Polycom CX - Recent Questions, Troubleshooting & Support - Page 7

Hilsu198,

Have you tried restoring the unit back to the default setting? If not, please follow the steps below.

1. While viewing a line settings status, press SELECT to view the line settings options for that line.

2. Scroll down to RESTORE DEFAULT LINE SETTINGS?

3. Press SELECT again.

4. If you are sure you wish to restore all line settings, press SELECT again.

Hope it helps.

Thank you for using FixYa.

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.

In the NBX 100. Creat a group other then default. Then assing ip300 to the new group.

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.

I have a polycom soundpoint pro SE-225. when i either pick up the hand set to make a call OR press the speaker button to make a call i get a dial tone for a few seconds, then the dial tone jumps up and gets real loud for a few seconds and then goes completely away and will not let me make any calls. Any ideas what the heck is wrong??

I'm assuming you already have a voip account set up. sounds like you've not set the dtmf tones to the correct setting, usually set to rfc2833 and/or sip info - set in audio to no.

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.

This means that your phone will not dial a url number, i.e. sip://[email protected]. This is generally a network or configuration error, and I would specifically make sure the phone either has static DNS entries entered or is receiving DNS entries via DHCP.

I answered this question for another user:
http://www.fixya.com/support/t8134583-coult_not_using_boot_server_error

"
This occurs when the phone is set up to load it's configuration file from a ftp or tftp server and it cannot find the file or server location. If the phone is supposed to do this, please check that the ftp or tftp service is running on the specified server. The server name or ip is specified in the setup of the phone, available when the phone is first booting up. The default username is 'Polycom' and password is '456'. If you DO NOT want your phone to get it's configuration from a network server, enter the setup area and remove the server name or IP address from the appropriate section.
"

Please verify that you have the headset plugged into the correct port on the back of the phone. The handset and headset plugins are right beside each other, and it's easy to get them backwards. This also explains the sound issue, as a headset outputs amplified audio out, whereas a headset is analog and the phone does the amplification of the audio you speak.

This occurs when the phone is set up to load it's configuration file from a ftp or tftp server and it cannot find the file or server location. If the phone is supposed to do this, please check that the ftp or tftp service is running on the specified server. The server name or ip is specified in the setup of the phone, available when the phone is first booting up. The default username is 'Polycom' and password is '456'. If you DO NOT want your phone to get it's configuration from a network server, enter the setup area and remove the server name or IP address from the appropriate section.

This is usually a problem with the phone's configuration. Is this phone getting it's configuration from a network server or pbx? If so, find the configuration file for your specific extension and find this location. This example is for a phone whose extension is 2704


msg.mwi.2.subscribe=""
msg.mwi.2.callBackMode="registration" msg.mwi.2.callBack="" msg.mwi.3.subscribe="" msg.mwi.3.callBackMode="registration" msg.mwi.3.callBack="" msg.mwi.4.subscribe="" msg.mwi.4.callBackMode="registration" msg.mwi.4.callBack="" msg.mwi.5.subscribe="" msg.mwi.5.callBackMode="registration" msg.mwi.5.callBack="" msg.mwi.6.subscribe="" msg.mwi.6.callBackMode="registration" msg.mwi.6.callBack=""/>


If you are not using a network-based configuration, the phone's settings can be changed via the phone's internal web server. Open a web browser window and enter the phones' IP address, i.e. http://192.168.1.109. The default user is 'Polycom' and the pass '456'. Go to the links at the top for each line you are using and scroll to the bottom, you will see the voicemail settings there. Use the above info to fill in the correct info, save the settings and the phone will reboot.

For one line, you need both upstream and downstream of 90kbps for good quality.