Please do not post same thing again and again . It wont help you get better replies , Post you asterisk cli output while call is in progress and when it disconnects prematurely .<br><br><div><span class="gmail_quote">On 18/06/07,
<b class="gmail_sendername">Don Kelly</b> <<a href="mailto:dk@donkelly.biz">dk@donkelly.biz</a>> wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
I am attempting to establish SIP peering between Asterisk and an AltiGen<br>soft PBX. This is my first experience with SIP peering.<br><br>I can successfully make both inbound and outbound calls to/from a softphone<br>on the AltiGen system (network access is provided by a PRI on the Asterisk
<br>system), but they are disconnected unexpectedly.<br><br>The attachment is a redirect of the Asterisk CLI during a call that is<br>disconnected prematurely.<br><br>Here's what's in SIP.conf:<br><br>[altigen]<br>
type=friend<br>username=altigen<br>secret=coolbeans<br>host=dynamic<br>deny=<a href="http://0.0.0.0/0.0.0.0">0.0.0.0/0.0.0.0</a><br>permit=<a href="http://10.0.2.150/255.255.255.255">10.0.2.150/255.255.255.255</a><br>qualify=yes
<br>disallow=all<br>allow=ulaw<br>context=altigen-inbound<br>dtmfmode=rfc2833<br><br>The machines are a couple feet apart on a LAN through a 100MB switch.<br><br>I'd appreciate any help.<br><br> --Don<br><br>Don Kelly
<br>PCF Corp<br>Real Support for your Virtual Office<br>651 842-1000<br>888 Don Kell(y)<br>651 842-1001 fax<br><br><br>_______________________________________________<br>--Bandwidth and Colocation provided by <a href="http://Easynews.com">
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