From: Preston Garrison <[EMAIL PROTECTED]>
Why not use a proxy, in theory it should pass through t.38 just fine since it doesn't get involved in the RTP streams at all. Is there a reason you need support DIRECTLy into asterisk.
Did you miss the part about the Zap channels?
I was thinking about adding proxy support to chan_sip for some time now, but have ended up making a stand alone proxy solution, and a universal system to authenticate users via asterisk or my proxy system, as well as DIDs etc. This way I can hand off my sip connections to asterisk if needed for voicemail etc, but most of the times my calls just pass through my proxy server only.
Does that sound remotely like any of the scenarios I described? _______________________________________________ Asterisk-Biz mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-biz
