The answer might be interesting to sipx users who really want a UC environment is this could be made to work.
Basically, I think what is needed is an 'asterisk bridge gateway' with t38 handler. In our testing, we found that we could call voice or send a fax to the same DID so long as the asterisk box had hylafax/nvfax installed and recognizing fax tones. We had to create a separate extension for each user who would be receiving faxes but the system worked perfectly. When faxes would come in, the server would intercept, convert, email the fax to the user. The one thing we were trying to do which we were not successful at was identifying the call as a fax call and then forwarding it to another asterisk box so that it could do the conversion and not use up resources on the initial asterisk box. We never found a way of doing this. So, I'm now wondering; PRI to SIP gateway providing SIP trunks to PBX servers | Call is sent to asterisk box | Asterisk acts as a gateway, processes fax if fax, forwards call to sipxecs if voice Thoughts? 1: I'm thinking since the asterisk box is acting as the media point, voice quality would suffer if it also does fax processing. 2: If there's no way of forwarding the IAX2 fax to a separate server, see 1. 3: If this is possible, maybe a very basic asterisk setup acting as a router? Sorry, really badly need to try and figure this out because UC really is UC and having to use separate DID's isn't. Mike _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users sipXecs IP PBX -- http://www.sipfoundry.org/
