innovaphone wiki - User contributions [en]

Reference16r1:Concept App Service Conference Scaler

2026-04-30T13:51:07Z

Nwe: /* Calling a conference room / Voice */

[[Category:Concept|Apps]]
The Conference Scaler app is required to use multiple conference interfaces within one conference.

== Applies to ==
* Conference Scaler app

== Requirements ==
* innovaphone PBX (V16r1)
* innovaphone Application Platform
* At least one innovaphone Gateway with SCNF Interfaces (V16r1)
* The Devices API must be available for the settings plugin therefore the gateways have to be connected to devices

== Settings Plugins ==
=== Conference Scaler ===
With settings plugin of the Conference Scaler a Conference Scaler app object can be created, edited and deleted on the PBX.
Furthermore, the conference interface to be used can be configured if the Devices API is available.
=== Conference ===
In the conference settings, the menu item "Additional Features" appears for Conference objects. With this setting you can enable the use of the Conference Scaler within the Conference object. Additionally, you can activate the use of PBX channels licenses.

== More information ==
=== Conference Scaler App (innovaphone-conference-scaler) ===
[[File:Conferencescaler.png|110px|conferencescaler.png/|conferencescaler.png/]]

The hidden app offers an interface via the Service API (com.innovaphone.conference-scaler), that is used by the Conference object.

Parameters:
;URL: <pre>http://<ap.domain.tld>/<domain.tld>/<instance-name>/innovaphone-conference-scaler</pre>

== Features ==
* Allows the use of multiple conference interfaces within multiple conference.
** This removes previous participant limits.
** Adds flexibility to existing infrastructure.
** The Conference Scaler app nominates one conference interface which will manage the control for voice streams. This conference interface will then receive and distribute the audio streams to all interfaces which are connected to the Conference Scaler app.
** The Conference Scaler app will split the participants in a Round Robin maner, therefore the load will be evenly distributed to the interfaces.
** If a conference interface has reached its limit for participants, the Conference Scaler app will select another free conference interface for a new incoming call.

== Configuration ==
To set up the Conference Scaler for a single cluster of conference interfaces, follow these steps:
# Download the Conference Scaler app via the App Store (or use the settings plugin "app installer" to install the app and instance automatically and skip to step 4).
# Install the Conference Scaler app on the App Platform.
# Create an instance in the Conference Scaler app on the App Platform and make sure the instance is running.
# Create via the settings plugin of the Conference Scaler a new Conference Scaler app object.
# Select the conference interfaces to be used by the Conference Scaler app inside the settings plugin of the Conference Scaler.
# Allow the use of the Conference Scaler app on the Conference settings.

To run multiple clusters, you need additional app instances. Repeat steps 3 through 6 to add them. Ensure there are no overlaps between the selected conference interface across the settings plugins of the Conference Scaler. Each Conference object should activate only one Conference Scaler app object to guarantee everything works as expected. Also, make sure that when adding app objects, this is done via the settings plugin of the new instance.

== Restrictions / Known issues ==
* Using the Conference Scaler app generates an overhead depending on usage patterns, so the increase in the number of participants does not correspond to a multiple of x interfaces. One participant will take up an estimation of 1,5x CPU-Load as the usual conference flow (because data is additionally send to a different interface). So for an interface with max. 100 Video participants this amount will decrease to a maximum participant limit of 66 (as 100/1,5=66). Therefore if you want to achieve a maximum of X video participants you need to take that information into account for calculating the needed interfaces.
* In case of failure of the App Platform the Conference Scaler app will not be available as well. Therefore it will not be possible to join or continue an ongoing conference.
* Room reservations are not needed anymore and are therefore not possible.
* Video sharing between the conference interfaces is handled via multicast and this requires that the interfaces are on the same network (Multicast addresses differs between 239.0.0.0 - 239.255.255.255).
* Only SCNF interfaces are currently supported.

== Troubleshooting ==
If you can reproduce the problem, clear the contents of the current app log, reproduce the issue, take a screenshot (depending on the issue), save the browser logs and the app logs, and send them as an attachment to your support ticket.
For the app, please make sure that following trace flags are enabled:
* App
* App Websocket

=== No Channel free ===
If you get a „No Channel free“ Error when trying to join a Conference Room after setting up the Conference, check if the desired room has any reserved channel configured, if so then remove the reservation of channels. The Conference Scaler does not consider channel reservations and therefore rejects the caller.

== Technical Workflow ==
This is the overview of the complete technical workflow

==== Settings Plugin ====
After installing the conference scaler you need to add a new App inside the settings plugin, this app let's you pick your desired interfaces which you would like to cluster for conferences. Here all (SCNF-) interfaces which are connected to devices will be found.

When adding an interface to the App and applying by clicking ok, the Interface will be automatically configured to register via Websocket to the App Service, also the MAC-adress and Interface indication will be added to the Hardware ID of the Conference Scaler App Object (This is used for the Settings plugin, this way the Plugin knows which Interfaces are already configured to use the Conference Scaler).

==== Conference Object ====
To activate the Conference Scaler at a conference object, you need to got to the Settings plugin "conferences". When selecting an existing conference object (or creating a new one) here you will have a new option called "additional features", the conference scaler will appear here and you can select it as used conference ressource. When selecting the conference scaler as ressource, after clicking ok, the conference object will open a Websocket connection to the conference scaler app service, the conference scaler app service will now register through this Websocket connection with Websocket at the conference object. The registration will be seen in PBX -> Registrations labeled with "JSON".

==== Calling a conference room / Voice ====
When you call a conference room the signaling will be send to the conference object with the room number, this signaling will be forwarded to the conference scaler (basically like in the normal scenario where an Interface registers directly at the conference object, where the signaling will be send directly to the interface). The conference scaler now forwards this signaling to any randomly selected interface which has a registration to the conference scaler app, this interface will be also selected as "Primary Interface" for mixing audio between all interfaces.

The conference scaler will open a new room at that Interface (for the called room), Audio will now be established through Interface -> Client.

Now we only have one participant inside the conference room at one interface connected, let's add another participant. Participant 2 calls inside the same conference room, here the signaling to the conference scaler app is the same, but now the conference scaler will select another interface in a round-robin manner to establish a connection to the client. Now we have two interfaces with the same Room, both interfaces are connected to two different participants, at this moment Participant 1 will not be able to hear participant 2 and vice versa.

To establish audio between Interface 1 and Interface 2 the conference scaler will allocate a free RTP Port on both Interfaces, then the conference scaler exchange these Connection information between both Interfaces. Now Interface 1 will join Interface 2 (at the right room on this interface) as participant and vice versa. Any new participant who joins the conference room will be distributed between these two interfaces.

Let's assume we have another Participant and another Interface registered at the conference scaler app service, therefore this participant will be transfered to Interface 3 when calling to the conference room, to distribute audio between Interface 1, Interface 2 and Interface 3 we will use the "Primary Interface" which was mentioned in the beginning. The "Primary Interface" is the one where all other interfaces joins to provide their own audio to the Primary Inertface, the Primary Interface will then distribute it's own and all received audio to all interfaces. Interface 3 will now connect as participant to interface 1 and vice versa, Interface 1 will now distribute audio between Interface 2, Interface 3 and it's own audio stream

==== Video across interfaces ====
To Provide Video across interfaces the Conferece Scaler will tell an Interface (the sender) to use a Multicast adress to send the video to, once requested. The other interface (the receiver) will subscribe to that Multicast Adress and then distribute this requested Video to it's participants which requested the Video. To save load and unnecessary traffic, the Interfaces only send Video when they are requested in real-time, also the sending interface will send maximum 3 Videos per Participant (In every possible reoslution which can be requested (100Kbit/s, 250Kbit/s, 1000Kbit/s)).

If a Video in a special resolution is requested more than once at the same time, the sending interface will still send only one Video to one Multicast adress - all interfaces which requests these videos can then subscribe to that Maulticast-adress to provide it to their own participants. This way only the requested video is sent and it is only sent once, therefore it doesn't matter how often it was requested.

== Related Articles ==
* [[Reference15r1:Concept_Conference_Web_Access]]
* [[Reference15r1:Concept_Conference]]
* [[Courseware:IT_Connect_-_08.1_Conferencing]]
* [[Courseware:IT_Plus_-_Public_Access_to_Conference_Resources#Bandwidth_and_CPU_consideration]]

Reference16r1:Concept App Service Conference Scaler

2026-04-30T13:36:23Z

Nwe: /* Calling a conference room / Voice */

[[Category:Concept|Apps]]
The Conference Scaler app is required to use multiple conference interfaces within one conference.

== Applies to ==
* Conference Scaler app

== Requirements ==
* innovaphone PBX (V16r1)
* innovaphone Application Platform
* At least one innovaphone Gateway with SCNF Interfaces (V16r1)
* The Devices API must be available for the settings plugin therefore the gateways have to be connected to devices

== Settings Plugins ==
=== Conference Scaler ===
With settings plugin of the Conference Scaler a Conference Scaler app object can be created, edited and deleted on the PBX.
Furthermore, the conference interface to be used can be configured if the Devices API is available.
=== Conference ===
In the conference settings, the menu item "Additional Features" appears for Conference objects. With this setting you can enable the use of the Conference Scaler within the Conference object. Additionally, you can activate the use of PBX channels licenses.

== More information ==
=== Conference Scaler App (innovaphone-conference-scaler) ===
[[File:Conferencescaler.png|110px|conferencescaler.png/|conferencescaler.png/]]

The hidden app offers an interface via the Service API (com.innovaphone.conference-scaler), that is used by the Conference object.

Parameters:
;URL: <pre>http://<ap.domain.tld>/<domain.tld>/<instance-name>/innovaphone-conference-scaler</pre>

== Features ==
* Allows the use of multiple conference interfaces within multiple conference.
** This removes previous participant limits.
** Adds flexibility to existing infrastructure.
** The Conference Scaler app nominates one conference interface which will manage the control for voice streams. This conference interface will then receive and distribute the audio streams to all interfaces which are connected to the Conference Scaler app.
** The Conference Scaler app will split the participants in a Round Robin maner, therefore the load will be evenly distributed to the interfaces.
** If a conference interface has reached its limit for participants, the Conference Scaler app will select another free conference interface for a new incoming call.

== Configuration ==
To set up the Conference Scaler for a single cluster of conference interfaces, follow these steps:
# Download the Conference Scaler app via the App Store (or use the settings plugin "app installer" to install the app and instance automatically and skip to step 4).
# Install the Conference Scaler app on the App Platform.
# Create an instance in the Conference Scaler app on the App Platform and make sure the instance is running.
# Create via the settings plugin of the Conference Scaler a new Conference Scaler app object.
# Select the conference interfaces to be used by the Conference Scaler app inside the settings plugin of the Conference Scaler.
# Allow the use of the Conference Scaler app on the Conference settings.

To run multiple clusters, you need additional app instances. Repeat steps 3 through 6 to add them. Ensure there are no overlaps between the selected conference interface across the settings plugins of the Conference Scaler. Each Conference object should activate only one Conference Scaler app object to guarantee everything works as expected. Also, make sure that when adding app objects, this is done via the settings plugin of the new instance.

== Restrictions / Known issues ==
* Using the Conference Scaler app generates an overhead depending on usage patterns, so the increase in the number of participants does not correspond to a multiple of x interfaces. One participant will take up an estimation of 1,5x CPU-Load as the usual conference flow (because data is additionally send to a different interface). So for an interface with max. 100 Video participants this amount will decrease to a maximum participant limit of 66 (as 100/1,5=66). Therefore if you want to achieve a maximum of X video participants you need to take that information into account for calculating the needed interfaces.
* In case of failure of the App Platform the Conference Scaler app will not be available as well. Therefore it will not be possible to join or continue an ongoing conference.
* Room reservations are not needed anymore and are therefore not possible.
* Video sharing between the conference interfaces is handled via multicast and this requires that the interfaces are on the same network (Multicast addresses differs between 239.0.0.0 - 239.255.255.255).
* Only SCNF interfaces are currently supported.

== Troubleshooting ==
If you can reproduce the problem, clear the contents of the current app log, reproduce the issue, take a screenshot (depending on the issue), save the browser logs and the app logs, and send them as an attachment to your support ticket.
For the app, please make sure that following trace flags are enabled:
* App
* App Websocket

=== No Channel free ===
If you get a „No Channel free“ Error when trying to join a Conference Room after setting up the Conference, check if the desired room has any reserved channel configured, if so then remove the reservation of channels. The Conference Scaler does not consider channel reservations and therefore rejects the caller.

== Technical Workflow ==
This is the overview of the complete technical workflow

==== Settings Plugin ====
After installing the conference scaler you need to add a new App inside the settings plugin, this app let's you pick your desired interfaces which you would like to cluster for conferences. Here all (SCNF-) interfaces which are connected to devices will be found.

When adding an interface to the App and applying by clicking ok, the Interface will be automatically configured to register via Websocket to the App Service, also the MAC-adress and Interface indication will be added to the Hardware ID of the Conference Scaler App Object (This is used for the Settings plugin, this way the Plugin knows which Interfaces are already configured to use the Conference Scaler).

==== Conference Object ====
To activate the Conference Scaler at a conference object, you need to got to the Settings plugin "conferences". When selecting an existing conference object (or creating a new one) here you will have a new option called "additional features", the conference scaler will appear here and you can select it as used conference ressource. When selecting the conference scaler as ressource, after clicking ok, the conference object will open a Websocket connection to the conference scaler app service, the conference scaler app service will now register through this Websocket connection with Websocket at the conference object. The registration will be seen in PBX -> Registrations labeled with "JSON".

==== Calling a conference room / Voice ====
When you call a conference room the signaling will be send to the conference object with the room number, this signaling will be forwarded to the conference scaler (basically like in the normal scenario where an Interface registers directly at the conference object, where the signaling will be send directly to the interface). The conference scaler now forwards this signaling to any randomly selected interface which has a registration to the conference scaler app, this interface will be also selected as "Primary Interface" for mixing audio between all interfaces.

The conference scaler will open a new room at that Interface (for the called room), Audio will now be established through Interface -> Client.

Now we only have one participant inside the conference room at one interface connected, let's add another participant. Participant 2 calls inside the same conference room, here the signaling to the conference scaler app is the same, but now the conference scaler will select another interface in a round-robin manner to establish a connection to the client. Now we have two interfaces with the same Room, both interfaces are connected to two different participants, at this moment Participant 1 will not be able to hear participant 2 and vice versa.

To establish audio between Interface 1 and Interface 2 the conference scaler will allocate a free RTP Port on both Interfaces, then the conference scaler exchange these Connection information between both Interfaces. Now Interface 1 will join Interface 2 (at the right room on this interface) as participant and vice versa. Any new participant who joins the conference room will be distributed between these two interfaces.

Let's assume we have another Participant and another Interface registered at the conference scaler app service, therefore this participant will be transfered to Interface 3 when calling to the conference room, to distribute audio between Interface 1, Interface 2 and Interface 3 we will use the "Primary Interface" which was mentioned in the beginning. Interface 3 will now connect as participant to interface 1 and vice versa, Interface 1 will now distribute audio between Interface 2, Interface 3 and it's own audio stream

== Related Articles ==
* [[Reference15r1:Concept_Conference_Web_Access]]
* [[Reference15r1:Concept_Conference]]
* [[Courseware:IT_Connect_-_08.1_Conferencing]]
* [[Courseware:IT_Plus_-_Public_Access_to_Conference_Resources#Bandwidth_and_CPU_consideration]]

Reference16r1:Concept App Service Conference Scaler

2026-04-30T13:14:30Z

Nwe: /* Calling a conference room / Voice */

[[Category:Concept|Apps]]
The Conference Scaler app is required to use multiple conference interfaces within one conference.

== Applies to ==
* Conference Scaler app

== Requirements ==
* innovaphone PBX (V16r1)
* innovaphone Application Platform
* At least one innovaphone Gateway with SCNF Interfaces (V16r1)
* The Devices API must be available for the settings plugin therefore the gateways have to be connected to devices

== Settings Plugins ==
=== Conference Scaler ===
With settings plugin of the Conference Scaler a Conference Scaler app object can be created, edited and deleted on the PBX.
Furthermore, the conference interface to be used can be configured if the Devices API is available.
=== Conference ===
In the conference settings, the menu item "Additional Features" appears for Conference objects. With this setting you can enable the use of the Conference Scaler within the Conference object. Additionally, you can activate the use of PBX channels licenses.

== More information ==
=== Conference Scaler App (innovaphone-conference-scaler) ===
[[File:Conferencescaler.png|110px|conferencescaler.png/|conferencescaler.png/]]

The hidden app offers an interface via the Service API (com.innovaphone.conference-scaler), that is used by the Conference object.

Parameters:
;URL: <pre>http://<ap.domain.tld>/<domain.tld>/<instance-name>/innovaphone-conference-scaler</pre>

== Features ==
* Allows the use of multiple conference interfaces within multiple conference.
** This removes previous participant limits.
** Adds flexibility to existing infrastructure.
** The Conference Scaler app nominates one conference interface which will manage the control for voice streams. This conference interface will then receive and distribute the audio streams to all interfaces which are connected to the Conference Scaler app.
** The Conference Scaler app will split the participants in a Round Robin maner, therefore the load will be evenly distributed to the interfaces.
** If a conference interface has reached its limit for participants, the Conference Scaler app will select another free conference interface for a new incoming call.

== Configuration ==
To set up the Conference Scaler for a single cluster of conference interfaces, follow these steps:
# Download the Conference Scaler app via the App Store (or use the settings plugin "app installer" to install the app and instance automatically and skip to step 4).
# Install the Conference Scaler app on the App Platform.
# Create an instance in the Conference Scaler app on the App Platform and make sure the instance is running.
# Create via the settings plugin of the Conference Scaler a new Conference Scaler app object.
# Select the conference interfaces to be used by the Conference Scaler app inside the settings plugin of the Conference Scaler.
# Allow the use of the Conference Scaler app on the Conference settings.

