<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
<head>
<meta content="text/html;charset=ISO-8859-1" http-equiv="Content-Type">
<title></title>
</head>
<body bgcolor="#ffffff" text="#000000">
In a2billing just change the 9 to what you need it is right in the conf
file.<br>
<pre class="moz-signature" cols="72">Best regards,
Al Bochter
Bochter Services
----------------------------------------------------------
Need to call me use our web phone at the link below
<a class="moz-txt-link-freetext" href="http://www.bochterservices.com/voip/iaxphone.php?cn=250">http://www.bochterservices.com/voip/iaxphone.php?cn=250</a>
----------------------------------------------------------
Can you WIN gold today? Click on the link and see.
<a class="moz-txt-link-freetext" href="http://www.bochterservices.com/?t=USbill_email">http://www.bochterservices.com/?t=USbill_email</a>
----------------------------------------------------------
Need cash we buy silver and gold
----------------------------------------------------------</pre>
<br>
<br>
Nitesh Divecha wrote:
<blockquote cite="mid467838B8.2060304@vipernetworks.com" type="cite">
<pre wrap="">Thanks everyone for the input...
In real world we can not ask the customers to dial 9, if they want to
call another SIP user... and trust me its confusing for a customer
also... meaning when to dial 9 and when to not...
We have a custom proprietary system which does this part very well...
Before it sends the call on a Trunk it will check the DID, if it exists
within the local system. If it does then it will just use IP to IP call,
else send the call to Trunk...
I think its possible to do this by creating some basic dial plans...
Same like creating local extensions.
Cheers,
Nitesh
John Novack wrote:
</pre>
<blockquote type="cite">
<pre wrap="">Given that Asterisk is modeled on, in the telephone industry, an
obsolete PBX design, without many of the modern day hybrid features, and
only recently has any effort been made to provide buttons and lights for
"lines" ( Is that yet working in 1.4??) one would have to do some very
careful number parsing to not use a trunk digit.
If every phone in the system had buttons and lights representing
external connections and internal connections on other button(s) (
intercom ) this wouldn't be an issue.
Most "legacy" systems have been able to do this for the last 20 years or so.
John Novack
Nitesh Divecha wrote:
</pre>
<blockquote type="cite">
<pre wrap="">Thanks man,
Is there any other way without dialing 9... it will be kinda pain for a
customer to dial 9 every time and plus they need to know also...
Is there any intelligent way to identify? if its a local SIP then don't
route to Trunk else route to Trunk.
Cheers,
Nitesh
Guillermo Salas M. wrote:
</pre>
<blockquote type="cite">
<pre wrap="">On Tue, 2007-06-19 at 09:36 -0400, Nitesh Divecha wrote:
</pre>
<blockquote type="cite">
<pre wrap="">Thanks man...
So far everything worked as expected...
How can I make internal calls stay within the PBX. For example, when
one
SIP-Friend tries to call another SIP-Friend without sending the call
out
on Trunk and receive it back. Same like dialing from one extension
number to another extension.
My SIP-Friends are using US DID numbers and I would like to keep the
local calls within the network.
Right now when I try to call other SIP-Friend, I get a message saying
"The number you have dialer is currently not available"... while the
SIP-Friend is registered.
</pre>
</blockquote>
<pre wrap="">Try dialing the number 9 before the sip/iax2 friend number.
Regards,
</pre>
<blockquote type="cite">
<pre wrap="">Cheers,
Nitesh
</pre>
</blockquote>
</blockquote>
<pre wrap="">_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
<a class="moz-txt-link-freetext" href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</a>
</pre>
</blockquote>
<pre wrap="">_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
<a class="moz-txt-link-freetext" href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</a>
</pre>
</blockquote>
<pre wrap=""><!---->
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
<a class="moz-txt-link-freetext" href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</a>
----------------------------------------------------
Inbound (clean). Database: 000750-2, 06/19/2007 - 6/19/2007 4:22:13 PM
</pre>
</blockquote>
</body>
</html>