Maxx Lobo
2006-Jul-20 19:16 UTC
[asterisk-users] Asterisk dead-air issues with Digium TE110P and IVR/meetme/internal directory-
I've developed an issue I was hoping someone on this list might be able to shed some light on. I have two identical asterisk servers: CentOS 4.3, single TE110P card. I've also got a SIP trunk configured with TelaSIP for in/out calling. Both boxes have been running the same versions of Asterisk, Zaptel, etc. In short, no difference between the two. One box is the primary system, all of the SIP clients on the network associate to it. I'll call this one PBX01. The other box, PBX02, is a test platform to install new releases on, and to make sure everything works as before before any upgrades take place (and new bugs get introduced ;-)) The plan was to get the system up and running using a VoIP trunk, then cut the T1 line over to the Digium TE110P card and keep the VoIP trunk as a backup for emergencies. Originally both boxes were running A@H 2.7, and there were no issues. I upgraded to A@H 2.8, and got hit by the spinlock.h kernel issue as a result of a yum update. No big deal, ran the fixes, everything seemed to work okay. VoIP trunk still in place. There were no errors with zaptel, and the module for the TE110P was detected on boot. Absolutely no asterisk errors either. Then I noticed that any voice menu-related items were broken - IVR menu was broken when outside callers called the VoIP trunk number, Meetme conferencing calls were also 'broken' in that there was nothing but silence when I called a working conference number, and internal directory calls to 7777 were also receiving dead air. I tried a number of things, none of which worked - so on a whim, I removed the TE110P. Amazingly, everything worked fine when the box restarted. Meetme conferencing, external calls, internal directory, everything. Thinking that the card may have developed hardware issues, I tried the TE110P from the standby server (PBX02). The same issues returned right away. So then I tried swapping servers, to make sure there weren't some hardware issues cropping up. Same result again - worked fine without the TE110P, did not work with it. Now I tried upgrading the test box to Trixbox 1.1, and upgrading the zaptel to 1.2.7 from source. Same issues all over again. I've seen the suggestions on Kennonsoft <http://www.kennonsoft.org/2006/07/trixbox-v10-quickie-how-to.htm> about the meetme conferencing directory permissions, and I've made all of those changes. It didn't work before the changes or after - same problem all along. The modules are getting loaded fine as evidenced by lsmod: -------- [root@pbx01 asterisk]# lsmod Module Size Used by wcte11xp 30496 0 zaptel 196740 3 wcte11xp crc_ccitt 6081 1 zaptel -------- No asterisk errors in the logs, either. No build errors when rebuilding zaptel-1.2.7. I've tried both kernel 2.6.9.34-0.1 and 2.6.9.34-0.2 (currently running the latter on both servers). I can post the full logs if necessary. The servers had sound cards in them, I removed these and made sure they were removed from both modprobe.conf and by kudzu. I didn't think this was the issue, but it was sound related and I am grasping at straws. Please tell me I'm missing something basic and what I can do to fix it. Without the TE110P cards I can't cut things over as I intended to, and I'm all out of ideas on this. Thanks- --Maxx
Maxx Lobo
2006-Jul-20 20:00 UTC
[asterisk-users] Asterisk dead-air issues with Digium TE110P and IVR/meetme/internal directory-
An update: I've found that I can leave the TE110P card in the server, unload the module and issue an 'amportal restart' - this brings the IVR/meetme/internal directory voice prompts all back again. So it looks like the issue is directly related to the TE110P module (wcte11xp) in kernel 2.6.9.34-0.2 with CentOS 4.3. Anyone else experience this issue or have any suggestions based on this new information? Thanks- --Maxx Maxx wrote:> I've developed an issue I was hoping someone on this list might be able > to shed some light on. > > I have two identical asterisk servers: CentOS 4.3, single TE110P card. > I've also got a SIP trunk configured with TelaSIP for in/out calling. > Both boxes have been running the same versions of Asterisk, Zaptel, etc. > In short, no difference between the two. One box is the primary system, > all of the SIP clients on the network associate to it. I'll call this > one PBX01. The other box, PBX02, is a test platform to install new > releases on, and to make sure everything works as before before any > upgrades take place (and new bugs get introduced ;-)) > > The plan was to get the system up and running using a VoIP trunk, then > cut the T1 line over to the Digium TE110P card and keep the VoIP trunk > as a backup for emergencies. > > Originally both boxes were running A@H 2.7, and there were no issues. I > upgraded to A@H 2.8, and got hit by the spinlock.h kernel issue as a > result of a yum update. No big deal, ran the fixes, everything seemed to > work okay. VoIP trunk still in place. There were no errors with zaptel, > and the module for the TE110P was detected on boot. Absolutely no > asterisk errors either. > > Then I noticed that any voice menu-related items were broken - IVR menu > was broken when outside callers called the VoIP trunk number, Meetme > conferencing calls were also 'broken' in that there was nothing but > silence when I called a working conference number, and internal > directory calls to 7777 were also receiving dead air. > > I tried a number of things, none of which worked - so on a whim, I > removed the TE110P. Amazingly, everything worked fine when the box > restarted. Meetme conferencing, external calls, internal directory, > everything. > > Thinking that the card may have developed hardware issues, I tried the > TE110P from the standby server (PBX02). The same issues returned right > away. > So then I tried swapping servers, to make sure there weren't some > hardware issues cropping up. Same result again - worked fine without the > TE110P, did not work with it. > Now I tried upgrading the test box to Trixbox 1.1, and upgrading the > zaptel to 1.2.7 from source. Same issues all over again. > > I've seen the suggestions on Kennonsoft > <http://www.kennonsoft.org/2006/07/trixbox-v10-quickie-how-to.htm> about > the meetme conferencing directory permissions, and I've made all of > those changes. It didn't work before the changes or after - same problem > all along. > > The modules are getting loaded fine as evidenced by lsmod: > -------- > [root@pbx01 asterisk]# lsmod > Module Size Used by > wcte11xp 30496 0 > zaptel 196740 3 wcte11xp > crc_ccitt 6081 1 zaptel > -------- > No asterisk errors in the logs, either. No build errors when rebuilding > zaptel-1.2.7. I've tried both kernel 2.6.9.34-0.1 and 2.6.9.34-0.2 > (currently running the latter on both servers). I can post the full logs > if necessary. > > The servers had sound cards in them, I removed these and made sure they > were removed from both modprobe.conf and by kudzu. I didn't think this > was the issue, but it was sound related and I am grasping at straws. > > Please tell me I'm missing something basic and what I can do to fix it. > Without the TE110P cards I can't cut things over as I intended to, and > I'm all out of ideas on this. > > Thanks- > > --Maxx > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users