To run multiple clusters, you need additional app instances. Repeat steps 3 through 6 to add them. Ensure there are no overlaps between the selected conference interface across the settings plugins of the Conference Scaler. Each Conference object should activate only one Conference Scaler app object to guarantee everything works as expected. Also, make sure that when adding app objects, this is done via the settings plugin of the new instance.

== Restrictions / Known issues ==
* Using the Conference Scaler app generates an overhead depending on usage patterns, so the increase in the number of participants does not correspond to a multiple of x interfaces. One participant will take up an estimation of 1,5x CPU-Load as the usual conference flow (because data is additionally send to a different interface). So for an interface with max. 100 Video participants this amount will decrease to a maximum participant limit of 66 (as 100/1,5=66). Therefore if you want to achieve a maximum of X video participants you need to take that information into account for calculating the needed interfaces.
* In case of failure of the App Platform the Conference Scaler app will not be available as well. Therefore it will not be possible to join or continue an ongoing conference.
* Room reservations are not needed anymore and are therefore not possible.
* Video sharing between the conference interfaces is handled via multicast and this requires that the interfaces are on the same network (Multicast addresses differs between 239.0.0.0 - 239.255.255.255).
* Only SCNF interfaces are currently supported.

== Troubleshooting ==
If you can reproduce the problem, clear the contents of the current app log, reproduce the issue, take a screenshot (depending on the issue), save the browser logs and the app logs, and send them as an attachment to your support ticket.
For the app, please make sure that following trace flags are enabled:
* App
* App Websocket

=== No Channel free ===
If you get a „No Channel free“ Error when trying to join a Conference Room after setting up the Conference, check if the desired room has any reserved channel configured, if so then remove the reservation of channels. The Conference Scaler does not consider channel reservations and therefore rejects the caller.

== Technical Workflow ==
This is the overview of the complete technical workflow

==== Settings Plugin ====
After installing the conference scaler you need to add a new App inside the settings plugin, this app let's you pick your desired interfaces which you would like to cluster for conferences. Here all (SCNF-) interfaces which are connected to devices will be found.

When adding an interface to the App and applying by clicking ok, the Interface will be automatically configured to register via Websocket to the App Service, also the MAC-adress and Interface indication will be added to the Hardware ID of the Conference Scaler App Object (This is used for the Settings plugin, this way the Plugin knows which Interfaces are already configured to use the Conference Scaler).

==== Conference Object ====
To activate the Conference Scaler at a conference object, you need to got to the Settings plugin "conferences". When selecting an existing conference object (or creating a new one) here you will have a new option called "additional features", the conference scaler will appear here and you can select it as used conference ressource. When selecting the conference scaler as ressource, after clicking ok, the conference object will open a Websocket connection to the conference scaler app service, the conference scaler app service will now register through this Websocket connection with Websocket at the conference object. The registration will be seen in PBX -> Registrations labeled with "JSON".

==== Calling a conference room / Voice ====
When you call a conference room the signaling will be send to the conference object with the room number, this signaling will be forwarded to the conference scaler (basically like in the normal scenario where an Interface registers directly at the conference object, where the signaling will be send directly to the interface). The conference scaler now forwards this signaling to any randomly selected interface which has a registration to the conference scaler app, this interface will be also selected as "Primary Interface" for mixing audio between all interfaces.

The conference scaler will open a new room at that Interface (for the called room), Audio will now be established through Interface -> Client.

Now we only have one participant inside the conference room at one interface connected, let's add another participant. Participant 2 calls inside the same conference room, here the signaling to the conference scaler app is the same, but now the conference scaler will select another interface in a round-robin manner to establish a connection to the client. Now we have two interfaces with the same Room, both interfaces are connected to two different participants, at this moment Participant 1 will not be able to hear participant 2 and vice versa.

To establish audio between Interface 1 and Interface 2 the conference scaler will allocate a free RTP Port on both Interfaces, then the conference scaler exchange these Connection information between both Interfaces. Now Interface 1 will join Interface 2 (at the right room on this interface) as participant and vice versa. Any new participant who joins the conference room will be distributed between these two interfaces.

Let's assume we have another Participant and another Interface registered at the conference scaler app service, therefore this participant will be transfered to Interface 3 when calling to the conference room, to distribute audio between Interface 1, Interface 2 and Interface 3 we will use the "Primary Interface" which was mentioned in the beginning. Interface 3 will now connect as participant to interface 1 and vice versa, Interface 1 will now distribute audi between Interface 2, Interface 3 and it's own audio stream

== Related Articles ==
* [[Reference15r1:Concept_Conference_Web_Access]]
* [[Reference15r1:Concept_Conference]]
* [[Courseware:IT_Connect_-_08.1_Conferencing]]
* [[Courseware:IT_Plus_-_Public_Access_to_Conference_Resources#Bandwidth_and_CPU_consideration]]

Reference16r1:Concept App Service Conference Scaler

2026-04-30T13:11:27Z

Nwe: /* Conference Object */

[[Category:Concept|Apps]]
The Conference Scaler app is required to use multiple conference interfaces within one conference.

== Applies to ==
* Conference Scaler app

== Requirements ==
* innovaphone PBX (V16r1)
* innovaphone Application Platform
* At least one innovaphone Gateway with SCNF Interfaces (V16r1)
* The Devices API must be available for the settings plugin therefore the gateways have to be connected to devices

== Settings Plugins ==
=== Conference Scaler ===
With settings plugin of the Conference Scaler a Conference Scaler app object can be created, edited and deleted on the PBX.
Furthermore, the conference interface to be used can be configured if the Devices API is available.
=== Conference ===
In the conference settings, the menu item "Additional Features" appears for Conference objects. With this setting you can enable the use of the Conference Scaler within the Conference object. Additionally, you can activate the use of PBX channels licenses.

== More information ==
=== Conference Scaler App (innovaphone-conference-scaler) ===
[[File:Conferencescaler.png|110px|conferencescaler.png/|conferencescaler.png/]]

The hidden app offers an interface via the Service API (com.innovaphone.conference-scaler), that is used by the Conference object.

Parameters:
;URL: <pre>http://<ap.domain.tld>/<domain.tld>/<instance-name>/innovaphone-conference-scaler</pre>

== Features ==
* Allows the use of multiple conference interfaces within multiple conference.
** This removes previous participant limits.
** Adds flexibility to existing infrastructure.
** The Conference Scaler app nominates one conference interface which will manage the control for voice streams. This conference interface will then receive and distribute the audio streams to all interfaces which are connected to the Conference Scaler app.
** The Conference Scaler app will split the participants in a Round Robin maner, therefore the load will be evenly distributed to the interfaces.
** If a conference interface has reached its limit for participants, the Conference Scaler app will select another free conference interface for a new incoming call.

== Configuration ==
To set up the Conference Scaler for a single cluster of conference interfaces, follow these steps:
# Download the Conference Scaler app via the App Store (or use the settings plugin "app installer" to install the app and instance automatically and skip to step 4).
# Install the Conference Scaler app on the App Platform.
# Create an instance in the Conference Scaler app on the App Platform and make sure the instance is running.
# Create via the settings plugin of the Conference Scaler a new Conference Scaler app object.
# Select the conference interfaces to be used by the Conference Scaler app inside the settings plugin of the Conference Scaler.
# Allow the use of the Conference Scaler app on the Conference settings.

To run multiple clusters, you need additional app instances. Repeat steps 3 through 6 to add them. Ensure there are no overlaps between the selected conference interface across the settings plugins of the Conference Scaler. Each Conference object should activate only one Conference Scaler app object to guarantee everything works as expected. Also, make sure that when adding app objects, this is done via the settings plugin of the new instance.

== Restrictions / Known issues ==
* Using the Conference Scaler app generates an overhead depending on usage patterns, so the increase in the number of participants does not correspond to a multiple of x interfaces. One participant will take up an estimation of 1,5x CPU-Load as the usual conference flow (because data is additionally send to a different interface). So for an interface with max. 100 Video participants this amount will decrease to a maximum participant limit of 66 (as 100/1,5=66). Therefore if you want to achieve a maximum of X video participants you need to take that information into account for calculating the needed interfaces.
* In case of failure of the App Platform the Conference Scaler app will not be available as well. Therefore it will not be possible to join or continue an ongoing conference.
* Room reservations are not needed anymore and are therefore not possible.
* Video sharing between the conference interfaces is handled via multicast and this requires that the interfaces are on the same network (Multicast addresses differs between 239.0.0.0 - 239.255.255.255).
* Only SCNF interfaces are currently supported.

== Troubleshooting ==
If you can reproduce the problem, clear the contents of the current app log, reproduce the issue, take a screenshot (depending on the issue), save the browser logs and the app logs, and send them as an attachment to your support ticket.
For the app, please make sure that following trace flags are enabled:
* App
* App Websocket

=== No Channel free ===
If you get a „No Channel free“ Error when trying to join a Conference Room after setting up the Conference, check if the desired room has any reserved channel configured, if so then remove the reservation of channels. The Conference Scaler does not consider channel reservations and therefore rejects the caller.

== Technical Workflow ==
This is the overview of the complete technical workflow

==== Settings Plugin ====
After installing the conference scaler you need to add a new App inside the settings plugin, this app let's you pick your desired interfaces which you would like to cluster for conferences. Here all (SCNF-) interfaces which are connected to devices will be found.

When adding an interface to the App and applying by clicking ok, the Interface will be automatically configured to register via Websocket to the App Service, also the MAC-adress and Interface indication will be added to the Hardware ID of the Conference Scaler App Object (This is used for the Settings plugin, this way the Plugin knows which Interfaces are already configured to use the Conference Scaler).

==== Conference Object ====
To activate the Conference Scaler at a conference object, you need to got to the Settings plugin "conferences". When selecting an existing conference object (or creating a new one) here you will have a new option called "additional features", the conference scaler will appear here and you can select it as used conference ressource. When selecting the conference scaler as ressource, after clicking ok, the conference object will open a Websocket connection to the conference scaler app service, the conference scaler app service will now register through this Websocket connection with Websocket at the conference object. The registration will be seen in PBX -> Registrations labeled with "JSON".

==== Calling a conference room / Voice ====
When you call a conference room the signaling will be send to the conference object with the room number, this signaling will be forwarded to the conference scaler (basically like in the normal scenario where an Interface registers directly at the conference object, where the signaling will be send directly to the interface). The conference scaler now forwards this signaling to any randomly selected interface which has a registration to the conference scaler app, this interface will be also selected as "Primary Interface" for mixing audio between all interfaces.

The conference scaler will open a new room at that Interface (for the called room), Audio will now be established through Interface -> Client.

Now we only have one participant inside the conference room at one interface connected, let's add another participant. Participant 2 calls inside the same conference room, here the signaling to the conference scaler app is the same, but now the conference scaler will select another interface in a round-robin manner to establish a connection to the client. Now we have two interfaces with the same Room, both interfaces are connected to two different participants, at this moment Participant 1 will not be able to hear participant 2 and vice versa.

To establish audio between Interface 1 and Interface 2 the conference scaler will allocate a free RTP Port on both Interfaces, then the conference scaler exchange these Connection information between both Interfaces. Now Interface 1 will join Interface 2 (at the right room on this interface) as participant and vice versa. Any new participant who joins the conference room will be distributed between these two interfaces.

Let's assume we have another Participant and another Interface registered at the conference scaler app service, therefore this participant will be transfered to Interface 3 when calling

== Related Articles ==
* [[Reference15r1:Concept_Conference_Web_Access]]
* [[Reference15r1:Concept_Conference]]
* [[Courseware:IT_Connect_-_08.1_Conferencing]]
* [[Courseware:IT_Plus_-_Public_Access_to_Conference_Resources#Bandwidth_and_CPU_consideration]]

Reference16r1:Concept App Service Conference Scaler

2026-04-30T12:13:15Z

Nwe: /* Technical Workflow */

Reference16r1:Concept App Service Conference Scaler

2026-04-30T12:00:30Z

Nwe: /* No Channel free */

Howto16r1:Conferences, Resources and Licenses

2026-04-30T11:36:36Z

Nwe: /* Ressource consideration */

==Applies To==
This information applies to

all innovaphone devices supporting a ''CONF'' or ''SCNF'' interface and hence ''Conference Channels'' (as shown in the devices home-page).



==More Information==
Performing conferencing on a [[Reference10:Gateway/Interfaces#Conferencing interface .28CONF.29| ''CONF or SCNF'' interface]] consumes both resources and licenses. Here is an overview of which and how much.

=== Overview ===

First let us have a look at the overall scheme:

[[Image:Conferences_Ressources_and_Licenses_-_Scheme2_G722.png|conferences_ressources_and_licenses_-_scheme2_g722.png/|conferences_ressources_and_licenses_-_scheme2_g722.png/]]

==== Coder channels ====
Both the ''CONF'' and ''SCNF'' interface provides the mixing of 8KHz G711 PCM audio channels.

If a call is done towards a CONF interface, it will allocate a ''Coder channel'' from the DSP coder channel bank for processing of the audio from the VoIP codec (e.g. G.711, G.722, OPUS) to PCM audio.

The SCNF supports G.711a/u and G.722 coder. No ''Coder channel'' needs to be allocated for a call.

==== (Soft) Conference Channels ====
Each call to a CONF interface consumes one of the ''Conference Channels''.

Each call to an SCNF interface consumes one of the ''Soft Conference Channels''.

==== Channels license ====

In addition to that, a call to either the CONF or SCNF requires a ''Channel'' license. This license can be obtained from the PBX if the ''PBX Channels licenses'' switch in the PBX [[Reference13r1:PBX/Objects/Conference| ''Conference'' object]] is activated (if the call comes through such an object) or the ''Obtain Channels lic on outgoing call'' check-mark is ticked in a [[Reference13r1:PBX/Objects/Gateway| PBX ''Gateway'' object]] (if the call comes through such an object). For this, ''PBX Channels licenses'' must be installed on the PBX.

This is the recommended configuration. However, the ''Channel'' license can also be obtained locally from the gateway where the CONF/SCNF is located on. All innovaphone gateways have a number of ''Channel'' licenses built-in. The number of licenses available is equal to the number of ''Coder'' channels the box supports. If no license is sent along with the call to the CONF or SCNF interface, the interface will try to obtain one from the pool of built-in licenses. This can save you some cost. However, be aware that these licenses (as well as the corresponding ''Channel'' coders) are also required for calls through the ISDN BRI/PRI interfaces or for audio fax calls. Calls to CONF/SCNF interfaces which consume local ''Channel'' licenses may inhibit such calls therefor.

The PBX-Channels License has order no. 02-00020-007 according to chapter "3.5 PBX Channels license" in the [https://www.innovaphone.com/content/downloads/innovaphone-Licensing%20Guidelines-V13r2-EN.pdf innovaphone license guide].

The [[Reference13r1:PBX/Config/General#License Status|status of the PBX Licenses]] shows assigned PBX-Channel licenses:

[[Image:Conferences,_Ressources_and_Licenses_-_pbx-channel_license_01.png|conferences,_ressources_and_licenses_-_pbx-channel_license_01.png/|conferences,_ressources_and_licenses_-_pbx-channel_license_01.png/]]

==== Port license ====
The registration of the CONF and/or SCNF to the PBX Conference object does not require a Port license.

Although this is not directly related to the CONF or SCNF interface, please note that a single ''Port'' license is required if any of the rooms defined in a PBX ''Conference'' object is accessed using the conference web access (from 13r3).

==== CPU usage ====
For a calculation of CPU usage see [[Howto:V13_Firmware_Upgrade_V13r2_V13r3#Conferences]].

===Example Scenarios===

====Conference on hardware gateway with local ISDN====
Let us assume we have a conference running on a CONF interface with one PSTN and 3 VoIP participants. In this case, we have

{| border=1
| ||Caller || DSP Coder Channel || Conference Channel || PBX Channel License
|-
| ||1 PSTN || 1 (CONF) + 1 (ISDN) **|| 1 || 1
|-
| ||3 VoIP || 3|| 3 || 3
|-
| Total || 4 || 5 || 4 || 4
|}
<nowiki>**</nowiki>: {{FIXME|reason=do we need only one or would this require ''enable PCM''?}}

Looking at an IP411, it supports the following resources:

[[Image:Conferences,_Ressources_and_Licenses_-_example_gateway_channels_IP411.png|conferences,_ressources_and_licenses_-_example_gateway_channels_ip411.png/|conferences,_ressources_and_licenses_-_example_gateway_channels_ip411.png/]]

We can see that our sample scenario will not work on an IP411 as there are no ''Conference Channels'' available. However, if we change the scenario so that an SCNF is used instead of a CONF, it would look as follows:

{| border=1
| ||Caller || DSP Coder Channel || Soft Conference Channel || PBX Channel License
|-
| ||1 PSTN || 1 (ISDN) **|| 1 || 1
|-
| ||3 VoIP || 3|| 3 || 3
|-
| Total || 4 || 4 || 4 || 4
|}

This will work on an IP411.

=== 3PTY Conferencing ===
3-way conferencing is special as it is usually implemented internally in the innovaphone IP phones. That is, each user of an innovaphone IP phone (except the IP61) has internal conferencing resources built-in to allow for a 3-way conference. No external ''CONF/SCNF'' interface is needed.

However, on innovaphone DECT phones, no internal conferencing resource is present and hence no internal 3PTY is possible. The DECT system can thus be [[Reference9:3pty conference on DECT phones|configured to use an external conferencing resource]] for 3PTY. In this case, the rules above apply.

