Mike,
Some questions/things to check.
Is your server inside a firewall?
If so is it in a DMZ, with all the proper ports being forwarded?
Is the sip client your trying to invite on the same subnet as the * server?
Have you checked that the sip client is registered with the server?
(from the CLI )
Can you call the sip client from an extension that is on the * server?
Mike
Mike C. Fletcher wrote:
Simon P. Ditner wrote:
Hi Mike,
It looks correct.
Thanks, that at least lets me know I'm not doing something painfully
stupid (well, yet).
I've found that asterisk doesn't tell you much when a call fails to go
through -- what does ethereal/tcpdump see if you sniff port 5060
traffic?
Just the 6 outgoing SIP invites, nothing ever comes back. I'm
beginning to suspect it's a network/NAT/firewall issue, as the
outgoing SIP invites all look fine. Have to get something setup
outside the firewalls to see if the invites are actually coming in to
the remote client. Strangely, invites from the PSTN bridge are all
coming into this Asterisk instance fine, not sure why an accept
wouldn't show up.
With an simpler test, I'm showing that the asterisk server just
doesn't seem to be able to invite *anyone* into itself, even with a
simple Dial(SIP/testout/someextension), it's invitations just
disappear into the ether.
Continuing to poke at it. Thanks,
Mike
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