[asterisk-users] Moving User Agent To Remote Location

[Nick Khamis]

Hello Everyone,

Before getting into SIP and RTP traces, I wanted to clarify some of
the sip.conf settings that may to some seem redundant or have a
misconception with. I do apologize if this has been discussed time and
time again as I would imagine. If anything, this email would make
google search results that much stronger :).

With the UA local to my network I had tested two way audio, and now
with the phone outside of network, we have no way audio. Before
discussing NAT (which is enabled on the peer), and port forwarding
(which is setup on the remote location), I would like to make sure I
fully understand all the sip.conf settings. We are using Asterisk
realtime via sip_buddies, and the fields in question are:

(Enclosed in brackets are an example value for the setting)

* host (dynamic): No problem here. Just wanted to mention that it's
set as such....
* nat (yes): No problem here either....
* defaultuser (1003 at example.com): Does the "@example.com" have to
point to the UA (i.e., (1003 at ua-public-ip), or is it just a name type
field?
* fullcontact: What to put here for a UA that is running at a remote
location with dynamic external IP?
* ipaddr (ua-public-ip): I did try setting it to the public ip of the
UA, but is that really practical?
What if I don't know where the initial registration request is coming
from? I am guessing "host=dynamic" takes care of that.
* defaultip??
* dynamic: Should this be set to yes, or is "host=dynamic" sufficient?

The phone registers fine, and terminates a call through our providers.
Just no audio both ways, which would suggest something more that an
RTP issue which should at least have one way outgoing audio.

Things that have been attempted:
* Port forwarding to the phone
* Changing defaultuser to 1003 at ua-public-ip. This made our OpenSIPS
sip proxy through a fit.

Things I will attempt today:
Calling the UA extension from an extension here
SIP trace

Your help is greatly appreciated!!!

Nick.