[Serusers] Is SER doc PSTN Gateway example incomplete/incompatible with Asterisk?

[Stefan Sayer]

Hello,

Frank Durda IV wrote:

> Thanks for the catch!
> 
> Now with 192.168.200.30 re-added to the domain table, things get further
> and a test call claims "Connected" at the SIP phone display for exactly
> 30 seconds before the SIP phone reports that the Call Ended.
> Based on the logs it does not appear that SER attempted to contact the
> PSTN switch, but SER certainly got closer.
to me it looks like the INVITE is sent out and retried. an ngrep would 
definitely be more revealing about what happens here (ngrep port 5060 -W 
byline -d any).

Stefan

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