R.I.P. Wable: Fun While It Lasted

wardmundy

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Just a heads up that DNS resolution can be a challenge with Wable especially on the latest CentOS platforms. Here's how we finally fixed it.

In /etc/sysconfig/network-scripts/ifcfg-venet0, add PEERDNS=no and chattr +i /etc/sysconfig/network-scripts/ifcfg-venet0

In /etc/resolv.conf, add nameserver 8.8.8.8 AND nameserver 8.8.4.4 AND chattr +i /etc/resolv.conf

Also, remove postfix since SendMail is included: rpm -e postfix

Then reboot and be sure you can ping incrediblepbx.com.
 
Last edited:

wardmundy

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Pictured is one of the Wable routers with over 100gbps of connectivity. Need more for free? Get 2 additional CPUs/VPS, 4GB additional RAM, and 30GB additional SSD storage by clicking here: https://wable.com/powerboost/camp-fb8-23 (only works on Bundle #3 or higher). Click the link, sign up at wable, choose bundle 3 or higher, pay for your bundle, then come back here and click the powerboost link again, you will see your resources increase).

11903970_1602767573318607_5624484177811121764_n.png
 

wardmundy

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Looks like the logs for Asterisk aren't rotating either. We'll fix it in future builds, but here's the fix.

Create /etc/logrotate.d/asterisk:

Code:
/var/log/asterisk/*log {
  missingok
  rotate 5
  weekly
  create 0640 asterisk asterisk
  postrotate
      /usr/sbin/asterisk -rx 'logger reload' > /dev/null 2> /dev/null
  endscript
}
 
/var/log/asterisk/full {
  missingok
  rotate 8
  daily
  create 0640 asterisk asterisk
  postrotate
      /usr/sbin/asterisk -rx 'logger reload' > /dev/null 2> /dev/null
  endscript
}
 
/var/log/asterisk/cdr-csv/*csv {
  missingok
  rotate 5
  monthly
  create 0640 asterisk asterisk
}
 
/var/log/atftp.log {
  missingok
  rotate 5
  monthly
  create 0640 nobody nobody
}
 

isaacl

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Been running a system on a CentOS VM on Wable that seems to be working nicely - thanks again Ward! One question - I'm currently running Incredible PBX with the version 12 GUI, from this post: http://nerdvittles.com/?p=13599 Is there any reason to be upgrading to Asterisk 13, as that seems to now be the recommended version? And is there any easy way to do that, if that would make sense? Thanks a lot!
 

wardmundy

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Asterisk 13 is a slick product. We've barely touched the surface of what it can do. SIP is much improved with PJSIP so, yes, it's worth the move. You'll have to create a new platform using IncrediblePBX13.iso. Then use incrediblebackup on your old platform to get a snapshot of your setup. Then run incrediblerestore on the new platform to put the pieces back in place. Test, test, test before deleting your old platform.
 

wardmundy

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Just wanted to provide a success story from Wable. When the forum began to self-destruct with the (very) old version of the forum software last week, we were scrambling to find a site that we could move to if things really went South. The XenForo developers had pretty much bowed out once they looked at our error log. Not encouraging! In less than an hour, we were able to build a server platform and migrate everything to Wable. We then built a script that could recreate the PIAF Forum from a current backup image in less than 60 seconds. That gave us a platform to try updates and patches quickly. We actually tried seven different versions of XenForo and more than a dozen different approaches before we found an upgrade methodology that didn't end with a blank screen. And here we are. Once we tested it (twice) at Wable, we shut down the production site yesterday morning for less than 5 minutes and applied the same fixes. As much as we like the pricing of CloudAtCost, we never could have moved things there in a disaster much less using it as a platform for experimentation. It would have taken DAYS! Downloads from pbxinaflash.com to Wable travel at 30MB/sec. So... Wable is the new backup home for the PIAF Forum at $8 a month with 5 CPUs, 6GB RAM, and an 80GB disk. Pretty cheap insurance.
 
Last edited:

Dave Gray

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Hmmm, I have to eat a bit of crow here. I had stated up-thread, that knockd wasn't going to work on Wable (as you're in a container, not a true virtual machine.) I was wrong.

On Wable, you just need to point knockd to the correct interface. There's a 'fake' eth0 if you requested that when you built the host. (It keeps the pbx software happy.) But, the real work is on venet0:0 (at least on my Centos 6.6 install.) Add OPTIONS=" -i 'venet0:0' " to the /etc/sysconfig/knockd config file, and it works just fine.
 

isaacl

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Asterisk 13 is a slick product. We've barely touched the surface of what it can do. SIP is much improved with PJSIP so, yes, it's worth the move. You'll have to create a new platform using IncrediblePBX13.iso. Then use incrediblebackup on your old platform to get a snapshot of your setup. Then run incrediblerestore on the new platform to put the pieces back in place. Test, test, test before deleting your old platform.

Just saw that I missed this reply...

Any way of getting this to work with CentOS 6.x?
Is there an IncrediblePBX13 script that works with CentOS 6.x?
I just saw the ISO when I looked around...

Thanks!
 

isaacl

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Thanks Ward, assuming I'll have to re-configure my system from scratch, but it's probably worthwhile...
 

isaacl

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Will a FreePBX backup/restore from Asterisk 11 to Asterisk 13 work?
 

isaacl

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Did it, and I have a system up and running - am I better off re-doing it?
 

isaacl

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Really not interested in having to re-do everything manually, but I'd rather do it now than later, if I'm going to have to re-do it.
Would import the old CDRs and manually copy over old voicemails, but shouldn't take too long, if I have to fully re-do from scratch...
 

isaacl

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Redid everything.
Imported voicemail, recordings, and CDRs from backups, but re-did the configurations manually.
 

isaacl

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Biggest issue, just realized what was happening and fixed it:
Asterisk 13 (or that version of FreePBX) by default is set up with the pjsip driver listening on port 5060, so if you import extensions set to listen on port 5060, you have to either disable pjsip (Advanced Settings -> SIP Channel Driver), or change chan sip to listen on port 5060, and switch pjsip to listen on 5061, on the Asterisk SIP Settings -> Chan SIP/Chan PJSIP pages (links are on the right side).
Once you put Chan SIP back to listen on port 5060, everything should again be working correctly.
 

isaacl

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Not sure why all of this is getting posted here, but for the lack of knowing a better place to put it -
In the new user control panel, outgoing call history entries by default show the outgoing caller ID instead of showing the number called.
I found a FreePBX bug report here: http://issues.freepbx.org/browse/FREEPBX-9401
The issue was supposedly fixed (here: https://github.com/FreePBX/cdr/commit/920c722b345ee6829440258bba8811c9121737c7) by switching the "$call['text'] = htmlentities($call['clid']" code in 3 places under the "elseif($call['cnum']" lines to "$call['text'] = htmlentities($call['dst']" in the ucp/Cdr.class.php file.
This was fixed in version 13.0.16, but that doesn't help for what appears to be the current version on my end, which is version 12.0.24.
So in the version that I had, I found the 3 occurrences of "$call['text'] = htmlentities($call['clid']);" under "} elseif($call['cnum'] == $self) {", and replaced them with "$call['text'] = htmlentities($call['dst']);", which fixed the issue.
Gotta figure a better way to write that up, without the quotes, and I can post the full code if it would help anyone, but that's the gist of it...
 

w1ve

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Wable NY been down for more than 12 hours!!!
 

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