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It bit difficult to know what is happening when the "call is
immediately dropped." That is not a very precise description...<br>
However, in general you want to do a test on the uri and send to the
PSTN route of the getting started config if the uri matches a pstn
number. That's it.<br>
g-)<br>
<br>
Murray Hooper wrote:
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<p class="MsoNormal">Newbie to SER.<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">I am a little confused about how much I have to
set up in databases
to get an incoming SIP call to call forward to a PSTN number through
SER.
I have setup the username, attribute and value for my number,
“callfwd”, <a href="sip:1234567890@gateway.com">sip:1234567890@gateway.com</a>
in usr_preferences and the username in the subscriber file , added
gateway ip
to trusted, but call is immediately dropped when it arrives at SER. <o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">I have been through the getting started and have
had success
in modifying the values in the scripts, but don’t see what data must be
added into MySQL.<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">Getting pointed in the right direction would be
greatly
appreciated<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">Thanx<o:p></o:p></p>
<p class="MsoNormal">murray<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
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