[Asterisk-Users] Optipoint 400 Standard SIP

[John Blackman]

Hi,

I'm kind of a newbie myself.  I've had similar problems and it can be
very frustrating. I did get them all resolved so I'll share some of what
I did in hopes that it will fix your issue.

To get some of my phones to work (Grandstream BT100) I had to add a line
"nat = yes" in my sip.conf under each phone config. I also had to set
the bindaddr, externip, and localnet (which is a network address not a
host address) in the general section of sip.conf.

I'll put a plug in for Grandstream here.  Their phones aren't nearly so
expensive as some others and they work very well.  Check to make sure
they have the features you need, if they do, I definitely recommend
them.

I believe the issues you are having are NAT related.  SIP uses one set
of rules for routing and RTP uses a different set of rules.  You may see
other things - like one way audio - in addition once you are getting the
config close.

If you have the equipment, one way to isolate NAT issues with
non-routeable addresses (192.168.x.x or 10.x.x.x) is to create a VPN
tunnel between the network where your server is located and the network
where your clients are located.  If the system works until you remove
the tunnel, you are definitely having NAT problems (the tunnel masks the
problem because it will actually send the non-routeable packets to the
other side).  My personal preference is IPSec, but PPTP or L2TP should
work fine for testing.

I keep reading everything I can.  The wiki is very helpful, even though
you have to search for a while to find some answers.  I also have pretty
good luck by searching in Google for examples of other people's files
(which an amazing number of people are kind enough to post).  I have
posted several issues to this bulletin board and I have gotten very good
answers that way too.

Don't give up.  Once you get your configuration correct, Asterisk works
amazingly well.  I prefer it over every commercial product I've seen.

Regards,



John