you're probably right about the compile.. MANY needed libraries are missing so you probably have modules from different versions loaded.

yum groupinstall "Development Tools"
yum groupinstall "Development Libraries"

thats a good start.. you probably still need to check to see if other devel packages are missing

i hope you didnt build them in the order you listed them..

first you build dahdi, then libpri, then asterisk, then asterisk-addon

btw for asterisk and asterisk-addons you should

make distclean
./configure
make menuselect
make
make install

Thanks! I had all of the Devel Tools, but missed most of the Libs.

No, I did not install in that order, but the one you listed. I scraped and pasted in the order `ls` output them.

I'll try to recompile tonight and I'll update you after I do some testing.

I was finally able to upgrade the system to 1.6.2.11. My CDRs seem to be working again as do the conference rooms.

Faxing is still an issue. When I ran `make menuselect' I noticed app_fax was disabled as it needed spandsp. Guessing that since spandsp was not available, it wasn't used prior, so I just continued the upgrade.

Doing some additional work on the server this morning, I found the spandsp RPMs used during the initial build and spandsp5 is still installed. So now I know spandsp is needed for the fax pass-through to work, but my question is now, why was I unable to compile the app_fax into Asterisk when I ran the menuselect?

screw app_fax, grab the res_fax files from digium's website. 1.8 is switching to res_fax and we will not support legacy TxFax,RxFax and the improper flags that app_fax uses.

I have not rebuilt Asterisk with the res_fax yet. For some reason, fax pass-through seems to be working somewhat.

I found another item that seems to have been broken during the upgrade. The "Reload PBX" link does not actually reload Asterisk. I have to go into the CLI and run `module reload' for any changes I made through TL to actually take effect.

Was there another step I may have missed?

You need to copy over the cli_aliases.conf sample config file.

theres one in /usr/libexec/webmin/asterisk/configs/asterisk/ thats already set up for reload, restart and a few other things to be compatible.

I've copied cli_aliases.conf.sample into /etc/asterisk/cli_aliases.conf and changed ownership to asterisk.asterisk. When I tested, I found no change. Will I need to restart asterisk for this to take effect?

eeman-

I do not see a cli_aliases.conf in /usr/libexec/webmin/asterisk/configs/asterisk/

I just tested this. A module reload was sufficient to read in the new aliases. No restart was required. Also, the newest sample config file seems to have everything you would need except:
meetme=meetme list

if you dont see a sample cli_alises.conf in that dir its becaues you have an older pbx manager.

chris metme = meetme list is in that sample file of pbxmanager.

Erik,

Yes. I was saying that it seems that the only thing in the pbxmanager config file that is not in the most recent sample from Asterisk was that line. So, if someone didn't have the pbxmanager version, you could just add that one line to the Asterisk sample and be all set.

Erik- I'm running 6.0.1.81 which I upgraded to just a couple of months ago.

Thank you both for the help with this, `module reload' is working properly through the GUI now.

pbx manager is up to 6.1.1.5 now