[Spce-user] PSTN SIP Peering
Juan José Ivars
juanjo at beeztel.com
Fri Jun 8 04:18:50 EDT 2012
I have resolved the issue, i forgot to configure a
Peering Rule
Best regards.
2012/6/8 Juan José Ivars <juanjo at beeztel.com>
> 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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