TelePresence Endpoints Registered with CUCM - PSTN Dialling

Simon Battye · ‎08-09-2016

Hi,

With TelePresence Endpoints registered with CUCM we are seeing users failing to make PSTN calls because they aren't physically specifying the call as audio (64kbps) when dialling.

In this scenario they have to use a prefix in front of the PSTN number, is there a way in which we can configure the endpoint to recognise a prefix locally and automatically select the call to be audio, making it simpler for the users?

Thanks, Simon

Dennis Mink · ‎08-09-2016

Simon,

I dont fully understand the issue, but video capable endpoints, when making audio only calls across the pSTN, will negotiate the call as audio, based on the Region settings in CUCM. There is absolutely no need for users to explicitly define the audio rate/codec. Even if your telepresence endpoints send video capabilites across to the PSTN (and they most likely will), video will not get established anyway.

does that answer your question?

cheers

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Simon Battye · ‎08-10-2016

Dennis,

Thanks for getting back to me, i'm not seeing this behaviour with these endpoints but yes - what you say makes sense. When they dial the prefix and PSTN number, the call rings but disconnects as soon as you answer; if they make the same call but specify audio the call works OK.

The endpoints are using CE Software, i'll go over the regions again to see if there are any issues.

Thanks, Simon

Shefali Sharma · ‎08-14-2016

Hi Simon,

If it is required for you to add a prefix before making a PSTN call, then you must be having such a Dial Plan on your CUCM.

To verify if your call is being routed outside the CUCM with and/or without the prefix, then please refer the following:

Cisco Unified Serviceability -> Tools -> Dialed Number Analysis -> Analyzer

Here, you will have to add the Calling Party number and other details. Refer to the following image:

If it shows that Result of the Call Analysis is "RouteThePattern" without the prefix, then it means the CUCM is routing the pattern; however, there is some call routing issue on your gateway.

Could you please inform me that which gateway is being used in your call flow?

Best Regards,

Shefali Sharma

Simon Battye · ‎08-23-2016

Hi All,

I was able to resolve the issue.

We were seeing intermittent call failures when dialling out via the PSTN using TelePresence Endpoints.

The endpoint was dialling 9 followed by a PSTN number, calls specified as audio (64kbps) worked every time but calls that used the default video bandwidth (1152kbps in this instance) would intermittently work depending on destination network carrier, calls to some mobile phones worked and calls to other mobile phones would ring, connect then drop straight away.

I can only presume that this is a limitation of the network carrier not being able handle/negotiate calls that offer a bandwidth above 64kbps.

To resolve this i configured no bandwidth for video and telepresence on the endpoints region pointing towards the ITSP SIP Trunk, this resulted in the endpoint offering 64kbps only for outbound PSTN calls.

Thanks, Simon