SPA502G w/ E3200 router - display yes, dial tone no, now what?

[fried_tomato]

I'm setting up a Cisco SPA502G phone and get the phone's display panel to work, but I can't get a dial tone.

The set up involved hooking an ethernet cable from my Cisco E3200 router to the phone's SW port, and then connecting another ethernet cable from the phobe's PC port to a computer.

I can get the internet just fine on the computer with this setup, so I know it's not an internet access problem.

My company's IT guy said the Cisco router was probably not auto-configuring. I went through the router's Advanced Settings and couldn't tell if it was or not. The Help link on the router didn't help.

So IT guy said to set up port forwarding on the router & gave me a list:

Port 5060 TCP & UDP

Port 5004 UDP

Port 10000 UDP (sipgate Stun service - usually 3478/9)

Ports 16348-32768 UDP

When I went to do this in the router's Advanced Settings, I found I had to pick whether the port was an external or internal one. IT guy said nothing about this. I set each of the single ports to both internal and external. (I had no idea what to do with the Stun service number, so I did nothing w/ the 3478/9 thing.) The Range of Ports page didn't have the internal/external option. *whew*

Alas, either I forwarded the ports incorrectly or the ports weren't the problem.

The IT guy disappeared - may be on vacation.

How caln I get a dial tone?

[nseto]

Dial tone occurs when the phone is registered to the sip proxy.  Did you populate the proxy, userID and password parameters with the settings provided by your service provider?  The parameters are located in the Ext tab.  Once registered, you will see the LED turn green and you will hear dial tone.

[JeffInAlberta]

According to Sipgate's Help Center, they do support STUN port 3478:

http://www.sipgate.com/faq/article/397/How_do_I_set_up_my_VoIP_device

Try using "Port Range Forwarding", or Port Range Triggering, instead of Single Port Forwarding.

Try setting the Start~End (or Trigger Range) Port Range for SIP as 5060 - 5060.

Try setting the RTP Start~End (or Trigger Range) range for RTP as 16384 - 16538.

Try setting the Stun Start~End (or Trigger Range) for Stun as 3478 - 3479.

You could have other configuration issues, but no way to know.

Try reviewing my blog post for configuring the SPA504G, which I myself use. 

http://voipdiy.blogspot.com/2011/11/configuring-cisco-spa504g-ip-phone-from.html

The SPA504G is a 4-line phone, but I expect the line settings are very similar to the SPA502G.

My examples for the SPA504G my reveal some settings that will be of help to your situation.

Just be sure to substitute your specific Sipgate settings in place of my CallCentric settings.

My CallCentric settings do not require NAT Settings to be enabled or Keep Alive.  However, I expect you may need to enable the NAT Mapping Enable to YES and NAT Keep Alive Enable also to YES, for Sipgate service.

Good Luck.

Jeff