[VoIP] Registered but will not work !

[John R. Covert]

We figured out the problem.  Gawd, I hate NAT.

Ian has his Asterisk box behind a Draytek 2800VG router.  Asterisk
as a client to sipgate through NAT usually works pretty well.  But
here it wasn't.  So we started trying to debug the problem.

I got him to enter "sip debug peer sipgate", and we could see his
register commands, and the responses, and the sipgate webpage
showed him registered at his IP address, and if he changed the
/xxxxx it would show the change.  (Sipgate shows nothing special
ifn you use the 7D sipgate internal number after the slash, but
shows the full contact info if you use something else.)

But calls weren't getting to him.  Although the REGISTER messages
were going out, and the responses were coming back, there really
were no INVITE messages coming into the Asterisk.  He said he had
the ports open, so I tried placing SIP calls to him.  I was getting
responses back, but he wasn't seeing my incoming SIP messages.  For
a while we chalked this up to all the extra traffic generated by
having qualify=yes.  After about an hour and a quarter, I was
called to dinner, and we took a break for about half an hour.

While I was gone, Ian turned off the qualifies.  We could see
that there really were no INVITES coming in when I sent them,
even though SOMEONE was responding.  Then I noticed that whoever
was responding did not have a "User-Agent: Asterisk PBX" header.

Ahah!  The light turned on.  Someone else was running SIP on
port 5060.  The Draytek router has two FXS ports, and he had
put a registration for sipgate in, and I made sure he took it
out.  I asked him to take everything for any registration out
of the Draytek router, and he told me he had cleared it all out.

Still someone else answered.  OK.  So we change the bindport in
his asterisk sip.conf to 5062.  sip reload.  I send a call.  It
comes back with a "User-Agent: Asterisk PBX" header.  I call his
sipgate number.  The call goes through to his time service.

Now.  Who's on port 5060?  Whoever they are, they're still
responding when I send a call.  Finally, Ian explains that he
has the Draytek registered with Free World Dialup on one of
its FXS ports, and with another ITSP on the other.  (When I had
asked him to take all the SIP stuff out of the Draytek earlier,
he thought I only meant the sipgate stuff; I meant EVERYTHING!)
I ask him for his FWD number, and call that through FWD.  It
rings.  I then dial SIP/HisFWDnumber at HisIPaddress directly,
and it also rings.  Now we know who was grabbing his incoming
SIP calls.

This will probably work correctly for him.  He has some ATAs
located at some friends or other CNET members homes, and they'll
have to be configured to try to register with port 5062.  They
may or may not play well with NAT, but they'll at least not get
confused by the Dratek's SIP stuff.  Well, at least as long as
no one tries to use the same RTP ports...

And for the future, I figure he should eventually put FWD and his
ITSP into Asterisk, and use those FXS ports on the Draytek as
stations on the Asterisk.

/john