check [general] section of your /etc/asterisk/sip.conf<br><br>disallow=all<br>allow=alaw<br>allow=ulaw<br>allow=gsm <br><br>This codecs depends on of your SIP provider as well as activation in your SIPphone<br><br><div><span class="gmail_quote">
On 5/30/06, <b class="gmail_sendername">George A. Roberts IV</b> <<a href="mailto:groberts@interjuncture.com">groberts@interjuncture.com</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;">
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<div><span><font face="Verdana" size="2">I just put in a
new <a href="mailto:Asterisk@Home" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">Asterisk@Home</a> 2.8 system. Trunk is
connected via SIP to ViaTalk.</font></span></div>
<div><span><font face="Verdana" size="2"></font></span> </div>
<div><span><font face="Verdana" size="2">I had an older <a href="mailto:Asterisk@Home" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">Asterisk@Home</a> system up and running that was
working fine and I replicated settings over to the new box. When I call
7777 from an internal SIP extension I can hear the IVR menu just fine.
However, when I call from a POTS phone to our number and it comes in via ViaTalk
over SIP the call connects but I do not get any sound. I'm sure it's a
setting or something I missed, but I'm not sure what it is. Anyone have
any ideas?</font></span></div>
<div><span><font face="Verdana" size="2"></font></span> </div>
<div><span><font face="Verdana" size="2">George</font></span></div></div>
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