=== Conference Scaler ===
With 16r1 we introduce a new App "Conference Scaler", with the Conference Scaler it is possible to cluster multiple SCNF Interfaces into one registering Endpoint at a conference Object, for more please refer to [[Reference16r1:Concept App Service Conference Scaler| ''the concept article'']]

==== Ressource consideration ====

==== Licenses ====
The conference scaler app is free to use, only the needed ressources (Hardware gateways or IPVAs) must be considered when using the conference scaler. For the license mechanism nothing changes when using the conference scaler, so for licensing questions see above

===== Conference Channels =====
If using the conference scaler and multiple interfaces you need to take into account to be sure how much conference channels are actually used, even if they are not be seen.
Let's have a look what happens for each room inside a Conference object that uses the conference scaler:

* As soon as a room has been opened on more than one interface (this will be the case when having at least 2 participants), the audio is only transmitted between the Interface and the locally connected participants
** To share audio between interfaces, the conference scaler will allocate a RTP Port on both interfaces
** This Socket will be exchanged between both interfaces, so that both interfaces know where to connect to
** Now, as both interfaces have the needed information, both interfaces will join to the other Interface as an "invisible" participant, taking up one more extra conference channel on both interfaces
* As we have learned in the concept article, the conference scaler will allocate one random interface as primary interface, this Primary interface will then be used to connect all used interfaces and mixing the audio. Let's go through this with our already existing scenario:
** Now we have two participants on two interfaces which are connected together ( and the conference scaler has already selected Interface 1 as the primary interface in the beginning of the conference call)
** Another participant joins the conference call, now let's assume we use 3 interfaces with the conference scaler, therefore this new participant will join at interface 3 and open this room there new
** As Interface 1 is the primary Interface, interface 3 will join the conference room ONLY at interface 1, taking up another conference channel as "invisible" participant (Interface 3 does not need to join interface 2 because the audio from interface 2 is already transmitted to interface 1, our primary interface)
** Interface 1 will also join interface 3 as "invisible" participant, taking up one conference channel on interface 3
* Now we have 2 Interfaces connected to interface 1 and interface 1 is connected to 2 interfaces as participant
* This is the flow for every conference room you are using at that conference object, for the next room the conference scaler might select Interface 3 as primary interface, in this case interface 3 will join both interfaces as participant and vice versa (also taking up conference channels here as well) - but only if this conference call has at least these 3 participants (as the interfaces will be used in a round robin manner)
* The primary interface will now receive audio from both interfaces, which will be the mixed audio from both interfaces, and distributes it to the other interfaces + it's own audio mixed together 

For Room 1 it will look like this:
{| class="wikitable"
|+
!Interface 1 (Primary Interface)
!Interface 2
!Interface 3
|-
|Participant 1
|Participant 2
|Participant 3
|-
|Interface 2 ("invisbile participant")
|Interface 1 ("invisbile participant")
|Interface 1 ("invisbile participant")
|-
|Interface 3 ("invisbile participant")
|
|
|}
Let's see what it looks like when you use another room at that conference object (Room 1 will be displayed in italic, to demonstrate what it looks like with multiple rooms):
{| class="wikitable"
|+
!Interface 1 ''(Primary Interface room 1)''
!Interface 2
!Interface 3 (Primary interface for room 2)
|-
|''Participant 1''
|''Participant 2''
|''Participant 3''
|-
|''Interface 2 ("invisbile participant")''
|''Interface 1 ("invisbile participant")''
|''Interface 1 ("invisbile participant")''
|-
|''Interface 3 ("invisbile participant")''
|Participant 2 - room 2
|Participant 1 - room 2
|-
|Participant 3 - room 2
|Interface 3 ("invisbile participant") - room 2
|Interface 2 ("invisbile participant") - room 2
|-
|Interface 3 ("invisbile participant") - room 2
|
|Interface 1 ("invisbile participant") - room 2
|}

== Related Articles ==
* [[Reference9:Concept myPBX]]
* [[Reference10:Concept Innovaphone Virtual Appliance]]
* [[Reference9:3pty conference on DECT phones]]
* [[Reference10:Gateway/Interfaces#Conferencing_interface_.28CONF.29]]
* [[Howto:How_to_implement_large_PBXs#Technical_data_and_recommended_number_of_users_supported]], lists DSP and CONF resources for the different gateway platforms

[[Category:Howto|{{PAGENAME}}]]
[[Category:Concept Conference]]

Howto16r1:Conferences, Resources and Licenses

2026-04-30T11:24:06Z

Nwe: /* Conference Channels */

==Applies To==
This information applies to

all innovaphone devices supporting a ''CONF'' or ''SCNF'' interface and hence ''Conference Channels'' (as shown in the devices home-page).



==More Information==
Performing conferencing on a [[Reference10:Gateway/Interfaces#Conferencing interface .28CONF.29| ''CONF or SCNF'' interface]] consumes both resources and licenses. Here is an overview of which and how much.

=== Overview ===

First let us have a look at the overall scheme:

[[Image:Conferences_Ressources_and_Licenses_-_Scheme2_G722.png|conferences_ressources_and_licenses_-_scheme2_g722.png/|conferences_ressources_and_licenses_-_scheme2_g722.png/]]

==== Coder channels ====
Both the ''CONF'' and ''SCNF'' interface provides the mixing of 8KHz G711 PCM audio channels.

If a call is done towards a CONF interface, it will allocate a ''Coder channel'' from the DSP coder channel bank for processing of the audio from the VoIP codec (e.g. G.711, G.722, OPUS) to PCM audio.

The SCNF supports G.711a/u and G.722 coder. No ''Coder channel'' needs to be allocated for a call.

==== (Soft) Conference Channels ====
Each call to a CONF interface consumes one of the ''Conference Channels''.

Each call to an SCNF interface consumes one of the ''Soft Conference Channels''.

==== Channels license ====

In addition to that, a call to either the CONF or SCNF requires a ''Channel'' license. This license can be obtained from the PBX if the ''PBX Channels licenses'' switch in the PBX [[Reference13r1:PBX/Objects/Conference| ''Conference'' object]] is activated (if the call comes through such an object) or the ''Obtain Channels lic on outgoing call'' check-mark is ticked in a [[Reference13r1:PBX/Objects/Gateway| PBX ''Gateway'' object]] (if the call comes through such an object). For this, ''PBX Channels licenses'' must be installed on the PBX.

This is the recommended configuration. However, the ''Channel'' license can also be obtained locally from the gateway where the CONF/SCNF is located on. All innovaphone gateways have a number of ''Channel'' licenses built-in. The number of licenses available is equal to the number of ''Coder'' channels the box supports. If no license is sent along with the call to the CONF or SCNF interface, the interface will try to obtain one from the pool of built-in licenses. This can save you some cost. However, be aware that these licenses (as well as the corresponding ''Channel'' coders) are also required for calls through the ISDN BRI/PRI interfaces or for audio fax calls. Calls to CONF/SCNF interfaces which consume local ''Channel'' licenses may inhibit such calls therefor.

The PBX-Channels License has order no. 02-00020-007 according to chapter "3.5 PBX Channels license" in the [https://www.innovaphone.com/content/downloads/innovaphone-Licensing%20Guidelines-V13r2-EN.pdf innovaphone license guide].

The [[Reference13r1:PBX/Config/General#License Status|status of the PBX Licenses]] shows assigned PBX-Channel licenses:

[[Image:Conferences,_Ressources_and_Licenses_-_pbx-channel_license_01.png|conferences,_ressources_and_licenses_-_pbx-channel_license_01.png/|conferences,_ressources_and_licenses_-_pbx-channel_license_01.png/]]

==== Port license ====
The registration of the CONF and/or SCNF to the PBX Conference object does not require a Port license.

Although this is not directly related to the CONF or SCNF interface, please note that a single ''Port'' license is required if any of the rooms defined in a PBX ''Conference'' object is accessed using the conference web access (from 13r3).

==== CPU usage ====
For a calculation of CPU usage see [[Howto:V13_Firmware_Upgrade_V13r2_V13r3#Conferences]].

===Example Scenarios===

====Conference on hardware gateway with local ISDN====
Let us assume we have a conference running on a CONF interface with one PSTN and 3 VoIP participants. In this case, we have

{| border=1
| ||Caller || DSP Coder Channel || Conference Channel || PBX Channel License
|-
| ||1 PSTN || 1 (CONF) + 1 (ISDN) **|| 1 || 1
|-
| ||3 VoIP || 3|| 3 || 3
|-
| Total || 4 || 5 || 4 || 4
|}
<nowiki>**</nowiki>: {{FIXME|reason=do we need only one or would this require ''enable PCM''?}}

Looking at an IP411, it supports the following resources:

[[Image:Conferences,_Ressources_and_Licenses_-_example_gateway_channels_IP411.png|conferences,_ressources_and_licenses_-_example_gateway_channels_ip411.png/|conferences,_ressources_and_licenses_-_example_gateway_channels_ip411.png/]]

We can see that our sample scenario will not work on an IP411 as there are no ''Conference Channels'' available. However, if we change the scenario so that an SCNF is used instead of a CONF, it would look as follows:

{| border=1
| ||Caller || DSP Coder Channel || Soft Conference Channel || PBX Channel License
|-
| ||1 PSTN || 1 (ISDN) **|| 1 || 1
|-
| ||3 VoIP || 3|| 3 || 3
|-
| Total || 4 || 4 || 4 || 4
|}

This will work on an IP411.

=== 3PTY Conferencing ===
3-way conferencing is special as it is usually implemented internally in the innovaphone IP phones. That is, each user of an innovaphone IP phone (except the IP61) has internal conferencing resources built-in to allow for a 3-way conference. No external ''CONF/SCNF'' interface is needed.

However, on innovaphone DECT phones, no internal conferencing resource is present and hence no internal 3PTY is possible. The DECT system can thus be [[Reference9:3pty conference on DECT phones|configured to use an external conferencing resource]] for 3PTY. In this case, the rules above apply.

=== Conference Scaler ===
With 16r1 we introduce a new App "Conference Scaler", with the Conference Scaler it is possible to cluster multiple SCNF Interfaces into one registering Endpoint at a conference Object, for more please refer to [[Reference16r1:Concept App Service Conference Scaler| ''the concept article'']]

==== Ressource consideration ====
===== Conference Channels =====
If using the conference scaler and multiple interfaces you need to take into account to be sure how much conference channels are actually used, even if they are not be seen.
Let's have a look what happens for each room inside a Conference object that uses the conference scaler:

* As soon as a room has been opened on more than one interface (this will be the case when having at least 2 participants), the audio is only transmitted between the Interface and the locally connected participants
** To share audio between interfaces, the conference scaler will allocate a RTP Port on both interfaces
** This Socket will be exchanged between both interfaces, so that both interfaces know where to connect to
** Now, as both interfaces have the needed information, both interfaces will join to the other Interface as an "invisible" participant, taking up one more extra conference channel on both interfaces
* As we have learned in the concept article, the conference scaler will allocate one random interface as primary interface, this Primary interface will then be used to connect all used interfaces and mixing the audio. Let's go through this with our already existing scenario:
** Now we have two participants on two interfaces which are connected together ( and the conference scaler has already selected Interface 1 as the primary interface in the beginning of the conference call)
** Another participant joins the conference call, now let's assume we use 3 interfaces with the conference scaler, therefore this new participant will join at interface 3 and open this room there new
** As Interface 1 is the primary Interface, interface 3 will join the conference room ONLY at interface 1, taking up another conference channel as "invisible" participant (Interface 3 does not need to join interface 2 because the audio from interface 2 is already transmitted to interface 1, our primary interface)
** Interface 1 will also join interface 3 as "invisible" participant, taking up one conference channel on interface 3
* Now we have 2 Interfaces connected to interface 1 and interface 1 is connected to 2 interfaces as participant
* This is the flow for every conference room you are using at that conference object, for the next room the conference scaler might select Interface 3 as primary interface, in this case interface 3 will join both interfaces as participant and vice versa (also taking up conference channels here as well) - but only if this conference call has at least these 3 participants (as the interfaces will be used in a round robin manner)
* The primary interface will now receive audio from both interfaces, which will be the mixed audio from both interfaces, and distributes it to the other interfaces + it's own audio mixed together 

For Room 1 it will look like this:
{| class="wikitable"
|+
!Interface 1 (Primary Interface)
!Interface 2
!Interface 3
|-
|Participant 1
|Participant 2
|Participant 3
|-
|Interface 2 ("invisbile participant")
|Interface 1 ("invisbile participant")
|Interface 1 ("invisbile participant")
|-
|Interface 3 ("invisbile participant")
|
|
|}
Let's see what it looks like when you use another room at that conference object (Room 1 will be displayed in italic, to demonstrate what it looks like with multiple rooms):
{| class="wikitable"
|+
!Interface 1 ''(Primary Interface room 1)''
!Interface 2
!Interface 3 (Primary interface for room 2)
|-
|''Participant 1''
|''Participant 2''
|''Participant 3''
|-
|''Interface 2 ("invisbile participant")''
|''Interface 1 ("invisbile participant")''
|''Interface 1 ("invisbile participant")''
|-
|''Interface 3 ("invisbile participant")''
|Participant 2 - room 2
|Participant 1 - room 2
|-
|Participant 3 - room 2
|Interface 3 ("invisbile participant") - room 2
|Interface 2 ("invisbile participant") - room 2
|-
|Interface 3 ("invisbile participant") - room 2
|
|Interface 1 ("invisbile participant") - room 2
|}

== Related Articles ==
* [[Reference9:Concept myPBX]]
* [[Reference10:Concept Innovaphone Virtual Appliance]]
* [[Reference9:3pty conference on DECT phones]]
* [[Reference10:Gateway/Interfaces#Conferencing_interface_.28CONF.29]]
* [[Howto:How_to_implement_large_PBXs#Technical_data_and_recommended_number_of_users_supported]], lists DSP and CONF resources for the different gateway platforms

[[Category:Howto|{{PAGENAME}}]]
[[Category:Concept Conference]]

Howto16r1:Conferences, Resources and Licenses

2026-04-30T11:19:31Z

Nwe: /* Conference Channels */

==Applies To==
This information applies to

all innovaphone devices supporting a ''CONF'' or ''SCNF'' interface and hence ''Conference Channels'' (as shown in the devices home-page).



==More Information==
Performing conferencing on a [[Reference10:Gateway/Interfaces#Conferencing interface .28CONF.29| ''CONF or SCNF'' interface]] consumes both resources and licenses. Here is an overview of which and how much.

=== Overview ===

First let us have a look at the overall scheme:

[[Image:Conferences_Ressources_and_Licenses_-_Scheme2_G722.png|conferences_ressources_and_licenses_-_scheme2_g722.png/|conferences_ressources_and_licenses_-_scheme2_g722.png/]]

==== Coder channels ====
Both the ''CONF'' and ''SCNF'' interface provides the mixing of 8KHz G711 PCM audio channels.

If a call is done towards a CONF interface, it will allocate a ''Coder channel'' from the DSP coder channel bank for processing of the audio from the VoIP codec (e.g. G.711, G.722, OPUS) to PCM audio.

The SCNF supports G.711a/u and G.722 coder. No ''Coder channel'' needs to be allocated for a call.

==== (Soft) Conference Channels ====
Each call to a CONF interface consumes one of the ''Conference Channels''.

Each call to an SCNF interface consumes one of the ''Soft Conference Channels''.

==== Channels license ====

In addition to that, a call to either the CONF or SCNF requires a ''Channel'' license. This license can be obtained from the PBX if the ''PBX Channels licenses'' switch in the PBX [[Reference13r1:PBX/Objects/Conference| ''Conference'' object]] is activated (if the call comes through such an object) or the ''Obtain Channels lic on outgoing call'' check-mark is ticked in a [[Reference13r1:PBX/Objects/Gateway| PBX ''Gateway'' object]] (if the call comes through such an object). For this, ''PBX Channels licenses'' must be installed on the PBX.

This is the recommended configuration. However, the ''Channel'' license can also be obtained locally from the gateway where the CONF/SCNF is located on. All innovaphone gateways have a number of ''Channel'' licenses built-in. The number of licenses available is equal to the number of ''Coder'' channels the box supports. If no license is sent along with the call to the CONF or SCNF interface, the interface will try to obtain one from the pool of built-in licenses. This can save you some cost. However, be aware that these licenses (as well as the corresponding ''Channel'' coders) are also required for calls through the ISDN BRI/PRI interfaces or for audio fax calls. Calls to CONF/SCNF interfaces which consume local ''Channel'' licenses may inhibit such calls therefor.

The PBX-Channels License has order no. 02-00020-007 according to chapter "3.5 PBX Channels license" in the [https://www.innovaphone.com/content/downloads/innovaphone-Licensing%20Guidelines-V13r2-EN.pdf innovaphone license guide].

The [[Reference13r1:PBX/Config/General#License Status|status of the PBX Licenses]] shows assigned PBX-Channel licenses:

[[Image:Conferences,_Ressources_and_Licenses_-_pbx-channel_license_01.png|conferences,_ressources_and_licenses_-_pbx-channel_license_01.png/|conferences,_ressources_and_licenses_-_pbx-channel_license_01.png/]]

==== Port license ====
The registration of the CONF and/or SCNF to the PBX Conference object does not require a Port license.

Although this is not directly related to the CONF or SCNF interface, please note that a single ''Port'' license is required if any of the rooms defined in a PBX ''Conference'' object is accessed using the conference web access (from 13r3).

==== CPU usage ====
For a calculation of CPU usage see [[Howto:V13_Firmware_Upgrade_V13r2_V13r3#Conferences]].

