<HTML><BODY style="word-wrap: break-word; -khtml-nbsp-mode: space; -khtml-line-break: after-white-space; ">Thank you for your response!<DIV>That was exactly the problem - the 841s use a deprecated identifier for G.726. Behavior and solution is described here:</DIV><DIV><BR class="khtml-block-placeholder"></DIV><DIV><A href="http://www.voip-info.org/wiki/view/Sipura+SPA-1001">http://www.voip-info.org/wiki/view/Sipura+SPA-1001</A></DIV><DIV><FONT class="Apple-style-span" face="Verdana" size="3"><SPAN class="Apple-style-span" style="font-size: 12px;">1) To use G726 with asterisk 1.2.6 or later you must edit the rtp.c and either dfine USE_DEPRECATED_G726=1 or remove the #ifdef USE_DEPRECATED_G726 statement. If you do not do this you will be able to place calls using the SPA but not receive calls. This is because the SPA identifies the G726 codec using a deprecated paylode type.</SPAN></FONT><FONT class="Apple-style-span" face="Verdana" size="3"><SPAN class="Apple-style-span" style="font-size: 12px;"> </SPAN></FONT></DIV><DIV><BR class="khtml-block-placeholder"></DIV><DIV>I did that, make, make install, asterisk -rx "restart gracefully", and viola! it works.</DIV><DIV><BR class="khtml-block-placeholder"></DIV><DIV>THANK YOU!</DIV><DIV><BR class="khtml-block-placeholder"></DIV><DIV>Guy</DIV><DIV><BR><DIV><DIV>On Sep 25, 2006, at 6:33 PM, Eric ManxPower Wieling wrote:</DIV><BR class="Apple-interchange-newline"><BLOCKQUOTE type="cite"><DIV style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; ">That error message is almost always because the two sides cannot agree on a codec.<SPAN class="Apple-converted-space"> </SPAN>HOWEVER, if you are using SIPura and G726, there is a Makefile option for Asterisk to make it work.</DIV><DIV style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; min-height: 14px; "><BR></DIV><DIV style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; ">Guy M Guyadeen wrote:</DIV> <BLOCKQUOTE type="cite"><DIV style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; ">Our pbx is a Fedora Core 4 box running Asterisk 1.2.6. It has a public IP and is publicly accessible (no NAT, firewall, etc.).</DIV><DIV style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; ">Our Sipura 841 phones are in several locations, all NAT'ed behind PIX firewalls.</DIV><DIV style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; ">The system used to work flawlessly, but now any extension we dial comes back as "busy". On the Asterisk CLI, I always see something like:</DIV><DIV style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; ">Got SIP response 488 "Not Acceptable Here" back from {our IP address}</DIV><DIV style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; "><SPAN class="Apple-converted-space"> </SPAN>-- SIP/{extension}-d544 is circuit-busy</DIV><DIV style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; "><SPAN class="Apple-converted-space"> </SPAN>== Everyone is busy/congested at this time (1:0/1/0)</DIV><DIV style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; ">Would extensions.conf or CLI debug output be helpful?</DIV><DIV style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; ">Thank you in advance!</DIV><DIV style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; ">Guy</DIV><DIV style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; ">_______________________________________________</DIV><DIV style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; ">--Bandwidth and Colocation provided by Easynews.com --</DIV><DIV style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; ">asterisk-users mailing list</DIV><DIV style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; ">To UNSUBSCRIBE or update options visit:</DIV><DIV style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; "><SPAN class="Apple-converted-space"> </SPAN><A href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</A></DIV> </BLOCKQUOTE><DIV style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; min-height: 14px; "><BR></DIV><DIV style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; ">_______________________________________________</DIV><DIV style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; ">--Bandwidth and Colocation provided by Easynews.com --</DIV><DIV style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; min-height: 14px; "><BR></DIV><DIV style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; ">asterisk-users mailing list</DIV><DIV style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; ">To UNSUBSCRIBE or update options visit:</DIV><DIV style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; "><SPAN class="Apple-converted-space"> </SPAN><A href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</A></DIV><DIV style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; min-height: 14px; "><BR></DIV> </BLOCKQUOTE></DIV><BR></DIV></BODY></HTML>