More to it then just SIP registration ...

Hi: ....

So, on my TalkSwitch CVA 48 (VoIP PBX) ... which acts as the proxy/register, I have two 'built' in VoIP extensions (TalkSwitch allow a total of 8).  When a VoIP calls comes into one of those 8 'virtual' extensions, it 'translates' it over to a normal 'analog' extension (for example 299 is VoIP, but if it is called, it will ring my desktop analog phone which is extension 116).

Pretty simple stuff ...

So, I have a standard LinkSys SPA-841 IP phone (SIP phone) on my desk (actually sitting to right of my normal TS-400 extension 116 analog line).  This is ALL I did to the Linksys SPA-841:

1.  Reset it to factory defaults and rebooted
2.  Went to the web based page for the phone (to adjust its settings)
3.  Added ONLY the proxy/register IP address (our TalkSwitch) and a user ID (we have no authentication requirements at this time)
4.  Restarted the phone

BOOM ... I was in ... I was registered with the TS.  I used the User ID of: Linksys

I saw in the registration page of the TS, an entry for
The IP is our local NetGear SIP aware VPN/router.  

NOTE: xxxx = valid octet entries

I got dial tone!

When I tried to dial extension 299 (to make my other phone ring ... it returned with a BUSY signal).
When I tried to call from my extension 116 ... I didn't know how to 'dial' a SIP number that was a 'name', so I changed the User ID on the LinkSys from Linksys to 399.  When I tried to call 399, the TalkSwitch has built in prompts for errors, and I got one saying the extension is not available.

So, my questions are ...

1.  Do you have to do MORE to a SIP type phone other then enter the proxy and user ID?  I am getting dial tone with just those two entries

2.  In the past, I have been able to use the LinkSys SPA-841 and 'call' the two TS virtual extensions I mentioned above (250 and/or 299).  I just cannot re-call how it was setup.  Do you have any pointers?  250 is auto-answered by an attendant, and I set 299 to ring my personal extension (116).

3.  Finally, do you know of a good piece of programming that I could use to try and isolate down to a few choices where the VoIP (SIP/NAT/RTP) breakdown (and there-for anything else will not work).  It would be so cool to run that if I phone is not 'working' to see if it is a NAT, SIP, or RTP concern.
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Generally all you need to configure are the SIP server and the userid.
The proxy is something different and should not be needed for an internal phone.
pugdog_fanAuthor Commented:
Is that all you have to offer, or were you simply just looking to sort of add a tip, suggestion, or thread?

I didn't get much from your post.  Don't get me wrong, I apprecaite your contact and attempt to help, but it is vary vauge.

What is the 'difference' between these items:

SIP Server
Proxy Server
Outbound Proxy Server
Registration Server
You only get the following types of servers :-

SIP Server - This is the telephone system itself which the phones register themselves with. In your case I would expect this is what is refered to as the SIP server and the registration server.

STUN Proxy - This is a proxy server which phones can be configured to use. The proxy sits on the internet (often next to the SIP server) and accepts connections from the phones and passes the data on. The SIP protocol does not work well with NAT devices and the proxy detects of the client is behind a NAT device and tries to work around the problems they cause. See

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