goodchefro
join:2007-02-21
Macomb, IL
 ·Logonix Corporation
edit:
May 9th, @11:34PM
| Stewart,
I'm behind a NAT. As a matter of fact, the UDP port range specified by RTP Port Min and Max actually is already forwarded.
Before changing the TURN server from "0.0.0.0." to "relay.voice.yahoo.com" I had dial tone, but the calls would not complete. Only after changing to the latter the calls started to go through.
(Edit: forgot here to mention that I also added my external IP in the SIP tab)
As far as ringing goes, it makes no difference what I use in the DTMF Relay field (as I was asking earlier this morning) since neither the default value, nor RFC 2833 alleviated the no ring occurence.
I even changed the "Sticky 183" field to "yes", but to no avail.
I'll keep playing with it, maybe something different will do the trick.
Thanks. |