===Example Scenarios===

====Conference on hardware gateway with local ISDN====
Let us assume we have a conference running on a CONF interface with one PSTN and 3 VoIP participants. In this case, we have

{| border=1
| ||Caller || DSP Coder Channel || Conference Channel || PBX Channel License
|-
| ||1 PSTN || 1 (CONF) + 1 (ISDN) **|| 1 || 1
|-
| ||3 VoIP || 3|| 3 || 3
|-
| Total || 4 || 5 || 4 || 4
|}
<nowiki>**</nowiki>: {{FIXME|reason=do we need only one or would this require ''enable PCM''?}}

Looking at an IP411, it supports the following resources:

[[Image:Conferences,_Ressources_and_Licenses_-_example_gateway_channels_IP411.png|conferences,_ressources_and_licenses_-_example_gateway_channels_ip411.png/|conferences,_ressources_and_licenses_-_example_gateway_channels_ip411.png/]]

We can see that our sample scenario will not work on an IP411 as there are no ''Conference Channels'' available. However, if we change the scenario so that an SCNF is used instead of a CONF, it would look as follows:

{| border=1
| ||Caller || DSP Coder Channel || Soft Conference Channel || PBX Channel License
|-
| ||1 PSTN || 1 (ISDN) **|| 1 || 1
|-
| ||3 VoIP || 3|| 3 || 3
|-
| Total || 4 || 4 || 4 || 4
|}

This will work on an IP411.

=== 3PTY Conferencing ===
3-way conferencing is special as it is usually implemented internally in the innovaphone IP phones. That is, each user of an innovaphone IP phone (except the IP61) has internal conferencing resources built-in to allow for a 3-way conference. No external ''CONF/SCNF'' interface is needed.

However, on innovaphone DECT phones, no internal conferencing resource is present and hence no internal 3PTY is possible. The DECT system can thus be [[Reference9:3pty conference on DECT phones|configured to use an external conferencing resource]] for 3PTY. In this case, the rules above apply.

=== Conference Scaler ===
With 16r1 we introduce a new App "Conference Scaler", with the Conference Scaler it is possible to cluster multiple SCNF Interfaces into one registering Endpoint at a conference Object, for more please refer to [[Reference16r1:Concept App Service Conference Scaler| ''the concept article'']]

==== Ressource consideration ====
===== Conference Channels =====
If using the conference scaler and multiple interfaces you need to take into account to be sure how much conference channels are actually used, even if they are not be seen.
Let's have a look what happens for each room inside a Conference object that uses the conference scaler:

* As soon as a room has been opened on more than one interface (this will be the case when having at least 2 participants), the audio is only transmitted between the Interface and the locally connected participants
** To share audio between interfaces, the conference scaler will allocate a RTP Port on both interfaces
** This Socket will be exchanged between both interfaces, so that both interfaces know where to connect to
** Now, as both interfaces have the needed information, both interfaces will join to the other Interface as an "invisible" participant, taking up one more extra conference channel on both interfaces
* As we have learned in the concept article, the conference scaler will allocate one random interface as primary interface, this Primary interface will then be used to connect all used interfaces and mixing the audio. Let's go through this with our already existing scenario:
** Now we have two participants on two interfaces which are connected together ( and the conference scaler has already selected Interface 1 as the primary interface in the beginning of the conference call)
** Another participant joins the conference call, now let's assume we use 3 interfaces with the conference scaler, therefore this new participant will join at interface 3 and open this room there new
** As Interface 1 is the primary Interface, interface 3 will join the conference room ONLY at interface 1, taking up another conference channel as "invisible" participant (Interface 3 does not need to join interface 2 because the audio from interface 2 is already transmitted to interface 1, our primary interface)
** Interface 1 will also join interface 3 as "invisible" participant, taking up one conference channel on interface 3
* Now we have 2 Interfaces connected to interface 1 and interface 1 is connected to 2 interfaces as participant
* This is the flow for every conference room you are using at that conference object, for the next room the conference scaler might select Interface 3 as primary interface, in this case interface 3 will join both interfaces as participant and vice versa (also taking up conference channels here as well) - but only if this conference call has at least these 3 participants (as the interfaces will be used in a round robin manner)
* The primary interface will now receive audio from both interfaces, which will be the mixed audio from both interfaces, and distributes it to the other interfaces + it's own audio mixed together 

For Room 1 it will look like this:
{| class="wikitable"
|+
!Interface 1 (Primary Interface)
!Interface 2
!Interface 3
|-
|Participant 1
|Participant 2
|Participant 3
|-
|Interface 2 ("invisbile participant")
|Interface 1 ("invisbile participant")
|Interface 1 ("invisbile participant")
|-
|Interface 3 ("invisbile participant")
|
|
|}
Let's see what it looks like when you use another room at that conference object (Room 1 will be displayed grey, to demonstrate what it looks like with multiple rooms):
{| class="wikitable"
|+
!Interface 1 (Primary Interface)
!Interface 2
!Interface 3
|-
|Participant 1
|Participant 2
|Participant 3
|-
|Interface 2 ("invisbile participant")
|Interface 1 ("invisbile participant")
|Interface 1 ("invisbile participant")
|-
|Interface 3 ("invisbile participant")
|
|
|}

== Related Articles ==
* [[Reference9:Concept myPBX]]
* [[Reference10:Concept Innovaphone Virtual Appliance]]
* [[Reference9:3pty conference on DECT phones]]
* [[Reference10:Gateway/Interfaces#Conferencing_interface_.28CONF.29]]
* [[Howto:How_to_implement_large_PBXs#Technical_data_and_recommended_number_of_users_supported]], lists DSP and CONF resources for the different gateway platforms

[[Category:Howto|{{PAGENAME}}]]
[[Category:Concept Conference]]

Howto16r1:Conferences, Resources and Licenses

2026-04-30T11:19:09Z

Nwe: /* Conference Channels */

==Applies To==
This information applies to

all innovaphone devices supporting a ''CONF'' or ''SCNF'' interface and hence ''Conference Channels'' (as shown in the devices home-page).



==More Information==
Performing conferencing on a [[Reference10:Gateway/Interfaces#Conferencing interface .28CONF.29| ''CONF or SCNF'' interface]] consumes both resources and licenses. Here is an overview of which and how much.

=== Overview ===

First let us have a look at the overall scheme:

[[Image:Conferences_Ressources_and_Licenses_-_Scheme2_G722.png|conferences_ressources_and_licenses_-_scheme2_g722.png/|conferences_ressources_and_licenses_-_scheme2_g722.png/]]

==== Coder channels ====
Both the ''CONF'' and ''SCNF'' interface provides the mixing of 8KHz G711 PCM audio channels.

If a call is done towards a CONF interface, it will allocate a ''Coder channel'' from the DSP coder channel bank for processing of the audio from the VoIP codec (e.g. G.711, G.722, OPUS) to PCM audio.

The SCNF supports G.711a/u and G.722 coder. No ''Coder channel'' needs to be allocated for a call.

==== (Soft) Conference Channels ====
Each call to a CONF interface consumes one of the ''Conference Channels''.

Each call to an SCNF interface consumes one of the ''Soft Conference Channels''.

==== Channels license ====

In addition to that, a call to either the CONF or SCNF requires a ''Channel'' license. This license can be obtained from the PBX if the ''PBX Channels licenses'' switch in the PBX [[Reference13r1:PBX/Objects/Conference| ''Conference'' object]] is activated (if the call comes through such an object) or the ''Obtain Channels lic on outgoing call'' check-mark is ticked in a [[Reference13r1:PBX/Objects/Gateway| PBX ''Gateway'' object]] (if the call comes through such an object). For this, ''PBX Channels licenses'' must be installed on the PBX.

This is the recommended configuration. However, the ''Channel'' license can also be obtained locally from the gateway where the CONF/SCNF is located on. All innovaphone gateways have a number of ''Channel'' licenses built-in. The number of licenses available is equal to the number of ''Coder'' channels the box supports. If no license is sent along with the call to the CONF or SCNF interface, the interface will try to obtain one from the pool of built-in licenses. This can save you some cost. However, be aware that these licenses (as well as the corresponding ''Channel'' coders) are also required for calls through the ISDN BRI/PRI interfaces or for audio fax calls. Calls to CONF/SCNF interfaces which consume local ''Channel'' licenses may inhibit such calls therefor.

The PBX-Channels License has order no. 02-00020-007 according to chapter "3.5 PBX Channels license" in the [https://www.innovaphone.com/content/downloads/innovaphone-Licensing%20Guidelines-V13r2-EN.pdf innovaphone license guide].

The [[Reference13r1:PBX/Config/General#License Status|status of the PBX Licenses]] shows assigned PBX-Channel licenses:

[[Image:Conferences,_Ressources_and_Licenses_-_pbx-channel_license_01.png|conferences,_ressources_and_licenses_-_pbx-channel_license_01.png/|conferences,_ressources_and_licenses_-_pbx-channel_license_01.png/]]

==== Port license ====
The registration of the CONF and/or SCNF to the PBX Conference object does not require a Port license.

Although this is not directly related to the CONF or SCNF interface, please note that a single ''Port'' license is required if any of the rooms defined in a PBX ''Conference'' object is accessed using the conference web access (from 13r3).

==== CPU usage ====
For a calculation of CPU usage see [[Howto:V13_Firmware_Upgrade_V13r2_V13r3#Conferences]].

===Example Scenarios===

====Conference on hardware gateway with local ISDN====
Let us assume we have a conference running on a CONF interface with one PSTN and 3 VoIP participants. In this case, we have

{| border=1
| ||Caller || DSP Coder Channel || Conference Channel || PBX Channel License
|-
| ||1 PSTN || 1 (CONF) + 1 (ISDN) **|| 1 || 1
|-
| ||3 VoIP || 3|| 3 || 3
|-
| Total || 4 || 5 || 4 || 4
|}
<nowiki>**</nowiki>: {{FIXME|reason=do we need only one or would this require ''enable PCM''?}}

Looking at an IP411, it supports the following resources:

[[Image:Conferences,_Ressources_and_Licenses_-_example_gateway_channels_IP411.png|conferences,_ressources_and_licenses_-_example_gateway_channels_ip411.png/|conferences,_ressources_and_licenses_-_example_gateway_channels_ip411.png/]]

We can see that our sample scenario will not work on an IP411 as there are no ''Conference Channels'' available. However, if we change the scenario so that an SCNF is used instead of a CONF, it would look as follows:

{| border=1
| ||Caller || DSP Coder Channel || Soft Conference Channel || PBX Channel License
|-
| ||1 PSTN || 1 (ISDN) **|| 1 || 1
|-
| ||3 VoIP || 3|| 3 || 3
|-
| Total || 4 || 4 || 4 || 4
|}

This will work on an IP411.

=== 3PTY Conferencing ===
3-way conferencing is special as it is usually implemented internally in the innovaphone IP phones. That is, each user of an innovaphone IP phone (except the IP61) has internal conferencing resources built-in to allow for a 3-way conference. No external ''CONF/SCNF'' interface is needed.

However, on innovaphone DECT phones, no internal conferencing resource is present and hence no internal 3PTY is possible. The DECT system can thus be [[Reference9:3pty conference on DECT phones|configured to use an external conferencing resource]] for 3PTY. In this case, the rules above apply.

=== Conference Scaler ===
With 16r1 we introduce a new App "Conference Scaler", with the Conference Scaler it is possible to cluster multiple SCNF Interfaces into one registering Endpoint at a conference Object, for more please refer to [[Reference16r1:Concept App Service Conference Scaler| ''the concept article'']]

==== Ressource consideration ====
===== Conference Channels =====
If using the conference scaler and multiple interfaces you need to take into account to be sure how much conference channels are actually used, even if they are not be seen.
Let's have a look what happens for each room inside a Conference object that uses the conference scaler:

* As soon as a room has been opened on more than one interface (this will be the case when having at least 2 participants), the audio is only transmitted between the Interface and the locally connected participants
** To share audio between interfaces, the conference scaler will allocate a RTP Port on both interfaces
** This Socket will be exchanged between both interfaces, so that both interfaces know where to connect to
** Now, as both interfaces have the needed information, both interfaces will join to the other Interface as an "invisible" participant, taking up one more extra conference channel on both interfaces
* As we have learned in the concept article, the conference scaler will allocate one random interface as primary interface, this Primary interface will then be used to connect all used interfaces and mixing the audio. Let's go through this with our already existing scenario:
** Now we have two participants on two interfaces which are connected together ( and the conference scaler has already selected Interface 1 as the primary interface in the beginning of the conference call)
** Another participant joins the conference call, now let's assume we use 3 interfaces with the conference scaler, therefore this new participant will join at interface 3 and open this room there new
** As Interface 1 is the primary Interface, interface 3 will join the conference room ONLY at interface 1, taking up another conference channel as "invisible" participant (Interface 3 does not need to join interface 2 because the audio from interface 2 is already transmitted to interface 1, our primary interface)
** Interface 1 will also join interface 3 as "invisible" participant, taking up one conference channel on interface 3
* Now we have 2 Interfaces connected to interface 1 and interface 1 is connected to 2 interfaces as participant
* This is the flow for every conference room you are using at that conference object, for the next room the conference scaler might select Interface 3 as primary interface, in this case interface 3 will join both interfaces as participant and vice versa (also taking up conference channels here as well) - but only if this conference call has at least these 3 participants (as the interfaces will be used in a round robin manner)
* The primary interface will now receive audio from both interfaces, which will be the mixed audio from both interfaces, and distributes it to the other interfaces + it's own audio mixed together 

For Room 1 it will look like this:
{| class="wikitable"
|+
!Interface 1 (Primary Interface)
!Interface 2
!Interface 3
|-
|Participant 1
|Participant 2
|Participant 3
|-
|Interface 2 ("invisbile participant")
|Interface 1 ("invisbile participant")
|Interface 1 ("invisbile participant")
|-
|Interface 3 ("invisbile participant")
|
|
|}
Let's see what it looks like when you use another room at that conference object (Room 1 will be displayed grey, to demonstrate what it looks like with multiple rooms):

== Related Articles ==
* [[Reference9:Concept myPBX]]
* [[Reference10:Concept Innovaphone Virtual Appliance]]
* [[Reference9:3pty conference on DECT phones]]
* [[Reference10:Gateway/Interfaces#Conferencing_interface_.28CONF.29]]
* [[Howto:How_to_implement_large_PBXs#Technical_data_and_recommended_number_of_users_supported]], lists DSP and CONF resources for the different gateway platforms

[[Category:Howto|{{PAGENAME}}]]
[[Category:Concept Conference]]

Howto16r1:Conferences, Resources and Licenses

2026-04-30T11:15:05Z

Nwe: /* Conference Channels */

==Applies To==
This information applies to

all innovaphone devices supporting a ''CONF'' or ''SCNF'' interface and hence ''Conference Channels'' (as shown in the devices home-page).



==More Information==
Performing conferencing on a [[Reference10:Gateway/Interfaces#Conferencing interface .28CONF.29| ''CONF or SCNF'' interface]] consumes both resources and licenses. Here is an overview of which and how much.

=== Overview ===

First let us have a look at the overall scheme:

[[Image:Conferences_Ressources_and_Licenses_-_Scheme2_G722.png|conferences_ressources_and_licenses_-_scheme2_g722.png/|conferences_ressources_and_licenses_-_scheme2_g722.png/]]

==== Coder channels ====
Both the ''CONF'' and ''SCNF'' interface provides the mixing of 8KHz G711 PCM audio channels.

If a call is done towards a CONF interface, it will allocate a ''Coder channel'' from the DSP coder channel bank for processing of the audio from the VoIP codec (e.g. G.711, G.722, OPUS) to PCM audio.

The SCNF supports G.711a/u and G.722 coder. No ''Coder channel'' needs to be allocated for a call.

==== (Soft) Conference Channels ====
Each call to a CONF interface consumes one of the ''Conference Channels''.

Each call to an SCNF interface consumes one of the ''Soft Conference Channels''.

==== Channels license ====

In addition to that, a call to either the CONF or SCNF requires a ''Channel'' license. This license can be obtained from the PBX if the ''PBX Channels licenses'' switch in the PBX [[Reference13r1:PBX/Objects/Conference| ''Conference'' object]] is activated (if the call comes through such an object) or the ''Obtain Channels lic on outgoing call'' check-mark is ticked in a [[Reference13r1:PBX/Objects/Gateway| PBX ''Gateway'' object]] (if the call comes through such an object). For this, ''PBX Channels licenses'' must be installed on the PBX.

This is the recommended configuration. However, the ''Channel'' license can also be obtained locally from the gateway where the CONF/SCNF is located on. All innovaphone gateways have a number of ''Channel'' licenses built-in. The number of licenses available is equal to the number of ''Coder'' channels the box supports. If no license is sent along with the call to the CONF or SCNF interface, the interface will try to obtain one from the pool of built-in licenses. This can save you some cost. However, be aware that these licenses (as well as the corresponding ''Channel'' coders) are also required for calls through the ISDN BRI/PRI interfaces or for audio fax calls. Calls to CONF/SCNF interfaces which consume local ''Channel'' licenses may inhibit such calls therefor.

The PBX-Channels License has order no. 02-00020-007 according to chapter "3.5 PBX Channels license" in the [https://www.innovaphone.com/content/downloads/innovaphone-Licensing%20Guidelines-V13r2-EN.pdf innovaphone license guide].

The [[Reference13r1:PBX/Config/General#License Status|status of the PBX Licenses]] shows assigned PBX-Channel licenses:

[[Image:Conferences,_Ressources_and_Licenses_-_pbx-channel_license_01.png|conferences,_ressources_and_licenses_-_pbx-channel_license_01.png/|conferences,_ressources_and_licenses_-_pbx-channel_license_01.png/]]

==== Port license ====
The registration of the CONF and/or SCNF to the PBX Conference object does not require a Port license.

Although this is not directly related to the CONF or SCNF interface, please note that a single ''Port'' license is required if any of the rooms defined in a PBX ''Conference'' object is accessed using the conference web access (from 13r3).

==== CPU usage ====
For a calculation of CPU usage see [[Howto:V13_Firmware_Upgrade_V13r2_V13r3#Conferences]].

