OpenPBX.org FAQ

Note: The OpenPBX.org project has been renamed to CallWeaver

Back to the __CallWeaver__ main entry on this site | Official CallWeaver website

I thought Asterisk was the only game in town. How come I never heard of you guys before?

There are a quite a few interesting open source telephony projects besides Asterisk, most notably and in alphabetical order Bayonne, CallWeaver, FreeSwitch, SipX and Yate. None of these can compete marketing wise with Asterisk. However, when it comes to the quality of the software, you will probably find that Asterisk is inferior to each and every one of its competitors. In any event, we think you should shop around and compare before you make a choice. In other words, don't judge a book by its cover nor by its position on the bestsellers list.


Why did you fork Asterisk?

  • We wanted community input and control, so that no single person or company can stop progress.
  • We wanted to be able to use the best libraries available. (e.g. SpanDSP, Unicall, Sofia SIP)
  • We wanted to avoid reinventing the wheel if it is not necessary.
  • We wanted to be free of limitations imposed by dual licensing.
  • We wanted to avoid commercial interests to interfere with quality of development.
  • We wanted more focus on reliability, generic solutions and cross-platform compatibility.
  • We wanted to allow everybody to participate and contribute without having to disclaim copyrights.
  • We wanted a level playing field, an independent project which does not compete with its customers.

What does CallWeaver have that Asterisk doesn't?

  • Official cross-platform support, currently Linux, FreeBSD, NetBSD and MacOS X
  • Conferencing without Zaptel hardware and without any kernel modules for timing
  • Built-in STUN support for SIP NAT traversal
  • Uses SpanDSP which means more efficient codecs and full T.38 fax over IP support
  • Uses SQLite instead of the (no longer maintained) Berkeley DB1 engine as its internal database
  • Allows CSRC entries in RTP for better compliance
  • A universal jitterbuffer for use with any channel type
  • Uses POSIX realtime extensions which means there are no Zaptel timing dependencies
  • Much faster and more efficient dialplan execution because it uses hashing
  • Much faster and more efficient extension matching engine
  • Variables and applications in extensions.conf are case sensitive, CallWeaver How to use the dialplan checking and conversion script
  • AGI has been renamed to OGI, Macro() has been renamed to Proc(), CallWeaver How to use the dialplan checking and conversion script
  • Evaluates correctness and integrity of configuration data (being introduced in stages, work in progress)
  • Provides conversion scripts to convert configuration files whenever changes are made to format or syntax
  • Runs well under a virtual machine such as Xen, VMware and VPS

What does Asterisk 1.4 have that CallWeaver 1.2 doesn't?

  • A vendor who has the last word on all decisions concerning the code base
  • A bugtracker where people get punished for reporting serious bugs which are difficult to fix
  • A licensing regime under which contributors who wish to contribute under GPL terms are rejected
  • Variable packetization
  • SNMP support

How reliable is the CallWeaver software?

Tests we carried out have indicated that RC releases of CallWeaver surpass the reliability of Asterisk 1.2.x releases and many users who have migrated from Asterisk to CallWeaver have confirmed this observation again and again.

If it's so reliable, then why are your releases only RC?

Our project charter puts equal importance on reliability and cross-platform compatibility. A full release will be made when cross-platform testing has been completed. The targeted platforms for the first release are Linux, FreeBSD, NetBSD, MacOSX/Darwin and Solaris.

Can I use my configuration files from Asterisk with CallWeaver?

Yes, you can. The dialplan (extensions.conf) has some minor changes though: Variables and application names are case sensitive, AGI has been renamed to OGI to avoid any potential trademark issues and Macro has been renamed to Proc because it was a misnomer. Don't worry, we have provided a script which CallWeaver How to use the dialplan checking and conversion script your dialplan accordingly after making a backup copy. We will continue to provide such checking and conversion scripts whenever we make any further changes in the future.

Can I use Asterisk plugin modules with CallWeaver?

Yes, you can. However, you will need to recompile the plugin after renaming some function names. For details refer to this HOWTO article.

Can I use the CDR processing/analysis utility/software I use now with Asterisk?

Yes, you can. We didn't touch the CDR engine at all. There is no difference between CallWeaver CDRs and Asterisk CDRs.

Can I use patented codecs such as G.723, G.729 or ILBC with CallWeaver?

Technically speaking, yes you can. There are open source implementations of G.723 and G.729. The ILBC codec is available for free personal use. However, for legal reasons, we cannot distribute these codecs together with CallWeaver. You will need to find, download and build them yourself. Please note that in most parts of the world you can only legally use G.723 and G.729 if you have paid royalites for the patents. For legal advice you should consult an attorney specialised in intellectual property law.

Can I use CallWeaver with telephony interface cards to connect to the PSTN?

Yes, you can. For a list of compatible hardware please see this section in the main wiki on this site.

On which operating systems does CallWeaver run?

