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Meet-Me Conference with TP Conductor

conie saga
Level 1
Level 1

Hello all,

 

We have TP Conductor XC2.3 (virtual) TP Server 4.0 (virtual) CUCM 8.6 & End point SX20.

We have setup meet-me conference as per TelePresence-Conductor-Unified-CM-Deployment-Guide-XC2-3.

When we dial meet me (6499) number from SX20 its keep dialing & the call is not answered. When dial it shows dialing 6499@22.130.10.5. I tried call 6499 number from other ip phones also but its same.

I verified TP Server is registered into TP Conductor, checked the dial-plan in conductor its working fine.

I have verified configuration step by step but couldn't figure out the issue.

I have attached the screen shots & logs, can any body help please.

 

Regards,

2 Accepted Solutions

Accepted Solutions

Jonathan Robb
Cisco Employee
Cisco Employee

Hi,

 

I see the following in the Conductor Event log:

All conference bridges unusable or in 'Busy out' state

A few questions:

Is your vTPS showing 'enabled' now in the Conference bridge section?

Is there an Encryption feature key on the vTPS?

 

Could you start Conductor diagnostic log from Maintenance -> Diagnostics -> Diagnostic logging, run a test call, stop the log and post that log to the thread.

 

-Jonathan

View solution in original post

 

Hi,

 

Keep it as TCP between the UCM and Conductor. Conductor may reflect different UCM IP Addresses based upon the UCM Group settings (server order).

 

Make sure the SIP Trunk Security profile is configured the following way:

- Device Security Mode: Non Secure

- Incoming Transport Type: TCP+UDP

- Outgoing Transport Type: TCP

- incoming Port: 5060 (Make sure your Conductor SIP Trunk is also 5060)

 

Conductor Meet-Me / Rendezvous Setting

- Trunk Port - 5060

- Trunk Transport TCP

 

Conductor Conference Bridge  is enabled (as per Jonathan's post)

 

vTPS Settings

Network Services > SIP TLS is checked (as per Jonathan's post)

Keys (Configuration > Upgrade) for Activation, Encryption, and Screen Licenses are added/enabled.

 

View solution in original post

13 Replies 13

heathrw
Level 4
Level 4

Hi,

Can you please send details/screen shots of your conference alias information through?

 

Regards

Heath

Hi Heath,

I have the screen shot of dial-checkup result of conference alias from conductor for now.

Check the attachment please.

 

 

If the call could not be established I would expect to see a but more in the conductor event logs, example attached.

 

If you have added additional IP Addresses for your locations restart the conductor.Also I would make your conference alias something more like (64..)@.*

 

Do you get a ring back tone or fast busy tone when trying to dial? Reset all your sip trunks, route lists, groups, etc

 

Are you trying to run TLS or TCP between the UCM and Conductor?

 

 

 

 

 

Hi,

One thing i noticed in conductor Event log which is attached in first post file name " Conductor-eventlog_beta-anycast_2014-07-27_08_22_20.txt",

2014-07-27T05:21:10-03:00 conferencefactory.switchboard: Level="INFO" Event="A management request has been received." Command="device.query" Requester_(VCS/Unified_CM/client)_address="22.130.11.5" UTCTime="2014-07-27 08:21:10,714"

My CUCM ip address is 22.130.10.5 why in logs its showing 22.130.11.5 ? Device is registered to 22.130.10.5 and 22.130.11.5 is a subscriber.

I have two additional ip addresses in conductor 1 for Adhoc Conference & 2 for Meet Me conference.

I will try restart the conductor now & will update you.

I tried conference alias as 64..@ before but it didn't worked thats why i change it to .*@. After restart i will try modify conference alias again & will update you.

I am not getting ring back tone or fast busy tone, its just showing dialing 6499@22.130.10.5 & after 30sec to 50 sec call drops automatically.

In conductor under location settings for Meet-Me i have setup TCP as Trunk Transport Protocol (CUCM 8 & below should be TCP, this is written in documentation)

In CUCM under SIP Trunk Security Profile, i have selected TLS as Incoming & Outgoing Transport port. I tried configuring TCP+UDP as incoming transport port & UDP as outgoing transport port but it gives me some duplication error & I didnt remember what exactly was the error. So i left it on TLS.

