> I want to make sure that RTP is not going thru my asterisk. > > I read you should avoid in the dial commands: > m music while ringing > t,T transfer calls from caller and called party > > What else do I need to take care? > > remote phone ===> registered to local asterisk ===> calling remote gateway > > should have the RTP remote phone ===(RTP)==> calling remote gateway
That should be doable "if" the remote phone and remote gateway are truly accessible to each other on registered IP addresses (and not behind nat boxes or firewalls). If there is a nat box or firewall involved with either device, you might be able to get it to function, but it can be very difficult without knowing exactly how the nat boxes function and exactly how the sip phones operate. Part of that understanding is knowing exactly which udp ports the phones & gateway use for the negotiated rtp session. (E.g., Xten uses udp ports in the 8000 range, cisco's in the 16384-32766 range, asterisk in the 10000-20000 range, etc. There is no industry or rfc standard.) In addition, the phones and gateway will likely need some sort of stun support or static parameters to define the true external ip address. Not all phones support that. If a nat box is present, the _only_ way for you to diagnose rtp port negotiation problems is to use a packet sniffer at the phone and/or gateway locations as the two will attempt to negotiate a set of rtp ports on their own. Don't bother posting "it don't work" to the list without such traces as absolutely no one is going to be able to help. Also keep in mind that not all nat boxes operate the way that you think they should. Some will actually do port mapping and the OEM doesn't necessarily document that in their spec sheets. _______________________________________________ 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
