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[FAQ] Hybrid Registration Feature

Polycom Employee & Community Manager

[FAQ] Hybrid Registration Feature

Polycom Trio’s supported Hybrid Registration since UC Software 5.5.2 and the VVX Business Media Phones added this feature parity with UC Software 5.8.0

 

The Hybrid Registration feature include the following limitations:

  • You cannot merge local conferences on Skype for Business registration lines. You can merge local conferences on Open SIP registration lines.

  • You cannot bridge Skype for Business and Open SIP registration lines.

  • Local merging of two point-to-point calls made using two different lines between two Polycom Trio systems is not supported.

  • Only call transfers between different SIP registrations with the same SIP call servers is
    supported. Call transfer between SIP registrations on different SIP call servers is not supported.

  • To avoid unexpected phone behaviour, do not use the same user name for multiple registrations.
    Use similar but not identical user names. For example, use:
    reg.
    1.address="John.Smith@company.com"
    and
    reg.2.address="J.Smith@business.com".

  • Transport Layer Security (TLS) encryption of Real-time Transport Protocol (RTP) media for
    secure communication in hybrid Open SIP registrations is not supported.

 

When a VVX or Trio phone is in Skype for Business base profile usually as a factory default only 1 line can be used.

 

Adding the following configuration:

 

reg.limit="3"

 

Will allow up to 3 registrations (VVX201 can only support 2).

 

The first line must always be Skype for Business and Line 2 or Line 3 can be openSIP or for Polycom Trio Zoom and/or BlueJeans.

 

The 2nd or 3rd line cannot be another Skype for Business account.

 

If the Trio is in generic mode then up to 3 Lines can be used for SIP.

 

VVX:

 

Hybrid_VVX01.pngHybrid_VVX_02.png

 

Trio:

 

Hybrid_Trio01.pngHybrid_Trio_02.png

 

NOTE:The below assumes the Trio is already configured with two lines and simply illustrates what configuration would be required in order to automatically route the calls to the correct Line.

 

Example 1:

 

  • Hybrid Configuration for all digits to be send to openSIP Lines and Skype SIPURIs and calendar join goes to line 1

 

<?xml version="1.0" encoding="utf-8" standalone="yes"?>
<!--Description: Sends all dialed digits to Opensip on line 2 while allowing Skype meetings and URL dialing on line 1-->
<!--Example is for 2 Lines-->
<PHONE_CONFIG>
	<ALL
		dialplan.1.digitmap.timeOut="8"
		dialplan.1.lyncDigitmap.timeOut="10"
		dialplan.2.digitmap="x.T"
		dialplan.2.digitmap.timeOut="4"
		reg.1.mergeServerDigitMapLocally="1"
		reg.limit="2"
		dialplan.digitmap.lineSwitching.enable="1"
	/>
</PHONE_CONFIG>

Example 2:

 

<?xml version="1.0" encoding="utf-8" standalone="yes"?>
<!--Description: Sends all dialed digits to Opensip on line 2--> 
<!--while allowing 11 digit numbers starting with 02 or 07 on line 1--> <!--Example is for 2 Lines--> <PHONE_CONFIG> <ALL dialplan.1.digitmap.timeOut="8" dialplan.1.lyncDigitmap.timeOut="10" dialplan.1.digitmap="0[27]xxxxxxxxx" dialplan.2.digitmap="123|456" dialplan.2.digitmap.timeOut="4" reg.1.mergeServerDigitMapLocally="1" reg.2.video.enabled="0" reg.limit="2" dialplan.digitmap.lineSwitching.enable="1" /> </PHONE_CONFIG>

 

  • Hybrid Configuration for Numbers 123 or 456 to be sent to openSIP Lines and Skype handles 0[27]xxxxxxxxx as example cover any 11 digit number starting with 02 or 07 and nothing else.