===Example Scenarios===

====Conference on hardware gateway with local ISDN====
Let us assume we have a conference running on a CONF interface with one PSTN and 3 VoIP participants. In this case, we have

{| border=1
| ||Caller || DSP Coder Channel || Conference Channel || PBX Channel License
|-
| ||1 PSTN || 1 (CONF) + 1 (ISDN) **|| 1 || 1
|-
| ||3 VoIP || 3|| 3 || 3
|-
| Total || 4 || 5 || 4 || 4
|}
<nowiki>**</nowiki>: {{FIXME|reason=do we need only one or would this require ''enable PCM''?}}

Looking at an IP411, it supports the following resources:

[[Image:Conferences,_Ressources_and_Licenses_-_example_gateway_channels_IP411.png|conferences,_ressources_and_licenses_-_example_gateway_channels_ip411.png/|conferences,_ressources_and_licenses_-_example_gateway_channels_ip411.png/]]

We can see that our sample scenario will not work on an IP411 as there are no ''Conference Channels'' available. However, if we change the scenario so that an SCNF is used instead of a CONF, it would look as follows:

{| border=1
| ||Caller || DSP Coder Channel || Soft Conference Channel || PBX Channel License
|-
| ||1 PSTN || 1 (ISDN) **|| 1 || 1
|-
| ||3 VoIP || 3|| 3 || 3
|-
| Total || 4 || 4 || 4 || 4
|}

This will work on an IP411.

=== 3PTY Conferencing ===
3-way conferencing is special as it is usually implemented internally in the innovaphone IP phones. That is, each user of an innovaphone IP phone (except the IP61) has internal conferencing resources built-in to allow for a 3-way conference. No external ''CONF/SCNF'' interface is needed.

However, on innovaphone DECT phones, no internal conferencing resource is present and hence no internal 3PTY is possible. The DECT system can thus be [[Reference9:3pty conference on DECT phones|configured to use an external conferencing resource]] for 3PTY. In this case, the rules above apply.

=== Conference Scaler ===
With 16r1 we introduce a new App "Conference Scaler", with the Conference Scaler it is possible to cluster multiple SCNF Interfaces into one registering Endpoint at a conference Object, for more please refer to [[Reference16r1:Concept App Service Conference Scaler| ''the concept article'']]

==== Ressource consideration ====
===== Conference Channels =====
If using the conference scaler and multiple interfaces you need to take into account to be sure how much conference channels are actually used, even if they are not be seen.
Let's have a look what happens for each room inside a Conference object that uses the conference scaler:

* As soon as a room has been opened on more than one interface (this will be the case when having at least 2 participants), the audio is only transmitted between the Interface and the locally connected participants
** To share audio between interfaces, the conference scaler will allocate a RTP Port on both interfaces
** This Socket will be exchanged between both interfaces, so that both interfaces know where to connect to
** Now, as both interfaces have the needed information, both interfaces will join to the other Interface as an "invisible" participant, taking up one more extra conference channel on both interfaces
* As we have learned in the concept article, the conference scaler will allocate one random interface as primary interface, this Primary interface will then be used to connect all used interfaces and mixing the audio. Let's go through this with our already existing scenario:
** Now we have two participants on two interfaces which are connected together ( and the conference scaler has already selected Interface 1 as the primary interface in the beginning of the conference call)
** Another participant joins the conference call, now let's assume we use 3 interfaces with the conference scaler, therefore this new participant will join at interface 3 and open this room there new
** As Interface 1 is the primary Interface, interface 3 will join the conference room ONLY at interface 1, taking up another conference channel as "invisible" participant (Interface 3 does not need to join interface 2 because the audio from interface 2 is already transmitted to interface 1, our primary interface)
** Interface 1 will also join interface 3 as "invisible" participant, taking up one conference channel on interface 3
* Now we have 2 Interfaces connected to interface 1 and interface 1 is connected to 2 interfaces as participant
* This is the flow for every conference room you are using at that conference object, for the next room the conference scaler might select Interface 3 as primary interface, in this case interface 3 will join both interfaces as participant and vice versa (also taking up conference channels here as well) - but only if this conference call has at least these 3 participants (as the interfaces will be used in a round robin manner)
* The primary interface will now receive audio from both interfaces, which will be the mixed audio from both interfaces, and distributes it to the other interfaces + it's own audio mixed together 

For Room 1 it will look like this:
{| class="wikitable"
|+
!Interface 1 (Primary Interface)
!Interface 2
!Interface 3
|-
|Participant 1
|Participant 2
|Participant 3
|-
|Interface 2 ("invisbile participant")
|Interface 1 ("invisbile participant")
|Interface 1 ("invisbile participant")
|-
|Interface 3 ("invisbile participant")
|
|
|}

== Related Articles ==
* [[Reference9:Concept myPBX]]
* [[Reference10:Concept Innovaphone Virtual Appliance]]
* [[Reference9:3pty conference on DECT phones]]
* [[Reference10:Gateway/Interfaces#Conferencing_interface_.28CONF.29]]
* [[Howto:How_to_implement_large_PBXs#Technical_data_and_recommended_number_of_users_supported]], lists DSP and CONF resources for the different gateway platforms

[[Category:Howto|{{PAGENAME}}]]
[[Category:Concept Conference]]

Howto16r1:Conferences, Resources and Licenses

2026-04-30T08:24:40Z

Nwe: /* Conference Channels */

==Applies To==
This information applies to

all innovaphone devices supporting a ''CONF'' or ''SCNF'' interface and hence ''Conference Channels'' (as shown in the devices home-page).



==More Information==
Performing conferencing on a [[Reference10:Gateway/Interfaces#Conferencing interface .28CONF.29| ''CONF or SCNF'' interface]] consumes both resources and licenses. Here is an overview of which and how much.

=== Overview ===

First let us have a look at the overall scheme:

[[Image:Conferences_Ressources_and_Licenses_-_Scheme2_G722.png|conferences_ressources_and_licenses_-_scheme2_g722.png/|conferences_ressources_and_licenses_-_scheme2_g722.png/]]

==== Coder channels ====
Both the ''CONF'' and ''SCNF'' interface provides the mixing of 8KHz G711 PCM audio channels.

If a call is done towards a CONF interface, it will allocate a ''Coder channel'' from the DSP coder channel bank for processing of the audio from the VoIP codec (e.g. G.711, G.722, OPUS) to PCM audio.

The SCNF supports G.711a/u and G.722 coder. No ''Coder channel'' needs to be allocated for a call.

==== (Soft) Conference Channels ====
Each call to a CONF interface consumes one of the ''Conference Channels''.

Each call to an SCNF interface consumes one of the ''Soft Conference Channels''.

==== Channels license ====

In addition to that, a call to either the CONF or SCNF requires a ''Channel'' license. This license can be obtained from the PBX if the ''PBX Channels licenses'' switch in the PBX [[Reference13r1:PBX/Objects/Conference| ''Conference'' object]] is activated (if the call comes through such an object) or the ''Obtain Channels lic on outgoing call'' check-mark is ticked in a [[Reference13r1:PBX/Objects/Gateway| PBX ''Gateway'' object]] (if the call comes through such an object). For this, ''PBX Channels licenses'' must be installed on the PBX.

This is the recommended configuration. However, the ''Channel'' license can also be obtained locally from the gateway where the CONF/SCNF is located on. All innovaphone gateways have a number of ''Channel'' licenses built-in. The number of licenses available is equal to the number of ''Coder'' channels the box supports. If no license is sent along with the call to the CONF or SCNF interface, the interface will try to obtain one from the pool of built-in licenses. This can save you some cost. However, be aware that these licenses (as well as the corresponding ''Channel'' coders) are also required for calls through the ISDN BRI/PRI interfaces or for audio fax calls. Calls to CONF/SCNF interfaces which consume local ''Channel'' licenses may inhibit such calls therefor.

The PBX-Channels License has order no. 02-00020-007 according to chapter "3.5 PBX Channels license" in the [https://www.innovaphone.com/content/downloads/innovaphone-Licensing%20Guidelines-V13r2-EN.pdf innovaphone license guide].

The [[Reference13r1:PBX/Config/General#License Status|status of the PBX Licenses]] shows assigned PBX-Channel licenses:

[[Image:Conferences,_Ressources_and_Licenses_-_pbx-channel_license_01.png|conferences,_ressources_and_licenses_-_pbx-channel_license_01.png/|conferences,_ressources_and_licenses_-_pbx-channel_license_01.png/]]

==== Port license ====
The registration of the CONF and/or SCNF to the PBX Conference object does not require a Port license.

Although this is not directly related to the CONF or SCNF interface, please note that a single ''Port'' license is required if any of the rooms defined in a PBX ''Conference'' object is accessed using the conference web access (from 13r3).

==== CPU usage ====
For a calculation of CPU usage see [[Howto:V13_Firmware_Upgrade_V13r2_V13r3#Conferences]].

===Example Scenarios===

====Conference on hardware gateway with local ISDN====
Let us assume we have a conference running on a CONF interface with one PSTN and 3 VoIP participants. In this case, we have

{| border=1
| ||Caller || DSP Coder Channel || Conference Channel || PBX Channel License
|-
| ||1 PSTN || 1 (CONF) + 1 (ISDN) **|| 1 || 1
|-
| ||3 VoIP || 3|| 3 || 3
|-
| Total || 4 || 5 || 4 || 4
|}
<nowiki>**</nowiki>: {{FIXME|reason=do we need only one or would this require ''enable PCM''?}}

Looking at an IP411, it supports the following resources:

[[Image:Conferences,_Ressources_and_Licenses_-_example_gateway_channels_IP411.png|conferences,_ressources_and_licenses_-_example_gateway_channels_ip411.png/|conferences,_ressources_and_licenses_-_example_gateway_channels_ip411.png/]]

We can see that our sample scenario will not work on an IP411 as there are no ''Conference Channels'' available. However, if we change the scenario so that an SCNF is used instead of a CONF, it would look as follows:

{| border=1
| ||Caller || DSP Coder Channel || Soft Conference Channel || PBX Channel License
|-
| ||1 PSTN || 1 (ISDN) **|| 1 || 1
|-
| ||3 VoIP || 3|| 3 || 3
|-
| Total || 4 || 4 || 4 || 4
|}

This will work on an IP411.

=== 3PTY Conferencing ===
3-way conferencing is special as it is usually implemented internally in the innovaphone IP phones. That is, each user of an innovaphone IP phone (except the IP61) has internal conferencing resources built-in to allow for a 3-way conference. No external ''CONF/SCNF'' interface is needed.

However, on innovaphone DECT phones, no internal conferencing resource is present and hence no internal 3PTY is possible. The DECT system can thus be [[Reference9:3pty conference on DECT phones|configured to use an external conferencing resource]] for 3PTY. In this case, the rules above apply.

=== Conference Scaler ===
With 16r1 we introduce a new App "Conference Scaler", with the Conference Scaler it is possible to cluster multiple SCNF Interfaces into one registering Endpoint at a conference Object, for more please refer to [[Reference16r1:Concept App Service Conference Scaler| ''the concept article'']]

==== Ressource consideration ====
===== Conference Channels =====
If using the conference scaler and multiple interfaces you need to take into account to be sure how much conference channels are actually used, even if they are not be seen.
Let's have a look what happens for each room inside a Conference object that uses the conference scaler:

* As soon as a room has been opened on more than one interface (this will be the case when having at least 2 participants), the audio is only transmitted between the Interface and the locally connected participants
** To share audio between interfaces, the conference scaler will allocate a RTP Port on both interfaces
** This Socket will be exchanged between both interfaces, so that both interfaces know where to connect to
** Now, as both interfaces have the needed information, both interfaces will join to the other Interface as an "invisible" participant, taking up one more extra conference channel on both interfaces
* As we have learned in the concept article, the conference scaler will allocate one random interface as primary interface, this Primary interface will then be used to connect all used interfaces and mixing the audio. Let's go through this with our already existing scenario:
** Now we have two participants on two interfaces which are connected together ( and the conference scaler has already selected Interface 1 as the primary interface in the beginning of the conference call)
** Another participant joins the conference call, therefore this room will also be opened at interface 3
** As Interface 1 is the primary Interface, interface 3 will join the conference room ONLY at interface 1, taking up another conference channel as "invisible" participant (Interface 3 does not need to join interface 2 because the audio from interface 2 is already transmitted to interface 1, our primary interface)
** Interface 1 will also join interface 3 as "invisible" participant, taking up one conference channel on interface 3
* Now we have 2 Interfaces connected to interface 1 and interface 1 is connected to 2 interfaces as participant
* This is the flow for every conference room you are using at that conference object, for the next room the conference scaler might select Interface 3 as primary interface, in this case interface 3 will join both interfaces as participant and vice versa (also taking up conference channels here as well) - but only if this conference call has at least these 3 participants (as the interfaces will be used in a round robin manner)
* The primary interface will now receive audio from both interfaces, which will be the mixed audio from both interfaces, and distributes it to the other interfaces + it's own audio mixed together 

For Room 1 it will look like this:
{| class="wikitable"
|+
!Interface 1 (Primary Interface)
!Interface 2
!Interface 3
|-
|Participant 1
|Participant 2
|Participant 3
|-
|Interface 2 ("invisbile participant")
|Interface 1 ("invisbile participant")
|Interface 1 ("invisbile participant")
|-
|Interface 3 ("invisbile participant")
|
|
|}

== Related Articles ==
* [[Reference9:Concept myPBX]]
* [[Reference10:Concept Innovaphone Virtual Appliance]]
* [[Reference9:3pty conference on DECT phones]]
* [[Reference10:Gateway/Interfaces#Conferencing_interface_.28CONF.29]]
* [[Howto:How_to_implement_large_PBXs#Technical_data_and_recommended_number_of_users_supported]], lists DSP and CONF resources for the different gateway platforms

[[Category:Howto|{{PAGENAME}}]]
[[Category:Concept Conference]]

Howto16r1:Conferences, Resources and Licenses

2026-04-24T11:41:08Z

Nwe: /* Conference Channels */

{{FIXME|reason=This product is in the beta phase and is not yet finished}}

==Applies To==
This information applies to

all innovaphone devices supporting a ''CONF'' or ''SCNF'' interface and hence ''Conference Channels'' (as shown in the devices home-page).



==More Information==
Performing conferencing on a [[Reference10:Gateway/Interfaces#Conferencing interface .28CONF.29| ''CONF or SCNF'' interface]] consumes both resources and licenses. Here is an overview of which and how much.

=== Overview ===

First let us have a look at the overall scheme:

[[Image:Conferences_Ressources_and_Licenses_-_Scheme2_G722.png|conferences_ressources_and_licenses_-_scheme2_g722.png/|conferences_ressources_and_licenses_-_scheme2_g722.png/]]

==== Coder channels ====
Both the ''CONF'' and ''SCNF'' interface provides the mixing of 8KHz G711 PCM audio channels.

If a call is done towards a CONF interface, it will allocate a ''Coder channel'' from the DSP coder channel bank for processing of the audio from the VoIP codec (e.g. G.711, G.722, OPUS) to PCM audio.

The SCNF supports G.711a/u and G.722 coder. No ''Coder channel'' needs to be allocated for a call.

==== (Soft) Conference Channels ====
Each call to a CONF interface consumes one of the ''Conference Channels''.

Each call to an SCNF interface consumes one of the ''Soft Conference Channels''.

==== Channels license ====

In addition to that, a call to either the CONF or SCNF requires a ''Channel'' license. This license can be obtained from the PBX if the ''PBX Channels licenses'' switch in the PBX [[Reference13r1:PBX/Objects/Conference| ''Conference'' object]] is activated (if the call comes through such an object) or the ''Obtain Channels lic on outgoing call'' check-mark is ticked in a [[Reference13r1:PBX/Objects/Gateway| PBX ''Gateway'' object]] (if the call comes through such an object). For this, ''PBX Channels licenses'' must be installed on the PBX.

This is the recommended configuration. However, the ''Channel'' license can also be obtained locally from the gateway where the CONF/SCNF is located on. All innovaphone gateways have a number of ''Channel'' licenses built-in. The number of licenses available is equal to the number of ''Coder'' channels the box supports. If no license is sent along with the call to the CONF or SCNF interface, the interface will try to obtain one from the pool of built-in licenses. This can save you some cost. However, be aware that these licenses (as well as the corresponding ''Channel'' coders) are also required for calls through the ISDN BRI/PRI interfaces or for audio fax calls. Calls to CONF/SCNF interfaces which consume local ''Channel'' licenses may inhibit such calls therefor.

The PBX-Channels License has order no. 02-00020-007 according to chapter "3.5 PBX Channels license" in the [https://www.innovaphone.com/content/downloads/innovaphone-Licensing%20Guidelines-V13r2-EN.pdf innovaphone license guide].

The [[Reference13r1:PBX/Config/General#License Status|status of the PBX Licenses]] shows assigned PBX-Channel licenses:

[[Image:Conferences,_Ressources_and_Licenses_-_pbx-channel_license_01.png|conferences,_ressources_and_licenses_-_pbx-channel_license_01.png/|conferences,_ressources_and_licenses_-_pbx-channel_license_01.png/]]

==== Port license ====
The registration of the CONF and/or SCNF to the PBX Conference object does not require a Port license.

Although this is not directly related to the CONF or SCNF interface, please note that a single ''Port'' license is required if any of the rooms defined in a PBX ''Conference'' object is accessed using the conference web access (from 13r3).

==== CPU usage ====
For a calculation of CPU usage see [[Howto:V13_Firmware_Upgrade_V13r2_V13r3#Conferences]].

