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VVX411 intermittent with skype

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Occasional Visitor

VVX411 intermittent with skype

Hello,

The business I work for has sales offices in remote locations. To overcome connectivity, a 4G solution is deployed with external antenna and 4G router.

Download speeds average at 25Mbps / upload around 8Mbps.

We are trying to use 3 VVX411 logged into to Skype for business online with the same account ‘example@mybusiness.com’

All 3 phone should need to ring when a call comes in, this can work for around 10 test calls, then when left for around 30 mins, a call will come in and randomly 1 or 2 phones will decide not to ring.

The phones are still logged in and display the Skype account as available. There is a dial tone but it once a number is dialled it will just hang. Then a reboot is required to get it back online.

This is intermittent to which phone(s) does this.

Are there any config settings which are able to be changed to help with this?. Why do the phones not fully log out if it is a network issue?

We have checked the latency and believe this is low enough being round 34 - 80 and have tested in another scenario where the ping has reached 2000ms and this issue has not occurred.

Can anyone give any insight or suggestions please? Does anyone else use Skype for business over 4G ? Any other solutions Available?

Many thanks
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1 REPLY 1
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Polycom Employee & Community Manager

Re: VVX411 intermittent with skype

Hello @Benrich189,

welcome to the Polycom Community.

It is always useful to include the currently used UC Software version as issues experienced or a question asked may already be addressed in a newer release.

This also allows yourself and others to check against current software release notes, Administrator Guides or FAQ post’s.

The above is also stated in the "Must Read First" and is the absolute minimum requirement every new post should include. .

In addition providing us with this basic information gives Polycom an idea what Software Versions are used in the field and avoids wasting time trying to troubleshoot issues which have already been addressed.

Therefore the Polycom VoIP FAQ contains this post here:

Question: How can I find out my SIP or UC Software Version of my Phone?
Resolution: Please check here

 

To answer your actual question you may want to look at the following Parameter:

 

nat.keepalive.interval

The keep-alive interval in seconds. Sets the interval at which phones sends a keep-alive packet to the gateway/NAT device to keep the communication port open so that NAT can continue to function. If Null or 0, the phone does not send out keep-alive messages.
0 (default)
0 - 3600


Please ensure to provide some feedback if this reply has helped you so other users can profit from your experience.

Best Regards

Steffen Baier

Polycom Global Services

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Notice: This community forum is not an official Poly support resource, thus responses from Poly employees, partners, and customers alike are best-effort in attempts to share learned knowledge. If you need immediate and/or official assistance please open a service ticket through your proper support channels.
Please also ensure you always check the VoIP , Video Endpoint , Skype for Business , PSTN or RPM FAQ's
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