[Spce-user] PSTN SIP Peering

[Juan José Ivars]

Hello

I have configured a trunk with an Asterisk to send calls to PSTN, but the
call never arrives to Asterisk system, if i capture traffic in SipWise i
get:
503 PSTN termination currently unavailable

My configuration in Asterisk is:

[sipwise]
type=friend
context=sipwise
host=XX.XX.XX.XX
insecure=very
disallow=all
allow=alaw

I can see that peer registered through Asterisk CLI.


Can anybody point me in the right direction?


Thanks in advance.
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