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Silent Calls being Cutoff


phsean
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It seems to us that outgoing calls that are silent for two minutes at a time are being cutoff.

 

forum wisdom seems to think that this can be resolved by turning off silence suppression, but as far as I can tell it's already set that way (for the handset, which is the issue).

 

The polycom_sip.xml in my html directory reads:

 

<NS voice.ns.hs.enable="0" voice.ns.hs.signalAttn="-6" voice.ns.hs.silenceAttn="-9" voice.ns.hd.enable="0" voice.ns.hd.signalAttn="0" voice.ns.hd.silenceAttn="0" voice.ns.hf.enable="1" voice.ns.hf.signalAttn="-6" voice.ns.hf.silenceAttn="-9" voice.ns.hf.IP_4000.enable="1" voice.ns.hf.IP_4000.signalAttn="-6" voice.ns.hf.IP_4000.silenceAttn="-9"/>

 

We're a health care organization and I'm in IT -- not a good combination for trying to avoid being on hold. Would appreciate a solution, thanks!

 

We're using Polycom IP 430's on version 2.1.10.2474.

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What firmware version are on you with the polycoms? You can download the latest here, and just put the sip.ld and bootrom.ld files in the tftp folder of PBXnSIP

 

http://www.polycom.com/emea/en/support/voi...oint_ip430.html

 

3.0.3 rev B seems to fine for me. Be aware though that you'll need to update your polycom_sip.xml and polycom_phone.xml files if you've crafted them by hand. I'm happy to share mine, but they include a couple of bits specific to us which you might want to change.

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The first thing they will tell you is upgrade to 2.2.2 on the Polycoms and try it , start there , I see this alot ,

 

I see alot of posts (and my own customers) experiencing something to do with muteed calls not sending RTP packets and then conference bridges will boot them alot ,

 

if you can upload an ethereal trace for support of an exapmle call being disco'd you will get better results ,

 

yori

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Our current Polycom SIP version is 2.1.0.2708, the bootrom version is 3.2.2.0019.

 

This is bringing back memories... I tried to upgrade the SIP application about 6-8 months ago without a lot of luck because we do have a few customizations in there, too. I ended up giving in.

 

Kristan, if you wouldn't mind sending me your config files, I could use WinMerge to compare them to the stock ones and see what you did to get it to work and start from there, that would be very helpful. An e-mail address that'll reach me is registrant -at- 3ipc -dot- com.

 

Thank you both - now it's down the rabbit hole...

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I do see an RTP Timeout in the logfile viewer when the call is cutoff:

[5] 2008/10/03 09:36:23: RTP Timeout on 9336d5f6@pbx#29825

 

here's another:

[5] 2008/10/03 09:44:03: RTP Timeout on 05a67596@pbx#6807

 

I just configured e-mail on the logging page, but it doesn't want to send me the RTP notification timeout via e-mail via either my Exchange or Exim mail server. Maybe I have something configured incorrectly (but the same settings in the domain CDR section work for sending a CDR report), or maybe it's something with pbxnsip 2.1.10.2474.

 

But that RTP Timeout you refer to is there in the logfile viewer.

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I do see an RTP Timeout in the logfile viewer when the call is cutoff:

[5] 2008/10/03 09:36:23: RTP Timeout on 9336d5f6@pbx#29825

 

Ok, that makes it clear that the PBX disconnects the call because it did not get any media any more. Did you use the PnP configuration for the phones? Seems strange that the phone does not send any keep-alive traffic (not even silence indicators).

 

Maybe you can do a Wireshark trace when the situation happens to see if the phone really keeps completely quiet.

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Did you use the PnP configuration for the phones?

 

(not sure of the nomenclature. I think that this constitutes Plug n Play)

 

What we do to provision a phone:

Add MAC to a new extension in PBXnSIP.

On Polycom: Server Type is TFTP and the Server Address is <pbxnsip dns>

 

Then we have the generic config files in the html directory of pbxnsip that pbxnsip translates into configs for each phone (copies stored in generated directory).

 

I will work on getting a wireshark trace together as well.

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It seems the problem is not between the phone (192.168.11.50) and the PBX. It seems that the problem is on the other side of the call. The phone sends media all the time, but the PBX is only sending keep-alive packets. That looks like the PSTN termination has a problem with media.

 

BTW try to put the version 2.2.2 of the Polycom into the tftp directory, maybe there is a problem with the firmware.

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You are right.

 

We put 2.2.2 on the polycom and had the same result.

 

But when we called extension to extension and put the phones on mute, we ticked above three minutes of silence without an issue.

 

Next step seems to be to load a test box with pbxnsip so that we can do a wireshark trace on an isolated call and see what is being sent between pbxnsip and the quintum pstn equipment.

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Next step seems to be to load a test box with pbxnsip so that we can do a wireshark trace on an isolated call and see what is being sent between pbxnsip and the quintum pstn equipment.

 

Just don't use the 3-minute demo key :( ...

 

Wireshark is pretty good in filtering out conversations. Maybe you can get just a huge Wireshark file on the ITSP trunk and then filter for the SIP and RTP traffic. Maybe be more efficient that setting up a second server.

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:) to the demo key comment.

 

I was interested in seeing the new version layout anyway, so I loaded up my machine with 3.0.1, registered a phone, started up wireshark, and waited for the cutoff... which never came.

 

I'm running the same version bootrom.ld and sip.ld (now 2.2.2) on the Polycom, same config files, and the trunk settings are the same on both the Quintum and PBXnSIP.

 

Maybe something changed in the way that PBXnSIP handles the trunk somewhere between pbxnsip version 2.1.10 and 3.0.1 (most likely an understatement). I'm interested to load our production server with the new version and see what happens.

 

Thank you!

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