Linksys SPA IP phone reserves some * and # usages and it also has a default dial plan. You can see the definition in the Linksys GUI interface (admin login, advanced section, under EXT x tab). You can add or remove the dial plan entries as needed. For example, if you like to pass *xx commands to your IPX from the IP phone, you need to add *xx to the plan; if you like to use IPX ATA mode dialing 4 digit extension numbers, you need to add #xxxx to the plan.
If you configured an internal route: destination prefix 2, length 3, translation 8002.Dialing 201 will be the same as dialing 800201. But this route also changes 234567 to 800234, where 234567 is a valid remote extension while 800234 is an invalid extension. In this case, calling 234567 would fail. The correct route definition should be: destination prefix 2XX$, length 3, translation 800R. The number 234567 will not match 2XX$ any more.
Before an IPX license expires, the IPX alerts the operator user 15 days in advance. After expiration of the IPX license, all IPX functions will stop working and the lights on the system would turn to brown color. The IPX user can only get on the IPX through local LAN port of IPX to do license upgrade.
Most likely your IPX is behind a NAT/firewall and at the same time the DMZ or port forwarding on the NAT/Firewall is not set up properly. The incoming VoIP signaling and voice stream cannot come through the NAT/firewall. See "how to setup port forwarding on router”.
If any call forwarding to voice email is configured, a SMTP email server has to be configured in "system>system setting>email server" page. Make sure the SMTP server is working properly and the account on the server is valid.
If you place the service code *9 before a destination number, the system will follow the rule to select proper path for the outbound calls. If you place service code *8, the outbound calls must go out through a local VoIP account.
It is because the IPX could not recognize the busy tone signal from local telephone network. An external caller hangs up a call, but the IPX thinks the conversation is still on-going and the IPX did not release the port.
You need to change the PSTN standard selection in "basic voice > region and language" and reboot the system. If the problem remains after a few PSTN calls, you can ask local telephone company for busy tone frequency parameters and configure them in "advanced voice > advanced parameters > busy tone". You should also send email to support[@]simton.com asking for help. The Simton technical support team might be able to discover them for your region.
In VoIP systems, routing decision cannot be made until all digits are collected. A VoIP system completes the digit collection when
(1) time to wait for next digit expires;
(2) digit '#' is received.
After the IPX got all digits, it sends the digits one by one to FWT. The FWT needs to collect the digits in the same way and thus might have another waiting period in the process if '#' is not sent at the end.
There are multiple ways to short the process:
(a) configure the key interval to be short time;
(b) user to add "#" all the time when dialing;
(c) define a call destination rule in which you can add an extra ‘#’ at the end.
Although you and your neighbor are in the same office environment, your extension and your neighbor's extension might be on the different IPX systems. The command *61 is only useful between extensions on the same IPX. It is because IPX is not able to know the status of remote extensions.
First, you need to change call destination rules (i.e. call routing rules). If the numbers you called are covered by some rules, the rules will be used. If the calls are allowed, you will need to further check the group setting. If needed, you may further check the authorization in the extension settings.