Bob, can you paste your /etc/dahdi/system.conf, users.conf, and extensions.conf? Also if you are connected to analog lines have you run fxotune?<br><br>
<div class="gmail_quote">On Tue, Jul 14, 2009 at 11:45 AM, Bob Crandell <span dir="ltr"><<a href="mailto:bob@assuredcomp.com">bob@assuredcomp.com</a>></span> wrote:<br>
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<div>Hi All,</div>
<div> </div>
<div>Thanks to the help I'm getting here I can now make calls and the voice quality sounds pretty good.</div>
<div> </div>
<div>An incoming call does not get answered. This is the log:</div>
<div>Connected to Asterisk 1.6.0.9 currently running on PBX (pid = 5068)<br>Verbosity is at least 6<br> -- Starting simple switch on 'DAHDI/1-1'<br>[Jul 14 08:33:01] NOTICE[5692]: chan_dahdi.c:7490 ss_thread: Got event 18 (Ring Begin)...<br>
[Jul 14 08:33:03] NOTICE[5692]: chan_dahdi.c:7490 ss_thread: Got event 2 (Ring/Answered)...<br> -- Executing [s@DID_trunk_1:1] ExecIf("DAHDI/1-1", "1?SetCallerPres(unavailable)") in new stack<br> -- Auto fallthrough, channel 'DAHDI/1-1' status is 'UNKNOWN'<br>
-- Hungup 'DAHDI/1-1'<br> -- Starting simple switch on 'DAHDI/1-1'<br>[Jul 14 08:33:13] NOTICE[5697]: chan_dahdi.c:7490 ss_thread: Got event 18 (Ring Begin)...<br>[Jul 14 08:33:15] NOTICE[5697]: chan_dahdi.c:7490 ss_thread: Got event 2 (Ring/Answered)...<br>
-- Executing [s@DID_trunk_1:1] ExecIf("DAHDI/1-1", "1?SetCallerPres(unavailable)") in new stack<br> -- Auto fallthrough, channel 'DAHDI/1-1' status is 'UNKNOWN'<br> -- Hungup 'DAHDI/1-1'<br>
-- Starting simple switch on 'DAHDI/1-1'<br>[Jul 14 08:33:29] WARNING[5704]: chan_dahdi.c:7634 ss_thread: CallerID returned with error on channel 'DAHDI/1-1'<br> -- Executing [s@DID_trunk_1:1] ExecIf("DAHDI/1-1", "1?SetCallerPres(unavailable)") in new stack<br>
-- Auto fallthrough, channel 'DAHDI/1-1' status is 'UNKNOWN'<br> -- Hungup 'DAHDI/1-1'<br></div>
<div>It's says (Ring/Answered)... but it doesn't go to an extension or voice mail.</div>
<div> </div>
<div>I have incoming calling rules defined. I've tried directing the calls to a ring group, a voice mail and an extension. No joy.</div>
<div>This is running on openSuSE 11.1 fully patched and running asterisk16-1.6.0.9-73.18.</div>
<div>Asterisk GUI-version : SVN-branch-2.0-r4970</div>
<div> </div>
<div>Will you help me track this down?</div>
<div>Thanks</div>
<div> </div>
<div> </div>
<div><font face="Verdana">Bob Crandell</font></div>
<div><font face="Verdana">Assured Computing, Inc.</font></div>
<div><font face="Verdana">541-868-0331</font></div>
<div><font face="Verdana">ComputerBase USA</font></div>
<div><font face="Verdana">541-349-0404<br></font></div></div><br>_______________________________________________<br>--Bandwidth and Colocation Provided by <a href="http://www.api-digital.com--/" target="_blank">http://www.api-digital.com--</a><br>
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