[Asterisk-Users] Sipura 3000 does not dial out

Chris Mason (Lists) lists at masonc.com
Mon May 30 19:53:34 MST 2005


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 at pstn-spa3k|60|") in new
stack
    -- Called 2355670 at 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 at home, change the below line to context=from-internal
context=internal
insecure=very


Any ideas?

Chris Mason




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