Hi<br><br>We're looking at using OpenSER as a front-end to the Asterisk boxes we're using to provide a SIP VoIP service to distribute load and handle the RTP streams better. Our customers (who're all over the Internet) use the service primarily for calls to/from the PSTN with little, if any, SIP-to-SIP traffic. Having said that, we're providing PBX-like functions to some customers: voicemail, call groups, IVRs etc which we need to retain. Our PSTN connection is provided over SIP by a couple of different telcos; we just send/receive INVITES to/from them.
<br><br>How're people on this list implementing this kind of platform? OpenSER in front of Asterisk, Asterisk PSTN gateway? Something else? In particular, I'm having trouble seeing how DID calls will work. Let's take the following DIDs as examples and assume that inbound calls are sent to OpenSER:
<br><br>442071111111: straight mapping to a singe SIP account<br>442072222222: a group of 3 SIP accounts to ring simultaneously<br><br>The first case would work with users registering to either OpenSER or Asterisk, providing they shared a user database. In the second case, we would presumably have to forward the call from OpenSER to an Asterisk box to handle calling the 3 users, dealing with voicemail, etc.
<br><br>I suppose the crux of my problem is that I can't see how to get OpenSER to route calls based on the dialled number. Any ideas? I could use an external script and exec_dset but that seems clumsy; surely someone's done this before? This whole thing's a solved problem, right?
<br><br>How about outgoing calls? If a user puts a call through OpenSER to the PSTN it should be easy to check the usrloc DB to control access and allow the call out. Sound good?<br><br>Any hints on how to take this forward would be greatly appreciated. Once we get a plan in place the implementation should be reasonably straightforward (I hope!)
<br><br>Cheers,<br><br>MD<br>