===Example Scenarios===

====Conference on hardware gateway with local ISDN====
Let us assume we have a conference running on a CONF interface with one PSTN and 3 VoIP participants. In this case, we have

{| border=1
| ||Caller || DSP Coder Channel || Conference Channel || PBX Channel License
|-
| ||1 PSTN || 1 (CONF) + 1 (ISDN) **|| 1 || 1
|-
| ||3 VoIP || 3|| 3 || 3
|-
| Total || 4 || 5 || 4 || 4
|}
<nowiki>**</nowiki>: {{FIXME|reason=do we need only one or would this require ''enable PCM''?}}

Looking at an IP411, it supports the following resources:

[[Image:Conferences,_Ressources_and_Licenses_-_example_gateway_channels_IP411.png|conferences,_ressources_and_licenses_-_example_gateway_channels_ip411.png/|conferences,_ressources_and_licenses_-_example_gateway_channels_ip411.png/]]

We can see that our sample scenario will not work on an IP411 as there are no ''Conference Channels'' available. However, if we change the scenario so that an SCNF is used instead of a CONF, it would look as follows:

{| border=1
| ||Caller || DSP Coder Channel || Soft Conference Channel || PBX Channel License
|-
| ||1 PSTN || 1 (ISDN) **|| 1 || 1
|-
| ||3 VoIP || 3|| 3 || 3
|-
| Total || 4 || 4 || 4 || 4
|}

This will work on an IP411.

=== 3PTY Conferencing ===
3-way conferencing is special as it is usually implemented internally in the innovaphone IP phones. That is, each user of an innovaphone IP phone (except the IP61) has internal conferencing resources built-in to allow for a 3-way conference. No external ''CONF/SCNF'' interface is needed.

However, on innovaphone DECT phones, no internal conferencing resource is present and hence no internal 3PTY is possible. The DECT system can thus be [[Reference9:3pty conference on DECT phones|configured to use an external conferencing resource]] for 3PTY. In this case, the rules above apply.

=== Conference Scaler ===
With 16r1 we introduce a new App "Conference Scaler", with the Conference Scaler it is possible to cluster multiple SCNF Interfaces into one registering Endpoint at a conference Object, for more please refer to [[Reference16r1:Concept App Service Conference Scaler| ''the concept article'']]

==== Ressource consideration ====
===== Conference Channels =====
If using the conference scaler and multiple interfaces you need to take into account to be sure how much conference channels are actually used, even if they are not be seen.
Let's have a look what happens for each room inside a Conference object that uses the conference scaler:

* As soon as a room has been opened on more than one interface (this will be the case when having at least 2 participants), the audio is only transmitted between the Interface and the locally connected participants
** To share audio between interfaces, the conference scaler will allocate a RTP Port on both interfaces
** This Socket will be exchanged between both interfaces, so that both interfaces know where to connect to
** Now, as both interfaces have the needed information, both interfaces will join to the other Interface as an "invisible" participant, taking up one more extra conference channel on both interfaces
* As we have learned in the concept article, the conference scaler will allocate one random interface as primary interface, this Primary interface will then be used to connect all used interfaces and mixing the audio. Let's go through this with our already existing scenario:
** Now we have two participants on two interfaces which are connected together ( and the conference scaler has already selected Interface 1 as the primary interface in the beginning of the conference call)
** Another participant joins the conference call, therefore this room will also be opened at interface 3
** As Interface 1 is the primary Interface, interface 3 will join the conference room ONLY at interface 1, taking up another conference channel as "invisible" participant (Interface 3 does not need to join interface 2 because the audio from interface 2 is already transmitted to interface 1, our primary interface)
** Interface 1 will also join interface 3 as "invisible" participant, taking up one conference channel on interface 3
* Now we have 2 Interfaces connected to interface 1 and interface 1 is connected to 2 interfaces as participant
* This is the flow for every conference room you are using at that conference object, for the next room the conference scaler might select Interface 3 as primary interface, in this case he will join both interfaces as participant and vice versa (also taking up conference channels here as well) - but only if this conference call has at least these 3 participants (as the interfaces will be used in a round robin manner)
* The primary inertface will now receive audio from both interfaces, which will be the mixed audio from both interfaces, and distributes it to the other interfaces + it's own audio mixed together 

For Room 1 it will look like this:
{| class="wikitable"
|+
!Interface 1 (Primary Interface)
!Interface 2
!Interface 3
|-
|Participant 1
|Participant 2
|Participant 3
|-
|Interface 2 ("invisbile participant")
|Interface 1 ("invisbile participant")
|Interface 1 ("invisbile participant")
|-
|Interface 3 ("invisbile participant")
|
|
|}

== Related Articles ==
* [[Reference9:Concept myPBX]]
* [[Reference10:Concept Innovaphone Virtual Appliance]]
* [[Reference9:3pty conference on DECT phones]]
* [[Reference10:Gateway/Interfaces#Conferencing_interface_.28CONF.29]]
* [[Howto:How_to_implement_large_PBXs#Technical_data_and_recommended_number_of_users_supported]], lists DSP and CONF resources for the different gateway platforms

[[Category:Howto|{{PAGENAME}}]]
[[Category:Concept Conference]]

Howto16r1:Conferences, Resources and Licenses

2026-04-24T11:36:35Z

Nwe: /* Conference Channels */

{{FIXME|reason=This product is in the beta phase and is not yet finished}}

==Applies To==
This information applies to

all innovaphone devices supporting a ''CONF'' or ''SCNF'' interface and hence ''Conference Channels'' (as shown in the devices home-page).



==More Information==
Performing conferencing on a [[Reference10:Gateway/Interfaces#Conferencing interface .28CONF.29| ''CONF or SCNF'' interface]] consumes both resources and licenses. Here is an overview of which and how much.

=== Overview ===

First let us have a look at the overall scheme:

[[Image:Conferences_Ressources_and_Licenses_-_Scheme2_G722.png|conferences_ressources_and_licenses_-_scheme2_g722.png/]]

==== Coder channels ====
Both the ''CONF'' and ''SCNF'' interface provides the mixing of 8KHz G711 PCM audio channels.

If a call is done towards a CONF interface, it will allocate a ''Coder channel'' from the DSP coder channel bank for processing of the audio from the VoIP codec (e.g. G.711, G.722, OPUS) to PCM audio.

The SCNF supports G.711a/u and G.722 coder. No ''Coder channel'' needs to be allocated for a call.

==== (Soft) Conference Channels ====
Each call to a CONF interface consumes one of the ''Conference Channels''.

Each call to an SCNF interface consumes one of the ''Soft Conference Channels''.

==== Channels license ====

In addition to that, a call to either the CONF or SCNF requires a ''Channel'' license. This license can be obtained from the PBX if the ''PBX Channels licenses'' switch in the PBX [[Reference13r1:PBX/Objects/Conference| ''Conference'' object]] is activated (if the call comes through such an object) or the ''Obtain Channels lic on outgoing call'' check-mark is ticked in a [[Reference13r1:PBX/Objects/Gateway| PBX ''Gateway'' object]] (if the call comes through such an object). For this, ''PBX Channels licenses'' must be installed on the PBX.

This is the recommended configuration. However, the ''Channel'' license can also be obtained locally from the gateway where the CONF/SCNF is located on. All innovaphone gateways have a number of ''Channel'' licenses built-in. The number of licenses available is equal to the number of ''Coder'' channels the box supports. If no license is sent along with the call to the CONF or SCNF interface, the interface will try to obtain one from the pool of built-in licenses. This can save you some cost. However, be aware that these licenses (as well as the corresponding ''Channel'' coders) are also required for calls through the ISDN BRI/PRI interfaces or for audio fax calls. Calls to CONF/SCNF interfaces which consume local ''Channel'' licenses may inhibit such calls therefor.

The PBX-Channels License has order no. 02-00020-007 according to chapter "3.5 PBX Channels license" in the [https://www.innovaphone.com/content/downloads/innovaphone-Licensing%20Guidelines-V13r2-EN.pdf innovaphone license guide].

The [[Reference13r1:PBX/Config/General#License Status|status of the PBX Licenses]] shows assigned PBX-Channel licenses:

[[Image:Conferences,_Ressources_and_Licenses_-_pbx-channel_license_01.png|conferences,_ressources_and_licenses_-_pbx-channel_license_01.png/]]

==== Port license ====
The registration of the CONF and/or SCNF to the PBX Conference object does not require a Port license.

Although this is not directly related to the CONF or SCNF interface, please note that a single ''Port'' license is required if any of the rooms defined in a PBX ''Conference'' object is accessed using the conference web access (from 13r3).

==== CPU usage ====
For a calculation of CPU usage see [[Howto:V13_Firmware_Upgrade_V13r2_V13r3#Conferences]].

===Example Scenarios===

====Conference on hardware gateway with local ISDN====
Let us assume we have a conference running on a CONF interface with one PSTN and 3 VoIP participants. In this case, we have

{| border=1
| ||Caller || DSP Coder Channel || Conference Channel || PBX Channel License
|-
| ||1 PSTN || 1 (CONF) + 1 (ISDN) **|| 1 || 1
|-
| ||3 VoIP || 3|| 3 || 3
|-
| Total || 4 || 5 || 4 || 4
|}
<nowiki>**</nowiki>: {{FIXME|reason=do we need only one or would this require ''enable PCM''?}}

Looking at an IP411, it supports the following resources:

[[Image:Conferences,_Ressources_and_Licenses_-_example_gateway_channels_IP411.png|conferences,_ressources_and_licenses_-_example_gateway_channels_ip411.png/]]

We can see that our sample scenario will not work on an IP411 as there are no ''Conference Channels'' available. However, if we change the scenario so that an SCNF is used instead of a CONF, it would look as follows:

{| border=1
| ||Caller || DSP Coder Channel || Soft Conference Channel || PBX Channel License
|-
| ||1 PSTN || 1 (ISDN) **|| 1 || 1
|-
| ||3 VoIP || 3|| 3 || 3
|-
| Total || 4 || 4 || 4 || 4
|}

This will work on an IP411.

=== 3PTY Conferencing ===
3-way conferencing is special as it is usually implemented internally in the innovaphone IP phones. That is, each user of an innovaphone IP phone (except the IP61) has internal conferencing resources built-in to allow for a 3-way conference. No external ''CONF/SCNF'' interface is needed.

However, on innovaphone DECT phones, no internal conferencing resource is present and hence no internal 3PTY is possible. The DECT system can thus be [[Reference9:3pty conference on DECT phones|configured to use an external conferencing resource]] for 3PTY. In this case, the rules above apply.

=== Conference Scaler ===
With 16r1 we introduce a new App "Conference Scaler", with the Conference Scaler it is possible to cluster multiple SCNF Interfaces into one registering Endpoint at a conference Object, for more please refer to [[Reference16r1:Concept App Service Conference Scaler| ''the concept article'']]

==== Ressource consideration ====
===== Conference Channels =====
If using the conference scaler and multiple interfaces you need to take into account to be sure how much conference channels are actually used, even if they are not be seen.
Let's have a look what happens for each room inside a Conference object that uses the conference scaler:

* As soon as a room has been opened on more than one interface (this will be the case when having at least 2 participants), the audio is only transmitted between the Interface and the locally connected participants
** To share audio between interfaces, the conference scaler will allocate a RTP Port on both interfaces
** This Socket will be exchanged between both interfaces, so that both interfaces know where to connect to
** Now, as both interfaces have the needed information, both interfaces will join to the other Interface as an "invisible" participant, taking up one more extra conference channel on both interfaces
* As we have learned in the concept article, the conference scaler will allocate one random interface as primary interface, this Primary interface will then be used to connect all used interfaces and mixing the audio. Let's go through this with our already existing scenario:
** Now we have two participants on two interfaces which are connected together ( and the conference scaler has already selected Interface 1 as the primary interface in the beginning of the conference call)
** Another participant joins the conference call, therefore this room will also be opened at interface 3
** As Interface 1 is the primary Interface, interface 3 will join the conference room ONLY at interface 1, taking up another conference channel as "invisible" participant (Interface 3 does not need to join interface 2 because the audio from interface 2 is already transmitted to interface 1, our primary interface)
** Interface 1 will also join interface 3 as "invisible" participant, taking up one conference channel on interface 3
* Now we have 2 Interfaces connected to interface 1 and interface 1 is connected to 2 interfaces as participant
* This is the flow for every conference room you are using at that conference object, for the next room the conference scaler might select Interface 3 as primary interface, in this case he will join both interfaces as participant and vice versa (also taking up conference channels here as well) - but only if this conference call has at least these 3 participants (as the interfaces will be used in a round robin manner) 

== Related Articles ==
* [[Reference9:Concept myPBX]]
* [[Reference10:Concept Innovaphone Virtual Appliance]]
* [[Reference9:3pty conference on DECT phones]]
* [[Reference10:Gateway/Interfaces#Conferencing_interface_.28CONF.29]]
* [[Howto:How_to_implement_large_PBXs#Technical_data_and_recommended_number_of_users_supported]], lists DSP and CONF resources for the different gateway platforms

[[Category:Howto|{{PAGENAME}}]]
[[Category:Concept Conference]]

Howto16r1:Conferences, Resources and Licenses

2026-04-24T11:17:33Z

Nwe: Created page with "{{FIXME|reason=This product is in the beta phase and is not yet finished}} ==Applies To== This information applies to all innovaphone devices supporting a ''CONF'' or ''SCNF'' interface and hence ''Conference Channels'' (as shown in the devices home-page).  ==More Information== Performing conferencing on a..."

Reference14r2:Concept innovaphone Portable Platform

2026-04-20T14:57:24Z

Nwe: /* App Platform concept */

===innovaphone Portable Platform===

==What is iPP? (innovaphone Portable Platform)==
The iPP is a concept to demonstrate how a mobile communication solution based on the innovaphone system may look like. It provides the initial starting point to develop own concepts, ideas, solutions or products.
The requirements & the area of use are influencing factors, there are many possibilities and options to have it as dynamically as possible!

[[File:ipp_suitcase_2.jpg|ipp_suitcase_2.jpg/]]

In this picture you can see what this solution '''could''' look like, let's be creative and say: why have a Suitcase for that? We can even have a mobile solution built into a Car! As you can see you can do a lot with this Solution, and it also depends on your scenario, needs, use case and so on.

There are some things you have to consider, but I will get back to that later on.

==Where can it be useful?==
You can use the iPP as you desire. Here are some examples to show you where you might need to have an iPP:

*Secure communication system in disaster areas (e.g. Ahr Valley, Germany)
*Secure communication system at big events
*Secure communication system to reliably secure supply
*Secure communication system for critical infrastructures
*Secure communication system on a (big) construction site
*Do you have any more ideas?

The big advantage of the iPP is: you can use it nearly anywhere and design it specially for your needs and what you consider as important!

==Configuration scenario==
Depending on your needs and your infrastructure, you can have two scenarios to set up the iPP.

===Stand-Alone iPP===
*Autonomous communication solution without Master / Slave set-up
*Via internet connection
**Own trunk line possible
**Audio conferences with up to 100 internal participants (depending on your used Hardware). You can also have external participants depending on your used Hardware/Software!

'''Regarding external access:''' As a Stand-Alone iPP it might not be possible to have a fixed external IP-Address, depending on what kind of Data connection you use and what your ISP can offer. Although it would be possible to have DynDNS, this also relies on what kind of Hardware/Software you have deployed. Thus, it is dependent on these factors if you can have external access too.

===Master / Slave iPP===
*Communication solution with Master / Slave set-up
*Can also be operated as standalone if there is no internet connection available, for example
*Via internet connection
**Connection to Master or additional Slave systems available (Let's say you have a huge area to cover, then you might need more than one System on the area)
**Own and / or central trunk line, possible (you can keep the trunk redundant with central and also dedicated trunks)
**Audio conferences with up to 100 internal and external participants

'''Regarding external access for the Slaves:''' For our Scenario we used a VPN Tunnel from the Slaves to the HQ (with a fixed IP-Address in our Data center), all DNS-Queries for the Slaves then points to the HQ where the Reverse-Proxy then routes the Traffic accordingly to the right System. '''BUT''' in this case you will need a HQ, keep in mind that external access will also be down if the HQ (or the connection to the HQ) should be down.

==Use case scenario==
To give you a better feeling on how to deploy the iPP I will give you a use case scenario:

*Imagine a flooding of a whole suburban city
*The flooding has nearly destroyed everything (Streets, Infrastructure, Cellular towers, Houses, Electricity...)
*With the iPP you have control over the whole area and cover the whole Crisis area

[[File:ipp_crisis_area.png|ipp_crisis_area.png/]]

*Now Police, Ambulance, Firefighters, THW (Technisches Hilfswerk), volunteers and many others arrive at the spot
*With the iPP you can combine the communication between every Person there, this way everyone have the same source of information and is connected together
*You can have a connection to a Headquarter where you can ask for supplies or more help
*One site can request help from another site

[[File:ipp_connect.png|ipp_connect.png/]]

*You can additionally add Maps as an App so that Users can orientate if they're not familiar with the area

[[File:ipp_maps.png|ipp_maps.png/]]

*You can have emergency meetings

[[File:ipp_conferencing.png|ipp_conferencing.png/]]

*It's possible to give status updates to the Headquarter or to another location

And these are only a few options you have to secure a stable and secure communication solution in such situation.

==Things to have in consideration==
As I mentioned before, a lot of things depends on the situation you want to cover and possibilities you have. There are still some Key-things that you have to take in account when deploying an iPP.

The two main points which needs to be covered are:

===Power Supply===
Of course, you will need electricity to have this System working, Sometimes you have Low-voltage-network on spot but what if not? Let's take a look at the options!

*As already mentioned, depending on the situation and use case you can use Low-voltage-network to get electricity (If available!)
*Another option can be via battery network with following charging options:
**Solar
**Wind energy
**Generator (Diesel, gasoline, propane)

===Data connection===
Depending on if you want to have external connection too, you need a way for Data connection. In order for this too work, you have multiple options:

*Ethernet, if available you can use Ethernet on side but depending on the situation and availability it might not be available on side
*LTE/4G/5G Router
*Satellite Internet
*450MHz Network (Available in Germany from 2025, especially designed for critical Infrastructure)

==App Platform concept==
To provide Apps globally and also dedicated, you can have several App Platforms in your scenario.

For example, let's take a look at some Apps:

===Connect===
With a global Connect App, that is hosted on the Master, you can share and gain information that are by interest for everyone.

*Give status updates
*Request supplies
*share important information around the whole area and to the HQ

you can also have another Connect App that is hosted on the App Platform of your iPP, this way only helpers/workers from your side will get all the information.

