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Features Broke After Asterisk Upgrade

Posted by caw on Thu, 08/19/2010

All-

In an effort to troubleshoot and fix some other issues, the Digium support tech, et al advised me to upgrade Asterisk. In this particular case, also at the recommendation of the Digium support tech I only upgraded from 1.6.0.26 to 1.6.0.28 with the intent of upgrading to 1.6.2.x this weekend.

I have not performed an Asterisk upgrade since I installed MTE so I believe I did not compile Asterisk properly.

Since I upgraded:
1) My CDRs are broke.
2) The ability to send faxes through the system is greatly diminished (does not work at all).
3) The conference rooms no longer seem to work.

I will admit that my customers do not use the conference rooms very often, so this may be completely unrelated, but something I need to get fixed quickly.

My upgrade was very generic: ./configure; make; make install

The packages I upgraded were: asterisk-1.6.0.28, asterisk-addons-1.6.2.1, dahdi-linux-complete-2.3.0.1+2.3.0, & libpri-1.4.11.3.

Any assistance pointing out where I may have screwed up is greatly appreciated.


Submitted by eeman on Thu, 08/19/2010 Permalink

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

Submitted by eeman on Thu, 08/19/2010 Permalink

btw for asterisk and asterisk-addons you should

make distclean
./configure
make menuselect
make
make install

Submitted by caw on Thu, 08/19/2010 Permalink

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.

Submitted by caw on Mon, 08/23/2010 Permalink

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?

Submitted by eeman on Mon, 08/23/2010 Permalink

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.

Submitted by caw on Fri, 08/27/2010 Permalink

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?

Submitted by eeman on Fri, 08/27/2010 Permalink

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

Submitted by caw on Fri, 08/27/2010 Permalink

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?

Submitted by chris on Fri, 08/27/2010 Permalink

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

Submitted by eeman on Fri, 08/27/2010 Permalink

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.

Submitted by chris on Fri, 08/27/2010 Permalink

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.

Submitted by caw on Fri, 08/27/2010 Permalink

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.