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 sip:firstname.lastname@example.org:32110
The xx.xxx.xxx.250 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.