Sipura SPA-1001
The SPA-1001 features VoIP adapter functionality found in the SPA-1001 with the additional benefit of an integral connection for legacy telephone network “hop-on, hop-off” applications. SPA-1001 users will be able to leverage their broadband phone service connections more than ever by automatically routing local calls from cell phones and land lines to a VoIP service provider and vice versa.
A typical user calling from a land line or mobile phone will be able to reduce and even eliminate international and long distance telephone charges by first calling their SPA-1001 via a local phone number or by using a telephone connected directly to the unit. The advanced authentication and call routing intelligence programmed into the SPA-1001 will connect the caller via the Internet to the far end destination with security and ease. Using the SPA-1001 at the far end, calls can be answered immediately or further processed as a local call to any legacy land line or mobile phone allowed by the SPA-1001 dial plan.
If power is lost to the unit or the VoIP service is down, calls can be sent to a traditional carrier via the FXO interface.
NOTE: We have seen instances where installing the SPA-1001 behind a firewall which blocks ICMP packets causes problems with registration. Try turning off any ICMP blocking on your firewall.
STEP 1
You must first determine what IP address it received. To do this, you need to pick up the phone attached to the Line 1 jack and
dial: **** (four asterisks)
then dial: 110 #
and you will be told the IP address of your device
(e.g. 192.168.0.100)
STEP 2
Go to any browser equipped computer on your network and enter the address:
http://<IP ADDRESS>/
(where <IP ADDRESS> is replaced by the address that was given to you in STEP 1).
STEP 3
Click on the "Admin Login" button near the top right side of the screen, then click on the
"Line 1" tab.

STEP 4
You need to modify only a few parameters from the factory default. They are listed here:
Proxy: voip1.voicemailtel.com
User ID: Enter the user name provided by VoiceMailTel.
Password: Enter the password provided by VoiceMailTel.
Auth ID: Enter the user name provided by VoiceMailTel.
Register Expires:3600

STEP 5
Make calls!
NAT/Firewall Issues
If you get one-way audio, you are probably behind NAT. Make the following changes on LINE 1 (you have to click on advanced view to see these options) on the SIP menu;


If the phone fails to login, please take the time to double check your configuration as above. If everything appears to be correct, the problem may be your firewall.
- If your router/firewall supports DMZ, put your VoIP device in the DMZ area
- If you have an external firewall try opening SIP ports SIP signaling ports (UDP) = 5000 - 5500 RTP/RTCP ports (UDP) = 10000 - 30000