Tiki Suite
Component selection criteria
| FreeSwitch | Yate Server | Asterisk | sipXecs | |
| LDAP Support | Lookup, Possibility to create a bridge using mod_curl_xml | No (supports external file, so external support could be added) | LDAP RealTime ("broken" by design, doesn't work very well). Something can be hacked together with either RealTime MySQL or FreePBX's batch extensions& | Complete integration |
| License | BSD | GPL | Mixed (GPL + Proprietary) | LGPL |
| XMPP Support | Jingle, Conference advertising, Possible Presence | Jingle Only | Presence only (possible Jingle: http://www.venturevoip.com/news.php?rssid=2301 ) | Complete integration (uses Openfire) |
| LDAP Support | Lookup, Possibility to create a bridge using mod_curl_xml | No (supports external file, so external support could be added) | LDAP RealTime ("broken" by design, doesn't work very well). Something can be hacked together with either RealTime MySQL or FreePBX's batch extensions module. | Complete integration |
| Web UI | BlueBox (2600hz) [http://fusionpbx.com|FusionPBX] | YateAdmin (*very* basic) | FreePBX, Askozia | Yes, integrated |
| Distribution | BlueBox (CentOS), FusionPBX (Ubuntu 10.04) | FreeSentral (CentOS) | AsteriskNOW (CentOS), Askozia (Busybox), Trixbox (CentOS) | CentOS based |
| Non-Distribution Packages (Ubuntu) | Through PPA and manual installation (good docs on the wiki) | Through PPA | Up to date and complete | possible manual Ubuntu install |
| Language | C/C++ | C++ | C | C++ |
| Scripting | Configs in XML, Dial plan Applications may be written in C/C++, Java, .NET, Javascript/ECMAScript, Python, Perl, Ruby, PHP, Lua, etc | External Manager interface with libraries in PHP, Python and Perl | External modules can be written in many languages, but Asterisk has it's own dialplan language. However AGI scripts in many languages (Perl, PHP, BASH) can take over and the management interface is highly capable | ?? |
| sRTP/zRTP | Yes | No | No | Yes/Maybe, Freeswitch |
| Release cycle | Sporadic but frequent | ?? | Quite slow between majors, couple of years. Major usually take some years to move into common production usage after release | Slow |
| IRC | huge channel, FusionPBX has a med-sized channel also | small chanel | huge channel for both asterisk and freepbx, mostly useful | small channel |
| Backing | Baracuda Networks (turn-key IP PBX hardware), FusionPBX is a one man show (with external contributors). The main developer offers installation and service to fund development. | ?? | Backed commercially by Digium (Freepbx by SIPStation) | SIPFoundry Foundation and Ezuce service company, had support from Nortel and others in the past |
| DB Backend | Default SQLite, ODBC, MySQL, Postgres available | Mysql/Postgres | Mysql and others | Postgres only |
| Provisioning | Available | ?? | Available | Available |
| Documentation | Extensive | Smallish wiki | Extensive | Smallish wiki |
| Training/Learning Curve | Not so many FS admins available, however the online community is quite large | ?? | Lots of people already familiar with asterisk | Mostly everything is down through the Web UI |
| Other notes | Extremely flexible dialplan, Fusion uses nginx by default (lightweight)... FusionPBX dev's states he has ldap syncing high on his list (from the freenode channel) not real time solution but a start | Flexible dialplan | Implements it's own SIP stack, but uses FreeSwitch for other things (MoH, Conference) relies on OpenFire for XMPP. So far it is unclear how flexible the dialplan is. | |
| SAAS | Widely available | ??/No | Widely available | ??/No |
| Conclusion | Despite the fairly active core of solutions for VoIP including Asterisk, FreeSwitch, Yate and sipXecs; sipXecs is the only one of those that offers full integration with LDAP and XMPP (Jingle, presence and IM). However, it lacks flexibility at the VoIP le | |||
| Tiki | ||||
| FusionPBX BlueBox (2600hz) | ||||
