Back when I was a bit newer with Asterisk, I was setting up a rather large office and had a number of issues getting things going.  To make a long story short, I learned that Asterisk is a terrible SIP router.  Many of you probably already know this – if you have an Asterisk server that only hands off calls to other servers based on dialplan rules, you will eventually choke the server as if it was processing all the calls itself since it won’t route the SIP traffic without also processing the audio stream, which is taxing even when it’s just passthru.  How I wish for an option in SIP.CONF like “siprouter=yes” where you can use it as a SIP router.  This would save a lot of time – there is already code for dealing with a database, extensions.conf is creating in a language I know, and I would only be dealing with one piece of software.  How nice.  Anyone want to set that up for me?

Since that isn’t really an option, I have the need for a SIP routing package that is fairly powerful.  I’ve started to look at OpenSIPS, though the learning curve seems pretty steep and so far my other job functions (Linux admin, EMC CX3 SAN management) makes it hard to get to these other projects as quickly as I’d like.  Currently we’re using a windows-based SIP router that functions fine for large pattern-matching (555XXXX to one server, 666XXXX to another…), but I want more granularity and the ability to use a database so updates can be automated.  I’ll update regarding my progress as I go along and we’ll see what happens.