At this time we have tested and run CallWeaver on various Linux distros with 2.6 kernels, FreeBSD 6.1 (using generic timers), FreeBSD 6.2 and 7 (using POSIX timers), NetBSD and Darwin/MacOSX 10.4. The architectures we tested on are x86, x86-64 and PPC.

Can I run CallWeaver in a virtual machine environment?

Yes, you can. Many of our developers and testers run CallWeaver under virtual machine environments to test on different operating systems. Most common are Xen and VMware. The CallWeaver demo server runs under VPS.

Can I use the FreePBX web GUI with CallWeaver?

Not right now, but the FreePBX developers have told us that they will have CallWeaver support when we make our first regular release.

My Asterisk doesn't scale. How about the CallWeaver software?

Yes, indeed, Asterisk doesn't scale. Some of it has to do with an inefficient dialplan execution and pattern matching engine. We have made significant improvements to that and intend to replace the engine altogether with an efficient implementation. Another cause is a very naive implementation of internal storage using linked lists which is extremely inefficient. We are working on a replacement using hashtable storage. Yet another cause are locking issues. We have fixed quite a few locking related bugs, but to solve these issues properly, some structural changes in the core are necessary and we are working on that, too.

RTP handling is still the same in asterisk and openpbx, so this still doesn't scale too well (max ~400 bridged calls per ~3GHz CPU)

I have serious problems with the Asterisk manager interface. Did you guys fix this yet?

No, not yet, but we have a design for a reliable, scalable and more secure replacement. This will become a priority after the first regular release.

I have troubles with queues in Asterisk. Did you guys fix those yet?

No, we haven't touched the queues implementation yet. We are considering app_icd as a replacement for app_queue as the default queue manager but no work has been carried out thus far. However, we know this is a trouble spot and we will work on this in the future.

I have many issues with Asterisk's SIP support. Did you ever think of using a better SIP stack?

Yes, we did. We think we should use the open source Sofia SIP stack from Nokia in CallWeaver to replace chan_sip in the future. Our friends at the Freeswitch project evaluated a large number of open source SIP stacks and they tell us that Sofia is the best thing since sliced bread.

Where can I get support for CallWeaver?

For free community based support you can use the mailing list or the #openpbx IRC channel. For commercial support you may ask any consultant providing support for Asterisk as they will be able to also support CallWeaver and many of them already support forked or otherwise modified versions of Asterisk anyway. We have received pledges from several consulting companies and telephony integrators that they will provide commercial support for CallWeaver starting with the first public release version. We will list those companies at that time.

Why did you rename all the ast_ prefixes in the source code and AGI to OGI in the dialplan?

We wanted to make sure that we stay clear of anything that could be construed as improper use of trademarks. For the same reason we chose to rename AGI (an acronym for Asterisk Gateway Interface) to OGI, the idea is "better safe than sorry". However, IAX is known to be safe to use as the USPTO lists it as an abandoned trademark.

I don't always understand the telephony jargon. Do you have a glossary?

Yes, we do. It's right here.

OK, you convinced me, I am going to switch, how do I get started?

Download a copy of the latest release tarball and follow the instructions in the "docs/osnotes" directory. Check the Downloads section in the main wiki entry of this site. If you need help, please meet us in the #callweaver IRC channel at irc.freenode.org.


If I wanted to contribute, do I need to sign any disclaimer to assign copyrights?

No, you don't. CallWeaver is a true open source project. No proprietary versions. No dual licensing. All you have to do is release your contributions under licensing terms which are compatible with the GNU General Public License (GPL) version 2. You keep exclusive rights to your code.

OK, I want to contribute, what should I do and how do I get started?

Since you are volunteering, it is not anyone's business to tell you what to do. You decide for yourself. However, we are happy to make suggestions. The best way to get started is to meet us in the #openpbx IRC channel at irc.freenode.org. You may also want to join our developer mailing list.

What was wrong with OpenPBX.org? Why did you rename the project?

There is another PBX software called "OpenPBX" by Voicetronix, an Australian vendor of telephony cards and there is a voicemail software for Windows called "OpenPBX VXM" by Chips & Dale Communications. This caused confusion with the initial name of this project (OpenPBX.org) and it was therefore decided that the project should be renamed to something which was not already in use elsewhere.


More info

__CallWeaver__ wiki on VoIP-Info.org | CallWeaver home page | CallWeaver wiki | __CallWeaver__ glossary


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Legal Disclaimer
The name of this project is "CallWeaver" and not "OpenPBX".
OpenPBX is an Australian common law trademark by Voicetronix.
CallWeaver is a US registered trademark for the CallWeaver project.
Asterisk and Digium are US trademarks of Digium Inc., used on this page for reference only, not to indicate endorsement.
Any other trademarks mentioned on this page are the property of their respective owners, they are used strictly for reference only.
Created by: STS, Last modification: Tue 12 of Jun, 2012 (02:04 UTC) by admin


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