SIP Server to 'Front' voice app server

I need an open source solution to carry out call 'hairpining' to keep the number of ports in use to a minimum on a voice application server.

I have been trying out Asterisk, but it's tough going with virtually no documentation or support available.

I basically need a system that will take an inbound call from a PSTN gateway, route it through to a SIP URI, then accept INVITE, REFER, REFERJOIN types of transfers.

I would like to have Consultation transfer, eg: standard 3 way PBX transfers.

I have also looked at SER/OpenSER but it is purely a SIP proxy and in my use case doesn't provide me much more than Asterisk and a lot less.

I'm currently exploring sipX.

Has anyone got any hints/tips/suggestions?

Thanks
Paul

Are you looking for something that does all that in one box/application? have you got a separate SIP<->PSTN gateway? How many ports does it need to manage? How many concurrent calls?

If something like Asterisk can fulfill your requirements, then there is also Callweaver and FreeSWITCH.

However, of all the open source app's that can do what you want, Asterisk has more documentation available than all of the others put together.

Hi.

Are you sure the speech server will support this, assuming the Genesys does, and you are connection to it via sip then by ports you mean channels, and I would have assumned like most speech apps it only counts an active channels when the speech app is active on the channel. As allready mentioned asterisk will have the most docs, you just have to know what to look for and experiment a lot. also check the SIP rfcs to see if what you want to do is possible then contact the speech app vendor to check they support it.

Ian