[asterisk-users] Moving User Agent To Remote Location

[Danny Nicholas]

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Nick Khamis
Sent: Thursday, January 03, 2013 11:47 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Moving User Agent To Remote Location

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.

I'm going to vote for the RTP issue.  If you are establishing a call but
have no audio, you are getting the 5060 port, but not the 10000-20000 range
that RTP normally expects. A "better" practice is to allocate 4 ports per
line you expect to use in rtp.conf (10000-20000 would allow 2500 lines; much
more that most folks need and more "holes" to monitor).