<br><br><div class="gmail_quote">On Fri, Jul 10, 2009 at 6:44 PM, Wayne <span dir="ltr"><<a href="mailto:Wayne@planetwayne.com">Wayne@planetwayne.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
Sorry to bump my own message - but had a mail server problem so don't<br>
know if I missed any replys :(<br>
Ta<br>
<font color="#888888">Wayne.<br>
</font><div><div></div><div class="h5"><br>
<br>
<br>
Wayne wrote:<br>
> Hi all,<br>
> I've just built a new installation of CentOS release 5.3 (Final) and<br>
> have installed both<br>
> <<a href="http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-1.6.1.1.tar.gz" target="_blank">http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-1.6.1.1.tar.gz</a>>Asterisk<br>
> 1.6.1.1 and subsequently Asterisk 1.6.0.10 (thinking that I was maybe<br>
> trying to be too cutting edge) on a Dell PowerEdge sc440 server (nothing<br>
> complex - Pentium Dual core 2ghz - 1gb ram - 70gb sata hd).<br>
><br>
> The setup at this point is real simple with one Cisco 7960 phone<br>
> registering with Asterisk using Skinny.<br>
><br>
> I'm finding that simple things as pressing any of the buttons on the<br>
> phone is enough to cause Asterisk to randomly restart from a<br>
> segmentation fault.<br>
><br>
> I've tried this with 1.6.1.1 and, after recompiling and replacing, 1.6.0.10.<br>
><br>
> I followed<br>
> <a href="http://www.voip-info.org/wiki/view/CentOS+5.2+and+Asterisk+1.6.x+installation" target="_blank">http://www.voip-info.org/wiki/view/CentOS+5.2+and+Asterisk+1.6.x+installation</a><br>
> as a basis for installation leaving out things I didnt want to set up<br>
> (odbc / web admin ).<br>
><br>
> The only thing that didn't seem to go too well was the setup Dahdi<br>
> (dahdi-linux-2.2.0.1). Although I can do a 'make' and 'make install',<br>
> 'make config' didnt work and there are no etc/dahdi/ directory to change<br>
> any config files (as suggested by the guide). This may not be related<br>
> but just in case I thought I would mention it.<br>
><br>
><br>
> This is from the console after pressing the 'speaker' button a couple of<br>
> times.<br>
><br>
> /usr/sbin/safe_asterisk: line 146: 21513 Segmentation fault (core<br>
> dumped) nice -n $PRIORITY ${ASTSBINDIR}/asterisk -f ${CLIARGS}<br>
> ${ASTARGS} >&/dev/${TTY} < /dev/${TTY}<br>
> Asterisk ended with exit status 139<br>
> Asterisk exited on signal EXITSTATUS-128.<br>
> Automatically restarting Asterisk.<br>
><br>
><br>
> If I don't use the phone, Asterisk will stay running.<br>
> I can dial the 1000 test extension along with the 500 inter-asterisk<br>
> test, these seem to work as expected as long as I dial the number and<br>
> hit 'dial' on the phone rather than selecting the line and trying to<br>
> dial each digit in turn. If I try that then at some random point (but<br>
> not always) Asterisk will fault.<br>
><br>
> The firmware version on the phone is 7.2 to which I've had this phone<br>
> and several others running off a 1.2 setup for years (using<br>
> chan_skinny?) but thought it time to update Asterisk.<br>
><br>
><br>
> Anyone have any pointers please on what to check next?<br>
><br>
> Thanks,<br>
> Wayne<br>
><br>
><br>
</div></div></blockquote><div><br>If you are set on "beta" then read no further then the next line.<br><br>File a bug report with a core dump.<br><br>OK opinion time.<br><br>Your server is more than adequate. <br>
<br>For my tastes, you are beyond bleeding edge on the Asterisk front.<br><br>Simply my opinion but if this is going to be a "real" "production" "server" or something you want to use reliably then I would suggest.<br>
<br>1.4.Latest Zaptel <br></div></div>1.4.19 Asterisk (if infact that is the last version that had chan_zap and not DAHDI)<br>1.4.Current LibPRI <br>-- <br>Thanks,<br>Steve Totaro <br>+18887771888 (Toll Free)<br>+12409381212 (Cell)<br>
+12024369784 (Skype)<br>