Fedora Core 7: OpenPBX

Following the theme of my previous post on Fedora Core 7, I really wanted to touch on my experience with OpenPBX.

Just for a bit of background, we’ve opted out of traditional telephone in favour of a SIP based VoIP, provided by a Montreal area re-seller named BabyTel. I had figured out and configured BabyTel to run under Asterisk 1.2 on my HP NetServ machine where it was happily doing its job answering the phone and taking messages etc… However, with all this swapping and upgrading, the HP NetServ is being retired.

So, looking through the FC7 package manager I see a listing for OpenPBX; I dig a bit deeper. OpenPBX is built on Asterisk and is somewhat different. Here’s what the authors claim make it different.

• OpenPBX.org has built-in STUN support for SIP NAT traversal
• OpenPBX.org uses SpanDSP which means better codecs and full T.38 fax over IP support
• OpenPBX.org uses Sqlite instead of Berkeley DB as its internal database
• OpenPBX.org has a universal jitterbuffer for use with any channel type
• OpenPBX.org uses POSIX timers which means there are no Zaptel timing dependencies
• OpenPBX.org has much faster and more efficient dialplan execution because it uses hashing
• OpenPBX.org evaluates correctness and integrity of configuration data (work in progress)
• OpenPBX.org runs well under a virtual machine such as Xen or VMware

Looking at this list, one assumes that the basics work and the folks at OpenPBX are diligently working on icing and candy. Sounds pretty good right? Prepackaged solution, easy to install, standard Asterisk config? What could go wrong?

So I embarked on the journey; OpenPBX installed via the FC7 Package Manager and then configuration began…

Having working extension.conf, sip.conf and voicemail.conf files, the theory is you should be able to drop these files in and basically start OpenPBX and life should go on. So, I grabbed the files off the old Asterisk server and copied them to /etc/openpbx.org/ on the new rig.

The copy was immediately followed by a service openpbx start.

Lo and behold, OpenPBX did start and did register the SIP channel with BabyTel. So I proceeded with a test call to the local talking clock (613-745-1576); great success! The call was answered and I was told the time.

Next and finally I moved onto voicemail. My callplan has the * key linked with VoiceMailMain; so to access any voicemail that was left, one would simply dial * and follow the prompts.

As soon as I tried the * key from a desktop phone; “click” the call was dropped. Looking through the logs the problem seems obvious; OpenPBX doesn’t ship with sound files.

Right around here is where OpenPBX begins to resemble a jet engine, in that it concurrently sucks and blows. I begin to search of a tarball of sound files for OpenPBX. OpenPBX seems to be a product without a real identity; searching for OpenPBX yields a sparse wiki and something much worse called CallWeaver. It would seem that OpenPBX is now something called CallWeaver. Take your pick, documentation for both packages is just plain terrible.

After searching and following countless 404’s the sound files were eventually fished out of svn://svn.openpbx.org/openpbx-sounds/trunk/sounds/ using svn. These files included a Make that ran each WAV through SOX for formatting presumably. Once complete the files were copied to the sound path specified in OpenPBX’s config files.

So, I try again. Dial voicemail using “*”. Click… hangup.

In the log again, this time its bemoaning the fact that it cannot find a codec to handle the translation of ulaw to g729. Which I suppose is fine as there is no g729 codec installed under OpenPBX. I deal with this by removing all references to g729 from SIP.conf to see if that helps.

Reloaded OpenPBX and tried again… A similar error this time; Unable to find a codec translation path.

mmmmkay. So now its bitching about codec paths that aren’t even configured. I figured a “reload” was enough, so I stopped and started OpenPBX. During the subsequent start-up it complained about the g729 codec. Loader.c: /usr/lib/openpbx.org/modules/codec_g729.so: undefined symbol: ast_log.

Assuming that an Asterisk compatible module would work in OpenPBX was a mistake. Based on the error one can only assume that in the process of “making” OpenPBX the authors did a wholesale “Find and Replace” on certain “asterisk” like words that resulted in breaking binary compatibility all over the place.

With a sour mouthful of OpenPBX tainting my palette, enoughs enough. YUM was called in to remove OpenPBX with extreme prejudice. I visited the Asterisk home page, downloaded the latest version of Asterisk and the supporting files and rolled-my-own build of Asterisk. I copied my configuration and g729 codec over and rigged up an init.d script based on the stock proftpd script.

First try, the talking clock worked.

Second try, the voicemail worked.

Given the 5 minutes of effort it took to download, compile, install and configure it took to get Asterisk going, I feel I’m way ahead of the 2 hours I’ve been tooling around with OpenPBX.

Why I can’t recommend OpenPBX to anyone:

1. Its a package looking for an identity. Not very clear what it is… or who is in charge of it.
2. The name change from OpenPBX to Callweaver further skews this identity.
3. Documentation is piss-poor. You might as well look up Asterisk how-to docs or use voip.info.org .
4. Binary compatibility with Asterisk is broken
5. Doesn’t seem to persist SIP registration between shutdowns. All the devices need to re-register. There may be a setting for this… but Asterisk does this out-of-the-box
6. Uses SQLite… but it doesn’t work nor is there any documentation to help.
7. Installation does not work out-of-the-box. There are a number of obscure downloads and builds required.

What does it have going for it?

1. Its a FC7 package. Maybe the worst one, but its easy to install but easy stops there!

Recommendation? Stay away. Get the real-deal from Asterisk. It works and has a future.