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<DIV><FONT face=Arial size=2><SPAN class=674083817-08072005>My question: How do
I configure AAH via AMP to make a connection through our legacy PBX to the
PSTN?</SPAN></FONT></DIV>
<DIV><FONT face=Arial size=2><SPAN
class=674083817-08072005></SPAN></FONT> </DIV>
<DIV><FONT face=Arial size=2><SPAN
class=674083817-08072005>Details:</SPAN></FONT></DIV>
<DIV><FONT face=Arial size=2><SPAN class=674083817-08072005>We're trying out
Asterisk through Asterisk @ Home.</SPAN></FONT></DIV>
<DIV><FONT face=Arial size=2><SPAN
class=674083817-08072005></SPAN></FONT> </DIV>
<DIV><FONT face=Arial size=2><SPAN class=674083817-08072005>Our legacy PBX
has a modem type dial tone port that we hooked a Digium FXO
to.</SPAN></FONT></DIV>
<DIV><FONT face=Arial size=2><SPAN
class=674083817-08072005></SPAN></FONT> </DIV>
<DIV><FONT face=Arial size=2><SPAN class=674083817-08072005>Now I can
dial from the XTEN client on my computer to any legacy PBX
extension.</SPAN></FONT></DIV>
<DIV><FONT face=Arial size=2><SPAN
class=674083817-08072005></SPAN></FONT> </DIV>
<DIV><FONT face=Arial size=2><SPAN class=674083817-08072005>If I connect a
regular phone to the modem dial tone port, I can dial 9 to get an outside
line.</SPAN></FONT></DIV>
<DIV><FONT face=Arial size=2><SPAN
class=674083817-08072005></SPAN></FONT> </DIV>
<DIV><FONT face=Arial size=2><SPAN class=674083817-08072005>Replacing the phone
line back to the Digium FXO port, I cannot dial 9 and the phone number to route
my call to the PSTN through the legacy PBX.</SPAN></FONT></DIV>
<DIV><FONT face=Arial size=2><SPAN
class=674083817-08072005></SPAN></FONT> </DIV>
<DIV><FONT face=Arial size=2><SPAN class=674083817-08072005>Looking at the AMP
(Asterisk Management Portal)=>Outbound Route, I have two routes
created:</SPAN></FONT></DIV>
<DIV><FONT face=Arial size=2><SPAN class=674083817-08072005>PBX=>to several
legacy PBX extensions: 250, 270, 280 (these are the dial
patterns)</SPAN></FONT></DIV>
<DIV><FONT face=Arial size=2><SPAN class=674083817-08072005>9_outside=>to the
default dial pattern included with AAH: 9|. (that is the sole dial
pattern)</SPAN></FONT></DIV>
<DIV><FONT face=Arial size=2><SPAN
class=674083817-08072005></SPAN></FONT> </DIV>
<DIV><FONT face=Arial size=2><SPAN class=674083817-08072005>I wonder if the
digits get dialed too fast to connect to the PSTN? Can I put a pause in
somehow?</SPAN></FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><SPAN class=674083817-08072005><FONT face=Arial size=2>I can see that
Asterisk does grab the outside line.</FONT></SPAN></DIV>
<DIV><SPAN class=674083817-08072005><FONT face=Arial size=2>When I dial, I get
the following message, which I think is coming from the
PSTN:</FONT></SPAN></DIV>
<DIV><SPAN class=674083817-08072005><FONT face=Arial size=2>"We're sorry your
call did not go through, Will you please hang up and try your call again. This
is a recording."</FONT></SPAN></DIV>
<DIV><SPAN class=674083817-08072005><FONT face=Arial size=2>Any ideas
anyone?</FONT></SPAN></DIV>
<DIV><SPAN class=674083817-08072005><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=674083817-08072005><FONT face=Arial
size=2>Thanks,</FONT></SPAN></DIV>
<DIV><SPAN class=674083817-08072005><FONT face=Arial
size=2>--Bill</FONT></SPAN></DIV>
<DIV><SPAN class=674083817-08072005><FONT face=Arial size=2>Phoenix,
AZ</FONT></SPAN></DIV>
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