[asterisk-users] Moving User Agent To Remote Location

[Ishfaq Malik]

On Thu, 2013-01-03 at 12:47 -0500, Nick Khamis wrote:
> 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.
> 

Hi

Is your directmedia/canreinvite (depending on asterisk version) set to
no?

Regards

Ish

-- 
Ishfaq Malik <ish at pack-net.co.uk>
Department: VOIP Support
Company: Packnet Limited
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