    In addition, SIPURIs and calendar join goes to line 1

  • In addition, the configuration ensures that Video Calls are only made on line 1

Example 3:

 

  • No Skype for Business and all lines openSIP but Exchange Registered
<?xml version="1.0" encoding="UTF-8" standalone="yes"?>
<CONFIG>
	<test
		reg.limit="3"
                reg.1.address="3011"
		reg.1.auth.password="3011"
		reg.1.auth.userId="3011"
		reg.1.label="3011"
		reg.1.server.1.address="asterisk16.t2voice.tech-support.lab"
		reg.2.address="3030"
		reg.2.auth.password="3030"
		reg.2.auth.userId="3030"
		reg.2.label="3030"
		reg.2.server.1.address="server.sbaierhome.lab"
		reg.3.address="3333"
		reg.3.auth.password="3333"
		reg.3.auth.userId="3333"
		reg.3.label="3333"
		reg.3.server.1.address="3cx.sbaierhome.lab"
device.set="1"
device.logincred.password="Your Password"
device.logincred.password.set="1"
device.logincred.user="Your Username"
device.logincred.user.set="1"
exchange.server.url="https://server/EWS/Exchange.asmx"
feature.EWSAutodiscover.enabled="0"
feature.exchangeCalendar.enabled="1" /> </CONFIG>

 

More details regarding the digitmap can be found => here <=

 

Removal of the New Call Softkey:

 

When using a Hybrid registration the factory default would be to display the New Call Softkey which automatically uses Line 1 aka Skype for Business.

 

A user may follow above instructions to enable the relevant digitmap to either send all calls to their respective designations (Example 1) or based on the number dialled (Example 2) but may still want to remove the New Call Softkey.

 

<test feature.enhancedFeatureKeys.enabled="1"
 softkey.feature.directories="0"
 softkey.feature.newcall="0"
 efk.softkey.alignleft="1" />

This will result in this

Hybrid_Softkey_NewCall_Removed.jpg




<======== Signature / Disclaimer ========>
Please be aware:For questions about the type of support to expect please check here

Please also ensure you always check the VoIP , Video Endpoint , Skype for Business , PSTN or RPM FAQ's

Please remember, if you see a post that helped you , and it answers your question, please mark it as an "Accept as Solution".

The title Polycom Employee & Community Manager is an automatic setting within the community and any forum reply or post is based upon my personal experience and does not reflect the opinion or view of my employer.
Poly employee participation within this community is not mandatory and any post or FAQ article provided by myself is done either during my working hours or outside working hours, in my private time, and maybe answered on weekends, bank holidays or personal holidays.
1 REPLY 1
Polycom Employee & Community Manager

[FAQ] Customize Trio In-Call Screen Drawer Icons

Customize Trio In-Call Screen Drawer Icons

 

Trio UC Software 5.9.0 or later

 

Automatically displays the Trio’s dialplad for those call platforms that require you to enter a bridge number as soon as you join the service. Saves the end-users from having to find and press on the numeric keypad.

 

This can be used on a global or per line basis

 

Call_Screen_Drawer_03.pngDial ZoomCall_Screen_Drawer_04.pngPresented with Dialpad

 

This enables, for example, the Dial Pad to appear once a call is connected using a certain Line on a Poly Trio

 

up.callStateView

or

reg.x.callStateView

 

Automatically sets the Skype conference view to drawer for those customers that want access to the in-call options rather than the roster list.

 

Call_Screen_Gallery_View_01.pngGallery View (Default)Call_Screen_Drawer_View_01.pngDrawer view (New Option)

up.callStateView.conference

or

reg.x.callStateView.conference 

Simply download the attached example configuration and import via the Web Interface:

 

  • Utilities > Import & Export Configuration > Import Configuration



<======== Signature / Disclaimer ========>
Please be aware:For questions about the type of support to expect please check here

Please also ensure you always check the VoIP , Video Endpoint , Skype for Business , PSTN or RPM FAQ's

Please remember, if you see a post that helped you , and it answers your question, please mark it as an "Accept as Solution".

The title Polycom Employee & Community Manager is an automatic setting within the community and any forum reply or post is based upon my personal experience and does not reflect the opinion or view of my employer.
Poly employee participation within this community is not mandatory and any post or FAQ article provided by myself is done either during my working hours or outside working hours, in my private time, and maybe answered on weekends, bank holidays or personal holidays.