I have a Sipura 3000 ATA I am testing as a PSTN gateway, I have little interest in the fxs port, lthough the failover is cool I upgraded to the latest firmware. I used the wazard on Voxilla to configure the unit. Incoming works fine. When I dial out, the dial string is sent to the unit: -- Executing Dial("SIP/1001-9e54", "SIP/2355670@pstn-spa3k|60|") in new stack -- Called 2355670@pstn-spa3k -- SIP/pstn-spa3k-5a70 is ringing -- SIP/pstn-spa3k-5a70 answered SIP/1001-9e54 -- Attempting native bridge of SIP/1001-9e54 and SIP/pstn-spa3k-5a70 But all I get, after a few seconds, is a busy signal. I know the line is not busy, the number I am dialing is available, and I have tested the line with a handset. [pstn-spa3k] type=peer auth=md5 host=192.168.0.14 port=5061 secret=mysecret username=asterisk fromuser=asterisk dtmfmode=rfc2833 ; If using Asterisk@home, change the below line to context=from-internal context=internal insecure=very Any ideas? Chris Mason