The core of the server that does our SIP manipulations is based on this "Service Creation Environment" from here:
www.sipdev.org It's a really neat tool that is highly customizable. Full manipulation of SIP headers etc. You could build a server with your custom call flow scripts, then stick it in the middle between sipX and your gateway.. On Thu, Jun 18, 2009 at 12:56 PM, Damian Dowling <[email protected]> wrote: > James R wrote: > > Sorry about that.. message got sent early.... > > > > so I was saying our gateway (actually SIP proxy) takes the INVITE and > > converts it to a REFER which sends it to our PSTN gateway, then the > > proxy sends a bye to sipX to get the agent & sipX out of the call > > path. This is still a work in progress for us and it's a very messy > > solution.. If there was any other way to get sipX out of the media > > path we would love it. > > > > At the end of the day we just need to scale the number of agents per > > ACD to 100 to 150. We're not sure how to get there. > > > > On Thu, Jun 18, 2009 at 12:34 PM, James R <[email protected] > > <mailto:[email protected]>> wrote: > > > > We have the same problem. We have an agent application that sends > > a REFER back to the sipX server, but sipX turns around and sends > > an INVITE to our gateway... so sipX remains in the call for the > > entire transfer duration. With our call flow, every call to our > > agents gets transferred out.. so this is a major problem for us. > > > > Currently we are doing development on our gateway to take the INVITE > > > > > > > James, > > We are in the same boat. We need to get 80 to 100 active agents, and > unfortunately we cannot change or alter our gateways. So unless there is > a work around in SipXecs we are up sh#t creek without a paddle. > > Is there anyway we can light a fire under this problem? > > Damian >
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