Topic: OGM Stop playing until reboot [Comments: 5]
abongard

Thu, 12/01/2011 - 12:58 | OGM Stop playing until reboot

We have been tracking this for almost 2 weeks and I cant figure it out...

Basically random all OGM, VM Playback, and all other outgoing recording stops playing....

everything else works...

If you place an incoming call you hear no IVR but if you type say 1 for sales and that goes to a ring group then those phones ring and you can have a normal conversation.

Any ideas would be helpful...

Asterisk 1.6.0.6
MTE 6.1.1.11

Any ideas would be helpful as we cant see anything and nothing changed in the last few weeks except maybe added a tenant.

Regards,
Andrew

mattdarnell

Thu, 12/01/2011 - 21:07 | Reboot or restart asterisk?

Reboot or restart asterisk?

abongard

Thu, 12/01/2011 - 21:30 | OGM Stops Playing

We have tried to just restart Asterisk but that does not seem to work....!

We have had to do a full server reboot.

Has not happened since Sunday morning but happened twice today already...

abongard

Thu, 12/01/2011 - 21:45 | OGM Stops Playing

I am wondering since the files are all in wav and the played file is slin they must convert on the fly, could this app have just crashed? or forced to crash?

eeman

Fri, 12/02/2011 - 14:05 | well for starters 1.6.0.6 is

well for starters 1.6.0.6 is not supported. update to a supported, current, version of asterisk and see if the problem persists. Its a waste of our time and resources to look into a specific build of asterisk that was always considered beta release and was released nearly 3yrs ago. download and compile something like 1.8.0.8-rc4 and try testing again. make sure, before you start, that you remove the old-ass version of spandsp and replace it with the latest version, and that you download compile libresample beforehand.

1. download latest spandsp
2. delete old spandsp
3. compile/install latest spandsp
4. load libraries (check to see if /usr/local/lib is part of your ldconfig system, if not make it so)
5. download libresample via ' svn co http://svn.digium.com/svn/thirdparty/libresample/trunk libresample/ '
6. compile/install libresample and load libraries
7. download/compile asterisk 1.8.0.8-rc4 .. make sure you select the first 3 add-on modules when doing the make menuselect
8. delete existing modules (rm -rf /usr/lib/asterisk/modules/ )
9. install the compile asterisk
10. restart asterisk

you might want to go ahead and take the opportunity to upgrade DAHDI to the latest build before conducting these steps as there has been much improvements for timing source in the dahdi code.

Erik Smith
CTO
BluegrassNet Voice
dCAP
Thirdlane Support by BluegrassNet Voice
eeman at bluegrassnetvoice dot com

abongard

Thu, 02/02/2012 - 20:38 | Solved

Basically the issue was the RTC Battery which was causing interrupt issues and once replaced we have not had any issues...

I assume If we had used a hardware line card like a Sangoma this would provide timing not the RTC on the motherboard...

All is good now