===Recordings===
When in a rush and you need supplies or help as soon as possible, you don't have time to wait for your colleague on the phone to write everything down.

The best option is to tell everything you need and hang up again, the colleague could listen to that recording and write down everything that was requested.

*Local Recordings
*Global Recordings
*See how the situation of the area is
*Relisten to recordings to keep track of the timeline of the crisis situation

===Conferencing===
Having Conferences with external and/or internals Members is also something you might want to have dedicated and also globally. Even if the connection to the Master is interrupted, you can still have internal conferences.

*Emergency Meetings
*Meetings on the condition of the area
*Meetings with family and friends for safety checks

=== Projects ===
When having an Emergency you might already have some kind of plan for tasks to do, this will help scheduling the situation and also handle the tasks easier.

For this the Projects App will be used, here you can:

* Organize Tasks in Teams or for individual person
* Already be prepared for an Emergency situation
* Having Tasks for local iPPs which may not be neccessary for all iPPs
* Distribute Tasks to operators

=== Contacts App (+ Contacts Manager) ===
With the Contacts App you can have a globally distributed Contact List for all Emergency helpers and Emergency services. This will allow to:

* Have a complete overview of the Persons/Workers which are available and there to be contacted
* No single contacting point = All emergency services have the possibilty to contact other service members
* Always have the contact list up-to-date with an always running Contacts Manager App (must be installed seperately on Windows) to provide the right numbers and also have a synchronisation of all contacts 
==Our scenario and configuration==
In this Section I will get more into detail of our scenario.

For our Scenario we have a Master/Slave scenario. The whole System looks like this:

[[File:ipp_concept.png|ipp_concept.png/]]

The Master is located in our Datacenter and each PBX in the Suitcases acts as Slave PBX.

[[File:ipp_suitcase_2.jpg|ipp_suitcase_2.jpg/]]

This Suitcase consists of:

*2 IP-DECT Antennas with 2 Handsets
*Charging station for the Handsets
*IP102
*IP112
*POE+ Switch
*Notebook

Drawers we use:

*19 Inch 2HU drawer for the Devices
*Inlay 2U drawer
*19 Inch 1HU drawer for the Notebook

Rack case:

*19 Inch 6HU Rack case

For Data connection we have used:

*Satellite Internet
*LTE/4G Router

In summary we have 4 different iPPs prepared with different Hardware and looks, each iPP has its own App Platform with dedicated and also global Apps.

[[File:ipp_ap_concept.png|ipp_ap_concept.png/]]

Each one of these iPPs has a different size and also different Hardware used. This will show you what is possible to have and how you could realize your concept:

[[File:ipp_suitcase_3.jpg|ipp_suitcase_3.jpg/]] [[File:ipp_suitcase_4.jpg|ipp_suitcase_4.jpg/]]

==Power consumption==
This is only an example of how much Power one Suitcase might consume. For the Data connection we are using a Starlink mobile Antenna. The Power consumption will drop/rise depending on your chosen Data Connection.

===iPP02===
Because this is our biggest suitcase, it's also (obviously) the one which consumes the most Power:

*120Wh

===iPP03===

*40Wh

===iPP04===

*70Wh

Reference14r2:Concept innovaphone Portable Platform

2026-04-17T12:16:59Z

Nwe: /* Our scenario and configuration */

===innovaphone Portable Platform===

==What is iPP? (innovaphone Portable Platform)==
The iPP is a concept to demonstrate how a mobile communication solution based on the innovaphone system may look like. It provides the initial starting point to develop own concepts, ideas, solutions or products.
The requirements & the area of use are influencing factors, there are many possibilities and options to have it as dynamically as possible!

[[File:ipp_suitcase_2.jpg]]

In this picture you can see what this solution '''could''' look like, let's be creative and say: why have a Suitcase for that? We can even have a mobile solution built into a Car! As you can see you can do a lot with this Solution, and it also depends on your scenario, needs, use case and so on.

There are some things you have to consider, but I will get back to that later on.

==Where can it be useful?==
You can use the iPP as you desire. Here are some examples to show you where you might need to have an iPP:

*Secure communication system in disaster areas (e.g. Ahr Valley, Germany)
*Secure communication system at big events
*Secure communication system to reliably secure supply
*Secure communication system for critical infrastructures
*Secure communication system on a (big) construction site
*Do you have any more ideas?

The big advantage of the iPP is: you can use it nearly anywhere and design it specially for your needs and what you consider as important!

==Configuration scenario==
Depending on your needs and your infrastructure, you can have two scenarios to set up the iPP.

===Stand-Alone iPP===
*Autonomous communication solution without Master / Slave set-up
*Via internet connection
**Own trunk line possible
**Audio conferences with up to 100 internal participants (depending on your used Hardware). You can also have external participants depending on your used Hardware/Software!

'''Regarding external access:''' As a Stand-Alone iPP it might not be possible to have a fixed external IP-Address, depending on what kind of Data connection you use and what your ISP can offer. Although it would be possible to have DynDNS, this also relies on what kind of Hardware/Software you have deployed. Thus, it is dependent on these factors if you can have external access too.

===Master / Slave iPP===
*Communication solution with Master / Slave set-up
*Can also be operated as standalone if there is no internet connection available, for example
*Via internet connection
**Connection to Master or additional Slave systems available (Let's say you have a huge area to cover, then you might need more than one System on the area)
**Own and / or central trunk line, possible (you can keep the trunk redundant with central and also dedicated trunks)
**Audio conferences with up to 100 internal and external participants

'''Regarding external access for the Slaves:''' For our Scenario we used a VPN Tunnel from the Slaves to the HQ (with a fixed IP-Address in our Data center), all DNS-Queries for the Slaves then points to the HQ where the Reverse-Proxy then routes the Traffic accordingly to the right System. '''BUT''' in this case you will need a HQ, keep in mind that external access will also be down if the HQ (or the connection to the HQ) should be down.

==Use case scenario==
To give you a better feeling on how to deploy the iPP I will give you a use case scenario:

*Imagine a flooding of a whole suburban city
*The flooding has nearly destroyed everything (Streets, Infrastructure, Cellular towers, Houses, Electricity...)
*With the iPP you have control over the whole area and cover the whole Crisis area

[[File:ipp_crisis_area.png]]

*Now Police, Ambulance, Firefighters, THW (Technisches Hilfswerk), volunteers and many others arrive at the spot
*With the iPP you can combine the communication between every Person there, this way everyone have the same source of information and is connected together
*You can have a connection to a Headquarter where you can ask for supplies or more help
*One site can request help from another site

[[File:ipp_connect.png]]

*You can additionally add Maps as an App so that Users can orientate if they're not familiar with the area

[[File:ipp_maps.png]]

*You can have emergency meetings

[[File:ipp_conferencing.png]]

*It's possible to give status updates to the Headquarter or to another location

And these are only a few options you have to secure a stable and secure communication solution in such situation.

==Things to have in consideration==
As I mentioned before, a lot of things depends on the situation you want to cover and possibilities you have. There are still some Key-things that you have to take in account when deploying an iPP.

The two main points which needs to be covered are:

===Power Supply===
Of course, you will need electricity to have this System working, Sometimes you have Low-voltage-network on spot but what if not? Let's take a look at the options!

*As already mentioned, depending on the situation and use case you can use Low-voltage-network to get electricity (If available!)
*Another option can be via battery network with following charging options:
**Solar
**Wind energy
**Generator (Diesel, gasoline, propane)

===Data connection===
Depending on if you want to have external connection too, you need a way for Data connection. In order for this too work, you have multiple options:

*Ethernet, if available you can use Ethernet on side but depending on the situation and availability it might not be available on side
*LTE/4G/5G Router
*Satellite Internet
*450MHz Network (Available in Germany from 2025, especially designed for critical Infrastructure)

==App Platform concept==
To provide Apps globally and also dedicated, you can have several App Platforms in your scenario.

For example, let's take a look at some Apps:

===Connect===
With a global Connect App, that is hosted on the Master, you can share and gain information that are by interest for everyone.

*Give status updates
*Request supplies
*share important information around the whole area and to the HQ

you can also have another Connect App that is hosted on the App Platform of your iPP, this way only helpers/workers from your side will get all the information.

===Recordings===
When in a rush and you need supplies or help as soon as possible, you don't have time to wait for your colleague on the phone to write everything down.

The best option is to tell everything you need and hang up again, the colleague could listen to that recording and write down everything that was requested.

*Local Recordings
*Global Recordings
*See how the situation of the area is
*Relisten to recordings to keep track of the timeline of the crisis situation

===Conferencing===
Having Conferences with external and/or internals Members is also something you might want to have dedicated and also globally. Even if the connection to the Master is interrupted, you can still have internal conferences.

*Emergency Meetings
*Meetings on the condition of the area
*Meetings with family and friends for safety checks

==Our scenario and configuration==
In this Section I will get more into detail of our scenario.

For our Scenario we have a Master/Slave scenario. The whole System looks like this:

[[File:ipp_concept.png]]

The Master is located in our Datacenter and each PBX in the Suitcases acts as Slave PBX.

[[File:ipp_suitcase_2.jpg]]

This Suitcase consists of:

*2 IP-DECT Antennas with 2 Handsets
*Charging station for the Handsets
*IP102
*IP112
*POE+ Switch
*Notebook

Drawers we use:

*19 Inch 2HU drawer for the Devices
*Inlay 2U drawer
*19 Inch 1HU drawer for the Notebook

Rack case:

*19 Inch 6HU Rack case

For Data connection we have used:

*Satellite Internet
*LTE/4G Router

In summary we have 4 different iPPs prepared with different Hardware and looks, each iPP has its own App Platform with dedicated and also global Apps.

[[File:ipp_ap_concept.png]]

Each one of these iPPs has a different size and also different Hardware used. This will show you what is possible to have and how you could realize your concept:

[[File:ipp_suitcase_3.jpg]] [[File:ipp_suitcase_4.jpg]]

==Power consumption==
This is only an example of how much Power one Suitcase might consume. For the Data connection we are using a Starlink mobile Antenna. The Power consumption will drop/rise depending on your chosen Data Connection.

===iPP02===
Because this is our biggest suitcase, it's also (obviously) the one which consumes the most Power:

*120Wh

===iPP03===

*40Wh

===iPP04===

*70Wh

Reference14r2:Concept innovaphone Portable Platform

2026-04-17T12:16:32Z

Nwe: /* Our scenario and configuration */

===innovaphone Portable Platform===

==What is iPP? (innovaphone Portable Platform)==
The iPP is a concept to demonstrate how a mobile communication solution based on the innovaphone system may look like. It provides the initial starting point to develop own concepts, ideas, solutions or products.
The requirements & the area of use are influencing factors, there are many possibilities and options to have it as dynamically as possible!

[[File:ipp_suitcase_2.jpg]]

In this picture you can see what this solution '''could''' look like, let's be creative and say: why have a Suitcase for that? We can even have a mobile solution built into a Car! As you can see you can do a lot with this Solution, and it also depends on your scenario, needs, use case and so on.

There are some things you have to consider, but I will get back to that later on.

==Where can it be useful?==
You can use the iPP as you desire. Here are some examples to show you where you might need to have an iPP:

*Secure communication system in disaster areas (e.g. Ahr Valley, Germany)
*Secure communication system at big events
*Secure communication system to reliably secure supply
*Secure communication system for critical infrastructures
*Secure communication system on a (big) construction site
*Do you have any more ideas?

The big advantage of the iPP is: you can use it nearly anywhere and design it specially for your needs and what you consider as important!

==Configuration scenario==
Depending on your needs and your infrastructure, you can have two scenarios to set up the iPP.

===Stand-Alone iPP===
*Autonomous communication solution without Master / Slave set-up
*Via internet connection
**Own trunk line possible
**Audio conferences with up to 100 internal participants (depending on your used Hardware). You can also have external participants depending on your used Hardware/Software!

'''Regarding external access:''' As a Stand-Alone iPP it might not be possible to have a fixed external IP-Address, depending on what kind of Data connection you use and what your ISP can offer. Although it would be possible to have DynDNS, this also relies on what kind of Hardware/Software you have deployed. Thus, it is dependent on these factors if you can have external access too.

===Master / Slave iPP===
*Communication solution with Master / Slave set-up
*Can also be operated as standalone if there is no internet connection available, for example
*Via internet connection
**Connection to Master or additional Slave systems available (Let's say you have a huge area to cover, then you might need more than one System on the area)
**Own and / or central trunk line, possible (you can keep the trunk redundant with central and also dedicated trunks)
**Audio conferences with up to 100 internal and external participants

'''Regarding external access for the Slaves:''' For our Scenario we used a VPN Tunnel from the Slaves to the HQ (with a fixed IP-Address in our Data center), all DNS-Queries for the Slaves then points to the HQ where the Reverse-Proxy then routes the Traffic accordingly to the right System. '''BUT''' in this case you will need a HQ, keep in mind that external access will also be down if the HQ (or the connection to the HQ) should be down.

==Use case scenario==
To give you a better feeling on how to deploy the iPP I will give you a use case scenario:

*Imagine a flooding of a whole suburban city
*The flooding has nearly destroyed everything (Streets, Infrastructure, Cellular towers, Houses, Electricity...)
*With the iPP you have control over the whole area and cover the whole Crisis area

[[File:ipp_crisis_area.png]]

*Now Police, Ambulance, Firefighters, THW (Technisches Hilfswerk), volunteers and many others arrive at the spot
*With the iPP you can combine the communication between every Person there, this way everyone have the same source of information and is connected together
*You can have a connection to a Headquarter where you can ask for supplies or more help
*One site can request help from another site

[[File:ipp_connect.png]]

*You can additionally add Maps as an App so that Users can orientate if they're not familiar with the area

[[File:ipp_maps.png]]

*You can have emergency meetings

[[File:ipp_conferencing.png]]

*It's possible to give status updates to the Headquarter or to another location

And these are only a few options you have to secure a stable and secure communication solution in such situation.

==Things to have in consideration==
As I mentioned before, a lot of things depends on the situation you want to cover and possibilities you have. There are still some Key-things that you have to take in account when deploying an iPP.

The two main points which needs to be covered are:

===Power Supply===
Of course, you will need electricity to have this System working, Sometimes you have Low-voltage-network on spot but what if not? Let's take a look at the options!

*As already mentioned, depending on the situation and use case you can use Low-voltage-network to get electricity (If available!)
*Another option can be via battery network with following charging options:
**Solar
**Wind energy
**Generator (Diesel, gasoline, propane)

===Data connection===
Depending on if you want to have external connection too, you need a way for Data connection. In order for this too work, you have multiple options:

*Ethernet, if available you can use Ethernet on side but depending on the situation and availability it might not be available on side
*LTE/4G/5G Router
*Satellite Internet
*450MHz Network (Available in Germany from 2025, especially designed for critical Infrastructure)

==App Platform concept==
To provide Apps globally and also dedicated, you can have several App Platforms in your scenario.

For example, let's take a look at some Apps:

===Connect===
With a global Connect App, that is hosted on the Master, you can share and gain information that are by interest for everyone.

*Give status updates
*Request supplies
*share important information around the whole area and to the HQ

you can also have another Connect App that is hosted on the App Platform of your iPP, this way only helpers/workers from your side will get all the information.

===Recordings===
When in a rush and you need supplies or help as soon as possible, you don't have time to wait for your colleague on the phone to write everything down.

The best option is to tell everything you need and hang up again, the colleague could listen to that recording and write down everything that was requested.

*Local Recordings
*Global Recordings
*See how the situation of the area is
*Relisten to recordings to keep track of the timeline of the crisis situation

===Conferencing===
Having Conferences with external and/or internals Members is also something you might want to have dedicated and also globally. Even if the connection to the Master is interrupted, you can still have internal conferences.

*Emergency Meetings
*Meetings on the condition of the area
*Meetings with family and friends for safety checks

==Our scenario and configuration==
In this Section I will get more into detail of our scenario.

For our Scenario we have a Master/Slave scenario. The whole System looks like this:

[[File:ipp_concept.png]]

The Master is located in our Datacenter and each PBX in the Suitcases acts as Slave PBX.

[[File:ipp_suitcase_2.jpg]]

[[File:ipp_suitcase_back.jpeg]]

[[File:ipp_suitcase2_back2.jpeg]]

This Suitcase consists of:

*2 IP-DECT Antennas with 2 Handsets
*Charging station for the Handsets
*IP102
*IP112
*POE+ Switch
*Notebook

Drawers we use:

*19 Inch 2HU drawer for the Devices
*Inlay 2U drawer
*19 Inch 1HU drawer for the Notebook

Rack case:

*19 Inch 6HU Rack case

For Data connection we have used:

*Satellite Internet
*LTE/4G Router

In summary we have 4 different iPPs prepared with different Hardware and looks, each iPP has its own App Platform with dedicated and also global Apps.

[[File:ipp_ap_concept.png]]

Each one of these iPPs has a different size and also different Hardware used. This will show you what is possible to have and how you could realize your concept:

[[File:ipp_suitcase_3.jpg]] [[File:ipp_suitcase_4.jpg]]

==Power consumption==
This is only an example of how much Power one Suitcase might consume. For the Data connection we are using a Starlink mobile Antenna. The Power consumption will drop/rise depending on your chosen Data Connection.

