A customer is experiencing an issue with auto-dialed participant using XC2.3 and TPS4.0
Bridges are generated as expected and dial in endpoints can join as participant with no problem.
We can see the Conductor trying to call and send the instruction to the TPS, but the TPS never dial out (no SIP INVITE).
The Auto-dialed participant is associated to a Meeting Template, with Conference Name Match (.*) and the address in URI format. (email@example.com)
They tried with both SIP mode: Call Direct as described in the PDF and using Trunk pointing to the VCS C IP. Neither option compete the call.
The meeting template and the Alias used to generate the bridge is in a very simple format (80000.*)
When the Endpoint call 800001, the search rule add the domain and send to the Conductor Zone (SIP, TLS, 5061)
The Conductor generates the bridge at the TPS (firstname.lastname@example.org) and the Endpoint is connected. OK
At the Conference detail in TPS web page, we can briefly see the auto-dial attempt/instruction with failed status.
Even in Use trunk mode, we can´t see any Search Attempt on VCS C.
This behavior appeared after the upgrade to 4.0
Any idea or tip to solve this?
Thanks in advance
No. A MXP endpoint dial to the Alias URI manually and the 1st participant creates the conference.
There is a TMS but it is not used to schedule the calls yet (both Conductor and TPS are added as a system in TMS).
How is your "Sip Trunk Settings for out-dail calls" section configured under the Rendezvous location? Is this still the same as it was before the upgrade?
This wasn´t touched.
Rendesvouz IP is the secondary local IP configured for Rendesvouz
Trunk IP is VCS C IP
Trunk port 5061
*The same parameters as SIP Trunk zone at VCS C.
PD: I´ll double check this info tomorrow. I´ll request remote access to check if someone changed...
under Conference Bridge, the TPS is added using IP, HTTPS and port 443.
the SIP port at the end is 5061
PD: Under Location, it is also configured to use TLS/5061, the same for SIP setting on TPS.
Ya me funciona mi problema era también un bug,la opcion de sip en el telepresence server debe estar con tls y yo la tenía con tcp.
Hi. I double checked the config and it seems OK.
The Rendesvouz IP are correct and the mode still configured as Call Direct as mentioned on Deployment Guide.
Any help on this?