Polycom IP - Recent Questions, Troubleshooting & Support - Page 3

Go to support on the new phones manufacturers web site and you can download an owners manual, which will tell you how to re-program the phones extension. murf427

Are you using this as a stand alone unit or configured for an Asterisk server? You say you got a new modem, so are you using the same router? If not, you may need to open some ports using port forwarding rules. How that is done depends on the type of router.

1st. Get the IP address of the phone. and type that into browser.
*- default user: Polycom / password: 456

*- Under menu SIP, type in your voip provider
*-Under menu LINES type in user info and "secret" should be your password.

Make sure your ports are open (SIP is usually 5060) you can change it in that same config screen.

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.