Do you think this could be the reason ?

One step which i haven't done till yet is adding trust certificate between conductor & cucm. I suppose its not mandatory thats why i escape it for now.

Regards,

Hello Heath,

i tried by restarting conductor but the behavior is still same ...

Should i change CUCM configuration as TCP instead of TLS ?

Trusted Certificates are mandatory to be in CUCM & Conductor ?

Regards,

 

 

 

Hi,

 

Keep it as TCP between the UCM and Conductor. Conductor may reflect different UCM IP Addresses based upon the UCM Group settings (server order).

 

Make sure the SIP Trunk Security profile is configured the following way:

- Device Security Mode: Non Secure

- Incoming Transport Type: TCP+UDP

- Outgoing Transport Type: TCP

- incoming Port: 5060 (Make sure your Conductor SIP Trunk is also 5060)

 

Conductor Meet-Me / Rendezvous Setting

- Trunk Port - 5060

- Trunk Transport TCP

 

Conductor Conference Bridge  is enabled (as per Jonathan's post)

 

vTPS Settings

Network Services > SIP TLS is checked (as per Jonathan's post)

Keys (Configuration > Upgrade) for Activation, Encryption, and Screen Licenses are added/enabled.

 

Fine Heath,

I will change the sip security trunk profile & conductor meet-me setting by tomorrow as you suggested.

Conductor conference bridge is configured vTPS is added in it & shows status as active. I am reattaching the image for you showing the status of vTPS in Conductor.

 

Network Services > SIP TLS is checked .. Yes it is already checked.

Activation, Encryption and Screen licenses are already installed in vTPS. I will share the screen shots with you by tomorrow.

 

Regards

 

Gents,

Thank you so much for your valuable answers.

Heath i made changes on trunk as you suggested & it worked.

Now the problem is end points are disconnecting automatically after 20sec.

I have one SX 20 (ext 6497) in HO, one EX90 (ext 8300) in remote Branch1, one SX 20 (ext 4597) in remote Branch2.

After meet-me conference established, only first participant who initiated conference remain active in conference, rest two disconnect automatically after 20sec.

e.g.if conference initiated by ext 6497, then after 20sec 6497 will remain in conference and ext 8300 & ext 4597 will disconnect automatically.

In vTPS conference status screen the reason of disconnect showing "TimeOut".

We are using region with 64kbps (G722 & G711) in all locations include HO and we don't have any security device (like firewall or asa) between HO & Branches.

Attached are diagnostic logs & region setting.

Hello all,

Problem fixed.

I change conference resolution under conductor > conference template from 720 to 360 ....

After this no disconnection found.

Regards,
 

 

The telepresence server vm with 4 ports TS licence,L772D-UTJQA-XXJJC-BB9KK-ELLA4, needs at least 8 vCPU to enable 720p30 calls according to the document from http://www.cisco.com/c/dam/en/us/td/docs/telepresence/infrastructure/ts/install_guide/Cisco-TelePresence-Server-on-Virtual-Machine-Install-Guide-4-0-2-8.pdf, but your VM has only 4 vCPU. It could be the VMware resource that limited your concurrent call numbers. 

 

Thank Hong-Quan,

For put on light on cause.

Appreciated.

Regards

Jonathan Robb
Cisco Employee
Cisco Employee

Hi,

 

I see the following in the Conductor Event log:

All conference bridges unusable or in 'Busy out' state

A few questions:

Is your vTPS showing 'enabled' now in the Conference bridge section?

Is there an Encryption feature key on the vTPS?

 

Could you start Conductor diagnostic log from Maintenance -> Diagnostics -> Diagnostic logging, run a test call, stop the log and post that log to the thread.

 

-Jonathan

Hi jonathan,

Yes Conference Bridge is showing enable in TP Conductor.

Check the attachment.

Yes Encryption key is installed in vTPS.

I can capture the logs for you by tomorrow once i will be on client site.

Regards,