Stuart Ford wrote:
Seriously, this has to be the simplest NAT problem there is with
Asterisk. What's the secret? How do I learn the dark art? What am I
missing?
I'm guessing here, but the NAT'd grandstream does not have the correct external IP configured.
The phones are trying to establish a direct SIP to SIP connection, after SIP to SIP call is established asterisk tries to get out of the middle of the conversation. This decreases latency and save processing on the asterisk box. "canreinvite=no" sometimes helps this problem when asterisk is a sip client... don't know if it will have an effect here.
The thing to do is setup an extension with the Echo Application. Call that from each phone and see what happens. If it works for both phones you know the problem is a reinvite issue, if one phone or the other doesn't work it is a network or Nat config issue. No sense flailing about, try to reduce the problem space.
If your familiar with ethereal it can be used to snoop on the SIP connection.. SIP is human readable, so you might be able to learn something interesting.
But I really know almost nothing about this.
Mark Farver _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
