On Thu, Oct 14, 2010 at 11:57:24AM -0500, Carlos Chavez wrote: > > But what ports did you open? Only sip or also the RTP ports? > > > I opened SIP and RTP, after that I put the server on the DMZ but I > still get no audio on the external phone. > > My problem is that we do not administer the customers network and the > just bought their brand new "super" router.
This router is completely broken, it will never function since the router has no way to relate the RTP stream from * to external to the correct external host unless it has a really good SIP helper (one that actually works instead of just breaking more stuff :) But I guess you should have atleast 1 audio leg working if you call an internal phone (with canreinvite=no). That isn't the case? This has to be a configuration error, you'll have to get in touch with the admin to setup NAT routing without rewriting the external adress/port. -- Daniel Tryba -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
