On Sun, 2004-12-12 at 03:46, Eric Wieling aka ManxPower wrote: > Howard Lowndes wrote: > > When I make a call from a SIP phone to a speaking extension on *, such > > as one that speaks digits or similar, when I monitor * in very verbose > > mode I can see it running through the routine associated with the > > extension, but I am getting no RTP data stream back to the phone. > > > > Does the machine housing * need a sound card? > > Does it need OSS or ALSA modules installed? > > What actually generates the RTP data stream? > > > > You don't need a soundcard.
That's what I thought. > > Is Asterisk behind NAT? No, this is a local network. > If so look at localnet= and externip= in > sip.conf and look into portforwarding and rtp.conf. It won't need portforwarding being a local network. I might just check out rtp.conf. > Remember AUDIO on > SIP/H323/MGCP/SCCP are sent using the RTP protocol. Yes, I am aware of that, and that is what I am not getting back from *. > SIP is just a > signaling protocol. ...aware of that too. -- Howard. LANNet Computing Associates; Your Linux people <http://www.lannetlinux.com> ------------------------------------------ "When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft." ------------------------------------------ "Flatter government, not fatter government; Get rid of the Australian states." _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
