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

[Greger Viken Teigre]

I think this has to do with ACK handling. One can probably argue that this 
is a bug in the config, but due to a complete config change in 2.0, it was 
never fixed.   
If I recall correctly, there is a missing relaying of acks without route 
headers. I'm on mobile, so I cannot check.
g-)

------- Original message -------
From: Stefan Sayer <stefan.sayer at iptego.com>
Cc: serusers at lists.iptel.org
Sent: 6.2.'08,  20:59

> 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
> 
> -- 
> Stefan Sayer
> VoIP Services
> 
> stefan.sayer at iptego.com
> www.iptego.com
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