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Calls dropping


collinsit
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Hi there, I have a CS410 unit and a couple Snom phones, a 320 and a 360. I have been having an issue with calls dropping if I am not talking on the call. You may wonder why I wouldn't be talking but I often do conference calls or phone seminars so the phone will be on speakerphone and in most cases my microphone will be muted. It seems to happen after about 5 minutes. This has been going on for a while with the 360 but normally the 320 worked fine. I had that problem today with the 360 again so I dialed into the call with the 320 and it was doing it too. After it did it a few times at about 5 minute intervals I tested and just before the call got to about 5 minutes I picked up the handset, unmuted the line and made a noise. I then put it back on speaker and the call went fine. If I did this before about the 5 minute mark the call would go fine but if I forgot it would disconnect again after 5 minutes.

 

I have the newest software release on the CS410 and the newest firmware on the Snom phones so I am not sure what the problem is.

 

Any help would be greatly appreciated.

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Hi there, I have a CS410 unit and a couple Snom phones, a 320 and a 360. I have been having an issue with calls dropping if I am not talking on the call. You may wonder why I wouldn't be talking but I often do conference calls or phone seminars so the phone will be on speakerphone and in most cases my microphone will be muted. It seems to happen after about 5 minutes. This has been going on for a while with the 360 but normally the 320 worked fine. I had that problem today with the 360 again so I dialed into the call with the 320 and it was doing it too. After it did it a few times at about 5 minute intervals I tested and just before the call got to about 5 minutes I picked up the handset, unmuted the line and made a noise. I then put it back on speaker and the call went fine. If I did this before about the 5 minute mark the call would go fine but if I forgot it would disconnect again after 5 minutes.

 

I have the newest software release on the CS410 and the newest firmware on the Snom phones so I am not sure what the problem is.

 

Any help would be greatly appreciated.

 

It seems that there is a problem with the phone's ability to keep the connection alive during mute. From the PBX perspective the phone is "dead" and that is why the PBX hangs up. What version of the phone are you on? Maybe check release notes of firmware updates if there is anything on this.

 

Dirty workaround is to change the global settings for one-way audio timeout. The parameter has the name timeout_connected and you can see the current value in the pbx.xml file.

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I checked the version on the phones and it looks like they were running 7.1.3. I updated them to 7.1.33 and the problem still kept happening. I then updated one of them to the newest version which is 7.3.7 and I am still dropping the calls. I couldn't find anything in any of their release notes about this problem. You said that I could edit the pbx.xml to change a setting for the one way audio setting. It doesn't sound like the ideal option but if I wanted to test making this change, how do I edit this file?

 

If you can think of any other options that would be appreciated too. I don't think it is a Snom firmware problem since I have now tried it with three different versions.

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I checked the version on the phones and it looks like they were running 7.1.3. I updated them to 7.1.33 and the problem still kept happening. I then updated one of them to the newest version which is 7.3.7 and I am still dropping the calls. I couldn't find anything in any of their release notes about this problem. You said that I could edit the pbx.xml to change a setting for the one way audio setting. It doesn't sound like the ideal option but if I wanted to test making this change, how do I edit this file?

 

If you can think of any other options that would be appreciated too. I don't think it is a Snom firmware problem since I have now tried it with three different versions.

 

Okay, I think we need to try that in our test lab as well, should not be too hard to reproduce.

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That would be great. I really need to get this fixed since I do a lot of calls this way. The way I have been testing this after the call I had yesterday is that i'll dial a land line and answer the call. I then mute the call on the pbx and within a few minutes the call will disconnect.

 

If you can give me any suggestions as soon as possible I would really appreciate it.

 

Thanks

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That would be great. I really need to get this fixed since I do a lot of calls this way. The way I have been testing this after the call I had yesterday is that i'll dial a land line and answer the call. I then mute the call on the pbx and within a few minutes the call will disconnect.

 

If you can give me any suggestions as soon as possible I would really appreciate it.

 

Okay, the phone really does not send RTP, even if "RTP Keepalive" is set to "on". At least in version 7.1.33.

 

But it does send RTCP. I think we can also use that as a indication that the call is alive. It might have problems with NAT, but I think it is reasonable to say that RTCP also is a sign that the UA is still connected.

 

What OS are you on? May we give you a image to try it out?

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Could you please attach a pcap trace that would capture the packets. Make the call, start capturing, then mute the phone and after about 15 seconds stop capturing and attach the pcap trace. On mute no RTP packets go out but we do send keep alive packets every 5 seconds and we want to see if that still happens and are they being ignored by the PBX and is it broken (highly unlikely in all versions). Thanks.

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I tried the new firmware version on the PBX and I am still having the problem. Like clockwork, after the call is connected and on mute for 3 minutes it disconnects. I will try to get a capture of the traffic tomorrow to see if there is anything that will help there.

 

Are you using the 3 minute demo key???

 

Had the same problem with testing here :P .

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I don't think I am using the demo key. I should have a licensed CS410 and did before. Upgrading the software wouldn't have affected my license would it? How would I know if my license is limited to 3 minutes?

 

The name of the key would be something like "3 Minute Demo". That string shows up under admin/status.

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No, I am running a permanent key. Under status it says:

 

License Status: Appliance Key

License Duration: Permanent

 

The calls don't seem to be disconnecting when I am talking on them, just when they are muted which doesn't sound like a license limitation.

 

(A little bit at wits end)

 

Is there something like a firewall in between? Maybe the NAT closes after 3 minutes if there is no refresh from the other side?

 

We were able to reproduce the problem in the lab (though the timeout was not 3 minutes), and after the change it worked nice.

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What change did you make to get past it in your test environment? There really isn't any setting in the firewall as I am forwarding a complete IP through to the PBX.

 

Well, most (all?) low-end firewalls don't have a setting for that, it is just hard coded. I think it is time to think about getting a Wireshark trace. The CS410 also has a packet capture tool on shell level (tcpdump, see http://linux.die.net/man/8/tcpdump). Then we can give a definitive answer on what is going on here.

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