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I'm able to call people from the phone and they can hear whatever I say, but I hear nothing whatsoever.

When I first pickup the phone I'm able to hear the dial tone and I can hear the numbers as they are being pressed so it isn't a problem with the handset.

I've checked to make sure there isn't anything setup for a headset or anything like that, but I can't figure out what the problem is.

Anybody have any ideas?

Edit: Found somebody else with a similar problem and it ended up being that the firewall was letting traffic out but not back in, which is why the other person could hear them, but they couldn't hear the others. Will update this once I take a look.

To the guy who asked why I would ask the forum...the answer is because it was likely related to an IT issue (i.e. a firewall issue).

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Could you do me a favour please? could you explain how you think we might be able to help with this, as I'm at something of a loss to be honest. – Chopper3 Jul 29 '10 at 20:23
Because lots of sys admins work with VoIP systems and may have experienced similar issues. – user49032 Jul 29 '10 at 20:42
Do you have support with Polycom? Have you called them? – DanBig Jul 29 '10 at 20:52
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You should post your answer as an answer. And your question should have been framed in more of a sysadmin manner. It looks completely like an end-user question. – Dennis Williamson Jul 29 '10 at 22:06
@Chopper3, the problem is most likely that some networking device is not properly passing the SIP/H.323 packets. – Zoredache Jul 29 '10 at 22:23
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What do you see when you go to Menu→Status→Diagnostics→Media Statistics? (This is the path on a 430. It may be different on the 550.) Specifically, what does "Rx Packets" show? If it's zero then there is likely a network problem keeping RTP from making it back to your phone. If it's nonzero then there is likely some other problem.

AFAIK, tone and DTMF are generated by the handset so they're not a valid test of network connectivity.

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It shows 4-5 Rx packets so I guess that isn't the issue...hmmm – user49032 Jul 30 '10 at 15:36
Unless your most recent call was really short then that probably is the issue. Depending on the codec you're using you should see about 30 to 50 packets per second inbound and outbound. As Zoredache pointed out, it sounds like there's a firewall or a similar packet-mangling device blocking the RTP stream coming back to your handset. – Gerald Combs Jul 30 '10 at 17:15
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This is almost always related to firewalling. Keep in mind that both H.323 and SIP are very complex protocols. Data is sent as many separate channels.

If your network is also doing NAT you are going to have to either add a gatekeeper or proxy or update your endpoint configuration. Polycom devices do have a configuration option you can set to make them aware for a NAT.

If an endpoint doesn't have any configuration options to deal with NAT or limit the ports so you can allow them through the firewall then you will need to add a device to your network to make things work. On an H.323 network you need to setup a gatekeeper or application level proxy to get the traffic to work (I use gnugk). For SIP this is called the a session border controller.

For an IP video connection you will typically have at least the following separate connections. With the exception of the control channel any one of the other channels could be blocked by the packet filter and the call could remain open.

  • Control channel
  • Site A audio
  • Site B audio
  • Site A video
  • Site B video
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