Problems with SIP scheduling TMS > TP Server

[Nick Halbert-Lillyman]

I have a scheduling issue where TMS seems to be telling my TP server to dial out to pre-configured rooms/external endpoints at H323, despite the endpoints being only configured with a SIP URI and no H323 ID.

My external endpoints are added as "rooms" in TMS.  They have "allow bookings" and "allow inbound SIP URI dialling configured, but all other scheduling boxes unticked.  They have no H323 ID, E164 or gatekeeper configured (gatekeeper mode is set to "off".

When I schedule them in a conference, the connection settings come up as "H323-SIP" instead of just "SIP", so it tries dialling H323 first.  This is an issue because many these external endpoints are CTS-3000 units and if dialled as H323, TIP doesn't work they only connect with a single screen.

If I manually dial SIP from the TP Server, or if I add an external endpoint to the conference via TMS and specify SIP, it works fine. It's only an issue because TMS seems to want the MCU to try H323 first, despite the "room" configuration having no H323 options availible.

The reason this has just come up recently is because we have been using TP Server 2.2 in conjunction with CUBE and manually adding the endpoints to TP server as "legacy" CTS systems.  We are moving to TP Server 3.0 + VCS-E where TIP works automatically, but it doesn't seem to work if the call is interworked from H323 to SIP.

Any ideas how to get TMS to dial rooms as SIP only (or at least try SIP first)?

Versions:

TP Server 3.0(2.48)

TMS 14.2.2

VCS 7.2

[Paulo Souza · Accepted solution]

Hi Nick,

I had the same issue with TMS 14.2.2. I fixed that after configuring the field "Active SIP Server Address" into the Room/equipment configuration page. You can put any IP address on this field, no matter.

It seems TMS sets TP Server to call out by using SIP only when this field is configured on the Room/equipment configuration, If I leave this field blank, the result will be the same issue that you are having, a connection setup with H323 --> SIP.

Just to resume, these are the fields I have configured into room equipment configruation to have it working with SIP only:

SIP Mode: On

Active SIP Server Address: 10.10.10.10

SIP URI: ramal@domain.com

Gatekeeper Discovery: Off

Allow Booking: Check

Allow incoming SIP URI dialling: Check

Maximum IP bandwidth: 6000

You also must to set "Maximum IP bandwidth" field. If this field is "0", you will have a error message "No route possible between participant: TPserver and participant: Teste".

I hope this help.

Regards

Paulo Souza

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[Nick Halbert-Lillyman]

great, thanks Paulo!

[Paulo Souza]

You are welcome!

Thanks for the right answer.  =)

Paulo Souza

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