I have a problem that I cant seem to find in any forums. So here I am, wanted to know about numbers that are not in service (dialing from a polycom phone) when I run into a number that is NIS it gives me a funny double ring then within 10-15 seconds the call is routed to another my incoming calls (as if i called my own company) does anyone know what could be causing that. any assistance would help. thanks guys -Alex

SOURCE: just installing polycom 501 phones with nextiva

Ohh this mean, you did not register this extension on your phone system. Create this extension on your phone system. Be sure that password given on Phonesystem is also configured on phone too.

SOURCE: Polycom Soundpoint IP 501 - URL call is disabled

This in it's most basic description means that the phone is unable to register with the SIP server, most likely for lack of internet connectivity or a firewall blocking the packets.

If you've verified connectivity and opened the proper ports and the phone fails to initialize and still getting “URL Call Disabled” or all of the above parameters are correct, update the firmware for the router. This can usually be done within the router configuration. Then reboot in order, the cable modem or DSL, the router or switch, then the phone. This may solve the problem.

If still having the same issue, confirm in the router configuration for settings such as SIP proxy, gateway, or application layer or any configuration setting for SIP or VOIP. If any of these do exist, make sure they are turned off as these may only confuse or adversely affect many implementations.

SOURCE: Making a Polycom SoundPoint 501 work on an Avaya IPO

As you have probably found, SIP phones are not supported until R5, which just came out.

SOURCE: one of our Polycom 550 phones keeps randomly

It is not your T1. The T1 only controls the internet, not your local LAN. How old is the phone? It could be a bad phone....What system is it installed on?

SOURCE: I use polycom Sound Point

This sounds like an RTP stream issue. While SIP runs on port 5060 by default, the actual audio is run on much higher ports, 10000 and up. If your pbx system is not on a public address and you are forwarding traffic to your pbx via a router, check the RTP port range in your pbx and make sure you are forwarding those ports correctly. By putting the call on hold, you stop sending out audio traffic restart sending audio when you pick the call back up. By forwarding the entire RTP range to your pbx, this should fix your issue.