at SIPprovider") in new stack
-- Called 13239381067 at SIPprovider
-- SIP/SIPprovider-5e0c is making progress passing it
to SIP/-08114560
-- SIP/SIPprovider-5e0c answered SIP/-08114560
-- Attempting native bridge of SIP/-08114560 and
SIP/SIPprovider-5e0c

I have tried this with my SIP client behind a NAT and
outside of a NAT, so I don't that is the problem.
I have also tried this with both IAX and SIP providers
and the problem is the same.  One ring, and then
silence.

Any thoughts?

Thank you for your time.


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