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<DIV><SPAN class=451080815-27042006><FONT color=#0000ff size=2>I actually tried
that before but it didnt seem to work. I tried once again and still nothing
rings, whether I set the destination to a single extension, or a ring group. But
the suggestion from another user below did work, but wont go to voicemail yet
when its not answered.</FONT></SPAN></DIV>
<DIV><SPAN class=451080815-27042006><FONT color=#0000ff
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=451080815-27042006><FONT size=2>
<P>[from-pstn]</P>
<P>include => from-pstn-custom ; create this context in</P>
<P>extensions_custom.conf to include customizations</P>
<P>include => ext-did</P>
<P>exten => _.,1,Wait(1)</P>
<P>exten => _.,2,Goto(from-pstn,s,1)</P>
<P>exten => s,1,Answer</P>
<P>exten => s,2,Dial(SIP/100,20)</P></FONT></SPAN></DIV>
<BLOCKQUOTE dir=ltr
style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px solid; MARGIN-RIGHT: 0px">
<DIV class=OutlookMessageHeader dir=ltr align=left><FONT face=Tahoma
size=2>-----Original Message-----<BR><B>From:</B> Alex Robar
[mailto:alex.robar@thegoldfish.net]<BR><B>Sent:</B> Thursday, April 27, 2006
7:32 AM<BR><B>To:</B> Asterisk Users Mailing List - Non-Commercial
Discussion<BR><B>Subject:</B> Re: [Asterisk-Users] Unable to accept incoming
PSTN calls<BR><BR></FONT></DIV>
<P>Johnny,<BR><BR>You need to setup an Inbound Route that matches all DIDs and
all CIDs. In FreePBX, click on Inbound Routes, create a new route with blank
CID and DID, and point it where you want it to go. It should work after that.
<BR><BR>Alex<BR><BR>
<DIV><SPAN class=gmail_quote>On 4/27/06, <B class=gmail_sendername>Johnny
Stork</B> <<A
href="mailto:stork@openenterprise.ca">stork@openenterprise.ca</A>>
wrote:</SPAN>
<BLOCKQUOTE class=gmail_quote
style="PADDING-LEFT: 1ex; MARGIN: 0pt 0pt 0pt 0.8ex; BORDER-LEFT: rgb(204,204,204) 1px solid">Since
I am using A@H 2.8 which now uses freePBX, there does not seem to be a menu
area/settings for "Incoming Calls"?<BR><BR>If you have a similiar setup, or
know what the settings should be, could you possibly post them? If I were to
create a dial group <BR>to ring all extensions, could that be used in place
of "s"?<BR><BR>Thanks kindly<BR><BR>> -----Original Message-----<BR>>
From: Time Bandit [mailto:<A
href="mailto:timebandit001@gmail.com">timebandit001@gmail.com </A>]<BR>>
Sent: Thursday, April 27, 2006 6:19 AM<BR>> To: Asterisk Users Mailing
List - Non-Commercial Discussion<BR>> Subject: Re: [Asterisk-Users]
Unable to accept incoming PSTN calls<BR>><BR>><BR>> >
[from-pstn] <BR>> > include => from-pstn-custom ; create this
context in<BR>> extensions_custom.conf to include customizations<BR>>
> include => ext-did<BR>> > ;exten =>
fax,1,Goto(ext-fax,in_fax,1)<BR>> > exten => _.,1,Wait(1) <BR>>
> exten => _.,2,Goto(from-pstn,s,1)<BR>><BR>> Here is what is
happening :<BR>><BR>> Your ZAP channels are in the context
"from-pstn"<BR>> Since there is no "s" extension defined, it goes to "_."
<BR>> (which match anything)<BR>><BR>> So, like seen in the log,
Asterisk wait a second, then execute<BR>> "Goto(from-pstr,s,1)" which
brings it back to "_.,1". It just loop<BR>> there until the caller hangup
<BR>><BR>> Since you're using A@H, you have to go into AMP (or
FreePBX) and click<BR>> on Setup -> Incoming Calls and define
something to do with incoming<BR>> calls<BR>><BR>> hth<BR>>
_______________________________________________ <BR>> --Bandwidth and
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http://lists.digium.com/mailman/listinfo/asterisk-users</A><BR>><BR>_______________________________________________<BR>--Bandwidth
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visit:<BR> <A
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clear=all><BR>-- <BR>Alex Robar<BR><A
href="mailto:alex.robar@gmail.com">alex.robar@gmail.com</A>
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