system toolbox » voip

About

• Cross-platform (Linux, FreeBSD, NetBSD, OpenBSD, MacOS X/Darwin, Open/Solaris)
• PSTN connectivities (FXS/FXO, ISDN, PRI, E1, T1)
• Multi-protocol voice over IP (H.323, IAX2, MGCP and SIP and SCCP)
• STUN support for SIP communications
• Encrypted calls with TCP/TLS and SRTP support for SIP.
• T.38 Fax over IP (pass-through, termination and gateway)
• IVR, Conferencing, Queues
• CallWeaver project page at voip-info.org has a very detailed features list

http://www.voip-info.org/wiki/view/CallWeaver

• 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.

• 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• Uses SQLite instead of the (no longer maintained) Berkeley DB1 engine as its internal database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, conversion script provided
• AGI has been renamed to OGI, Macro() has been renamed to Proc(), conversion script provided
• 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
• Support for PostgreSQL in RealTime mode

Licensing¶

The CallWeaver software is released under the GNU General Public License (GPL) version 2. There is no dual licensing regime. Contributors need to release their contributions under GPL2 compatible terms, but they are not required to sign over or disclaim any rights. Contributors also do not need to use the GPL for their contributions, any GPL2 compatible license will be sufficient. For example, some contributors have chosen to contribute code under MIT licensing or BSD licensing terms. For documentation, the GNU Free Documentation License (GFDL) version 1.2 is recommended but not required.

http://www.callweaver.org/wiki/CallWeaver+on+Ubuntu

Asterisk vs. CallWeaver (formerly OpenPBX) vs. others

Mon Jul 9 14:56:48 CDT 2007

I'm currently running Asterisk for a few phones, and I've had some
annoyances with it. Their SIP implementation is crap. Total crap.
It hardly adheres to the RFC, and my application layer gateway throws
away the SIP packets generated by it because they don't follow the
RFC. Which means, I cannot use the ALG to take care of NAT and
dynamic firewall policy assignment for RTP traffic. Dialplan
searching is SLOW when you have a huge dialplan. Phones sometimes
get deregistered and authentication fails on the first attempt to
make an outgoing call, but succeeds on the next attempt.

I'm running the latest 1.4 code. Also, a couple of days ago, a
remote code execution exploit was found with their crappy SIP
implementation. To top it all off, Digium's PRI hardware is total
junk. I know people that have had 4 DOA cards in a row.

So, I'm thinking about going with CallWeaver, which used to be called
OpenPBX. They are based on Asterisk 1.2, but have utilized 3 party
libs for codecs, SIP implementation, etc. They also have T38 fax
support built in, no crazy patching/hacks to make it work. It really
looks like a solid replacement for Asterisk.

Does anyone have experience with both? I'm going to use Sangoma
cards instead of the Digium ones. I need DUNDI support, which
appears to still be in CallWeaver. However, the one thing I'm
worried about is that it looks like it is missing the SLA (shared/
bridged line appearance) functionality that was added in the 1.4
train of Asterisk.

Also, if there are any other free/open source projects out there that
you think are worth mentioning, definitely let me know. Right now,
I'm leaning towards CallWeaver.

Thanks.

Woomera¶

The Woomera protocol is intended for LAN-only use (no Internet) and makes it possible to put your voice over ip system in one server/process and your pbx in another and connect them with a simple raw-linear-over-udp protocol. Woomera provides a socket interface through which GPL software can utilise non-GPL compliant VoIP protocol stacks.

A channel driver module "chan_woomera" is available for Asterisk and CallWeaver to interface the PBX with woomera.

Woomera currently only supports H323 but it should soon support the OPAL VOIP abstraction layer which will allow it to speak many other protocols. The number of protocols supported by the Woomera server is irrelevant to chan_woomera which will support anything Woomera supports because of it's thin-client-like design.

With woomera you can connect Asterisk, Freeswitch or CallWeaver to a H.323 server (openh323 code) which will do H.323 over IPv6. Apparently openh323 also has some SIP code in their CVS. If added to chan_woomera, you'd get SIP over IPv6 as well, or vendor-specify PRI/BRI (example: Sangoma A500 BRI), or via OPAL protocols like ZRTP, IAX2 or T.38 (fax).

Nenad Corbic from Sangoma at Asterisk-Tag.org - 1/4

Nenad Corbic from Sangoma at Asterisk-Tag.org - 2/4

http://www.youtube.com/watch?v=B-VP-5K2wFs&feature=related

http://wiki.sangoma.com/wanpipe-linux-asterisk-ss7

http://www.telephonydepot.com/Catalog/Sangoma-A500