===iPP02===
Because this is our biggest suitcase, it's also (obviously) the one which consumes the most Power:

*120Wh

===iPP03===

*40Wh

===iPP04===

*70Wh

Reference13r2:Gateway/Interfaces/SIP

2026-04-13T10:48:45Z

Nwe:

== SIP Registration section ==
The entry fields for a '''SIP registration''' are:
{|
|valign=top nowrap=true|'''Name'''
|Descriptive name for this registration.
|-
|valign=top nowrap=true|'''Disable'''
|A switch to temporarily disable this interface without deleting the configuration.
|-
|valign=top nowrap=true|'''Type'''
|
*Provider: Creates a registration at a remote SIP server (of a SIP provider)
*Open Federation: SIP/TLS interface without registration used to send and receive calls to federation partners.
*Closed Federation: Like 'Open Federation' but even more restricted to pre-configured set of federation partners. In the 'Closed Federation' mode, the DNS queries for resolving the domain name of the federation partner are made without the '''Recursion Desired''' (RD) Flag. This means only DNS entries configured on the local DNS server are resolved, so the list of the partners for 'Closed Federation' must be maintained on the local DNS server.
|-
|valign=top nowrap=true|'''Transport'''
|
*UDP: SIP signaling using UDP as transport protocol (RFC 3261).
*TCP: SIP signaling using TCP as transport protocol (RFC 3261).
*TLS: SIP signaling using TLS as transport protocol (RFC 3261). [[Howto:Security works with innovaphone]]
*Without registration: By default sip trunks are using with registration. Can be disabled here.
|-
|valign=top nowrap=true|'''AOR'''
|Address of Record: SIP-URI used to register. Enter the registration ID followed by the SIP provider domain name (for example 8111111e0@sipgate.de or 8111111e0@x.x.x.x:5080 if you need to use the IP-address and a different Port number).
|-
|valign=top nowrap=true|'''Local Hostname'''
|The Local Domain for SIP Federation enables to select the TLS Certificate according to the Domain Name. On the incoming SIP calls the host part of the URI is removed if equals with the Local Domain configured here, and the user part is used as Name (H323-ID) or Number (E164).
|-
|valign=top nowrap=true|'''Local Port'''
|The local source port for SIP signalling can be configured here. If its empty a random port will be used. ''Dont use the same static port on multiple SIP Accounts!''
|-
|valign=top nowrap=true|'''Proxy'''
|DNS name or IP address of the SIP proxy where SIP messages (REGISTER,INVITE,etc) are to be sent to. Proxy can be omitted if domain part of AOR can be used as remote signaling destination. (append ":<port>" if you need a different destination Port)
|-
|valign=top nowrap=true|'''STUN Server'''
|The STUN servers to use. See [[{{NAMESPACE}}:IP4/General/STUN | STUN]] for details regarding the format.
|}

== Authorization ==
Username and password for authorization. Username can be omitted if equal to userpart of AOR.

== Media Properties ==

The configuration of the media properties is evaluated for calls from/to this interface to/from a physical (ISDN, analog, TEST, ...) only. If media relay is active for a call using this interface an 'exclusive' coder config is used to prohibit the use of any other coder. This 'exclusive code media-relay' config can be used to solve interop problems with other equipment which does not support media renegotiation, because with this config no media renegotiation will be performed.

For more information see [[{{NAMESPACE}}:Gateway/Interfaces/Media_Properties | Media Properties]] and [[Howto:Security works with innovaphone]]

== SIP Interop Tweaks ==

{|
|valign=top nowrap=true|'''Proposed Registration Interval'''
|Set in seconds, default is 120 seconds
|-
|valign=top nowrap=true|'''Accept INVITE's from Anywhere'''
|If disabled, registered interfaces will reject INVITE's not coming from the SIP server with "305 Use Proxy".
|-
|valign=top nowrap=true|'''Enforce Sending Complete'''
|Affects handling of "484 Address Incomplete" responses. If enabled and "484 Address Incomplete" is received, the call is cleared. If not enabled and "484 Address Incomplete" is received, the call is retained and re-initiated in case of new dialing digits.
|-
|valign=top nowrap=true|'''No Video'''
|Removes Video Capabilities from outgoing media offer.
|-

|valign=top nowrap=true|'''To Header when Sending INVITE'''
|Affects only outgoing diverted calls .
*'''Called Party''': If set we write CDPN into To header of outgoing INVITE (and DGPN into History-Info header).
*'''Original Called Party''': If set we write the DGPN into To header of outgoing INVITE (and CDPN into Request-URI).
|-
|valign=top nowrap=true|'''From Header when Sending INVITE'''
|Controls the local URI (From header) of outgoing calls. Applys to registered interfaces only.
*'''Fixed AOR''': Fixed AOR is used as From-URI regardless of the actual calling party number.
*'''AOR with CGPN as Display''': Fixed AOR is used as From-URI and calling party number is added as display string.
*'''CGPN in user part of URI''': Variable From-URI with actual calling party number.
|-
|valign=top nowrap=true|'''Identity Header when Sending INVITE'''
|Controls the identity header (P-Preferred-Identity, P-Asserted-Identity and Remote-Party-Id) sent on outgoing calls
*'''CGPN in user part of URI''': Variable Identity-URI with actual calling party number
*'''Fixed AOR''': Fixed AOR is used as Identity-URI regardless of the actual calling party number
*'''UUI''': the relayed call must include ''user-to-user-info'' (UUI). This UUI is used as Identity-URI. [[Howto:How_to_customize_the_From/Identity_header_value_at_SIP_interfaces|In the UUI]], the string <code>{initiator}</code> is replaced by the the call's ''diversion-info'' (a.k.a. ''leg-2-info'') if available or the calling party number otherwise
|-
|valign=top nowrap=true|'''Reliability of Provisional Responses'''
|Controls the support of PRACK (RFC-3262).
*'''Supported''': Supported as optional extension.
*'''Required''': Required as mandatory extension.
*'''Disabled''': Hide support for PRACK extension.
|-
|valign=top nowrap=true|'''Advanced'''
|Allows the configuration of additional, not further documented, interop tweaks(e.g. /pai on). The same tweaks can be configured also globally(i.e. not for this SIP-Interface) at the SIP(or TSIP/SIPS)-module. Any tweaks configured at the SIP-Interface will overwrite globally configured tweaks.
|}

=== More frequently used ''Advanced'' Parameters ===
There are some options which influence the stack behaviour to handle ambiguities in the SIP standard:

; /pai on : send identity URI in <code>P-Asserted-Identity</code> header. By default, it is sent in the <code>P-Asserted-Identity</code> header for calls from the PBX to the endpoint, in the <code>P-Preferred-Identity</code> header otherwise
; /ppi on : send identity URI in <code>P-Preferred-Identity</code> header

Other options are available which instruct the stack to use non-standard or deprecated behaviour. Note that this should only be used in rare cases. Better have the vendor of your 3rd party SIP equipment fix its stack implementation:

; /send-deprecated-diversion-header on : send call history information in the deprecated <code>Diversion</code> header in addition to the <code>History-Info</code> header
; /single-audio-description : don't send SAVP+AVP, but SAVP or AVP media description in SDP
; /no-authentication-info : don't send <code>Authentication-Info</code> header in REGISTER response
; /send-no-historyinfo : don't send call history information (i.e. call forward) in outgoing Invite
; /contact-addr [option] : adjust the Contact header to a fixed value necessary in scenarios of SIP Provider trunk without registration using TCP/TLS and uses Contact Header for Authentication. Example /contact-addr x.x.x.x:5061;user=phone;transport=TLS
; /options-interval n : Send SIP options every "n" seconds to the configured remote SIP Address configured on the Interface.
; /get-cdpn-from-to-uri : get CDPN of incoming provider-calls from To-URI instead of Request-URI
; /prefer-pai : get CGPN from PAI/PPI/RPID if present
; /take-sendonly-as-inactive on : some endpoints use sendonly instead of inactive (Useful if no MOH is played when hold is used from a SIP-Endpoint)
; /no-diverting-name: don't add display name to diverting party URI
; /installed-certificate: for SIP/TLS trunks where the installed certificate should be used instead of the build-in device certificate

=== Extra Options ===
{{3rd_Party_Input}}

==== Disable Interworking of Hold Notifications to SIP Provider ====

During the tests we concluded that when interworking the hold-notify message to SIP and sending to the SIP Provider two consecutive Re-Invites with "send-only" attributes, the IMS platform replies to the second re-invite with "inactive". By doing so this call is put on hold without any Music on Hold - just silence.
To avoid this behaviour we need to disable the interworking of the hold-notify message by this setting:

!config add SIP /no-hr-notify (Alternative: TSIP / SIPS)
!config write
!config activate

==== SIP Options Interval (Optional) ====

Some Provider uses SIP Options to monitor the SIP Trunks, so it's mandatory that Innovaphone replies to incoming SIP Options received. This is done by default. Additionally we can also send SIP Options to the SIP Proxy and have similar mechanism for redundancy. If the remote Proxy doesn't reply to outgoing SIP Options, the Innovaphone Gateway will send the call to the next interface. To enable sending of Options - messages, the following setting must be done:

!config add SIP /options-interval 30 (Alternative: TSIP / SIPS)
!config write
!config activate

This option will only take effect on connections "without registration". In connections with a registration there already exist a keep alive.

==== Remove Comfort Noise (CN) Capability from SDP ====

During the tests we found out that some specific 3rd party devices connected to the IMS network support only a single coder/payload in the offer. When doing the coder negotiation, this devices repeat the coder negotiation until they have only 1 coder in the offer or until they reach a specific number of retries. Since Innovaphone by default always include the payload 13 (Comfort Noise) in addition to the used voice coder/payload, this would make the remote device to do multiple re-invites to try to reach the single coder/payload in the offer.
To avoid unnecessary signalling, we should disable the sending of Comfort Noise capability.

!config add SIP /rem-cn-capability (Alternative: TSIP / SIPS)
!config write
!config activate

==== Setting of P-Asserted ID instead of P-Preferred ID ====

When using the feature of ReRouting the call (SIP 302 Move Temporary) to the SIP Trunk, the IMS platform checks the P-Asserted ID setting. By default we send as P-Preferred ID instead, so that will not work. As a result, we need to configure the following setting:

!config add SIP /pai (Alternative: TSIP / SIPS)
!config write
!config activate

Reference16r1:Concept App Service Conference Scaler

2026-04-02T05:13:42Z

Nwe: /* Troubleshooting */

{{FIXME|reason=This product is in the beta phase and is not yet finished}}

[[Category:Concept|Apps]]

The Conference Scaler app is required to use multiple conference interfaces within one conference.

== Applies to ==
* Conference Scaler app

== Requirements ==
* innovaphone PBX
* innovaphone Application Platform
* V16r1
* At least one innovaphone Gateway with SCNF Interfaces
* The Devices API must be available for the settings plugin therefore the gateways have to be connected to devices

== Setting Plugins ==
=== Conference Scaler ===
With the Conference Scaler settings plugin the Conference Scaler app object can be created, edited and deleted on the PBX.
Furthermore, the conference interface to be used can be configured if the Devices API is available.
=== Conference ===
In the conference plugin, the menu item "Additional Features" appears for conferences. With this setting you can enable the use of the Conference Scaler within the conference object. Additionally, you can activate the use of PBX channels licenses.

== More information ==
=== Conference Scaler App (innovaphone-conferencescaler) ===
[[File:Conferencescaler.png|110px|conferencescaler.png/|/Conferencescaler.png]]

The hidden app offers an interface via the Service API (com.innovaphone.conferencescaler), that is used by the conference object.

Parameters:
;URL: <pre>http://<ap.domain.tld>/<domain.tld>/<instance-name>/innovaphone-conferencescaler</pre>

== Features ==
* Allows the use of multiple conference interfaces within multiple conference
** This removes previous participant limits
** Adds flexibility to existing infrastructure
** The Conference Scaler app nominates one conference interface which will manage the control for voice streams. This conference interface will then receive and distribute the audio streams to all interfaces which are connected to the Conference Scaler app
** The Conference Scaler app will split the participants in a Round Robin maner, therefore the load will be evenly distributed to the interfaces
** If an interface has reached its limit for participants, the Conference Scaler app will then select a different and free Interface for a new incoming call

== Configuration ==
# Download the Conference Scaler app via App Store. (or use the Settings Plugin "app installer" to install the App + Instance automatically and skip to step 5)
# Install the Conference Scaler app on the App Platform.
# Create an instance in the Conference Scaler app on the App Platform.
# Make sure the Instance is running.
# Create via the settings plugin a new Conference Scaler app object.
# Select the Conference Interfaces to be used by the Conference Scaler app inside the Setting plugin.
# Allow the use of the Conference Scaler app on the Conference settings plugin.

== Restrictions / Known issues ==
* Using the Conference Scaler app generates an overhead depending on usage patterns, so the increase in the number of participants does not correspond to a multiple of x interfaces. (One participant will take up an estimation of 1,5x CPU-Load as the usual conference flow (because data is additionally send to a different interface). So for an interface with max. 100 Video participants this amount will decrease to a maximum participant limit of 66 (as 100/1,5=66). Therefore if you want to achieve a maximum of X video participants you need to take that information into account for calculating the needed interfaces)
* In case of failure of the App Platform the Conference Scaler app will not be available as well. Therefore it will not be possible to join or continue an ongoing conference.
* Room reservations are not needed anymore
* Video sharing between the conference interfaces is handled via multicast and this requires that the interfaces are on the same network (Multicast addresses differs between 239.0.0.0 - 239.255.255.255)
* Currently, some videos may not display if they are routed through different gateways (this issue is known and will be resolved before release)
* Only SCNF interfaces are currently supported

== Troubleshooting ==
If you can reproduce the problem, clear the contents of the current app log, reproduce the issue, take a screenshot (depending on the issue), save the browser logs and the app logs, and send them as an attachment to your support ticket.
For the app, please make sure that following trace flags are enabled:
* App
* App Websocket

=== No Channel free ===
If you get a „No Channel free“ Error when trying to join a Conference Room after setting up the Conference, check if the desired room has any reserved channel configured, if so then remove the reservation of channels. The Conference Scaler does not consider channel reservations and therefore rejects the caller.

== Related Articles ==
* [[Reference15r1:Concept_Conference_Web_Access]]
* [[Reference15r1:Concept_Conference]]
* [[Courseware:IT_Connect_-_08.1_Conferencing]]
* [[Courseware:IT_Plus_-_Public_Access_to_Conference_Resources#Bandwidth_and_CPU_consideration]]

Reference16r1:Concept App Service Conference Scaler

2026-03-13T14:04:40Z

Nwe: /* Configuration */

{{FIXME|reason=This product is in the beta phase and is not yet finished}}

[[Category:Concept|Apps]]

The Conference Scaler app is required to use multiple conference interfaces within one conference.

== Applies to ==
* Conference Scaler app

== Requirements ==
* innovaphone PBX
* innovaphone Application Platform
* V16r1
* At least one innovaphone Gateway with CONF/SCNF Interfaces
* The Devices API must be available for the settings plugin therefore the gateways have to be connected to devices

== Setting Plugins ==
=== Conference Scaler ===
With the Conference Scaler settings plugin the Conference Scaler app object can be created, edited and deleted on the PBX.
Furthermore, the conference interface to be used can be configured if the Devices API is available.
=== Conference ===
In the conference plugin, the menu item "Additional Features" appears for conferences. With this setting you can enable the use of the Conference Scaler within the conference object. Additionally, you can activate the use of PBX channels licenses.

== More information ==
=== Conference Scaler App (innovaphone-conferencescaler) ===
[[File:Conferencescaler.png|110px|conferencescaler.png/|conferencescaler.png/]]

The hidden app offers an interface via the Service API (com.innovaphone.conferencescaler), that is used by the conference object.

Parameters:
;URL: <pre>http://<ap.domain.tld>/<domain.tld>/<instance-name>/innovaphone-conferencescaler</pre>

== Features ==
* Allows the use of multiple conference interfaces within multiple conference
** This removes previous participant limits
** Adds flexibility to existing infrastructure
** The Conference Scaler app nominates one conference interface which will manage the control for voice streams. This conference interface will then receive and distribute the audio streams to all interfaces which are connected to the Conference Scaler app
** The Conference Scaler app will split the participants in a Round Robin maner, therefore the load will be evenly distributed to the interfaces
** If an interface has reached its limit for participants, the Conference Scaler app will then select a different and free Interface for a new incoming call

== Configuration ==
# Download the Conference Scaler app via App Store. (or use the Settings Plugin "app installer" to install the App + Instance automatically and skip to step 5)
# Install the Conference Scaler app on the App Platform.
# Create an instance in the Conference Scaler app on the App Platform.
# Make sure the Instance is running.
# Create via the settings plugin a new Conference Scaler app object.
# Select the Conference Interfaces to be used by the Conference Scaler app inside the Setting plugin.
# Allow the use of the Conference Scaler app on the Conference settings plugin.
# Currently, for technical reasons, the Conference Scaler app object requires video and app sharing licenses or a UC license, which should be configured in the PBX Advanced UI

== Restrictions / Known issues ==
* Currently, for technical reasons, the Conference Scaler app object requires video and app sharing licenses or a UC license, which should be configured in the PBX Advanced UI
* Using the Conference Scaler app generates an overhead depending on usage patterns, so the increase in the number of participants does not correspond to a multiple of x interfaces. (One participant will take up an estimation of 1,5x CPU-Load as the usual conference flow (because data is additionally send to a different interface). So for an interface with max. 100 Video participants this amount will decrease to a maximum participant limit of 66 (as 100/1,5=66). Therefore if you want to achieve a maximum of X video participants you need to take that information into account for calculating the needed interfaces)
* In case of failure of the App Platform the Conference Scaler app will not be available as well. Therefore it will not be possible to join or continue an ongoing conference.
* Room reservations are not needed anymore
* Video sharing between the conference interfaces is handled via multicast and this requires that the interfaces are on the same network (Multicast addresses differs between 239.0.0.0 - 239.255.255.255)
* Only SCNF interfaces are currently supported
* For now only Hardware Gateways are supported, no IPVAs are able to be used currently

== Troubleshooting ==
If you can reproduce the problem, take a screenshot (depending on the issue), save the browser logs and the app logs, and send them as an attachment to your support ticket.

== Related Articles ==
* [[Reference15r1:Concept_Conference_Web_Access]]
* [[Reference15r1:Concept_Conference]]
* [[Courseware:IT_Connect_-_08.1_Conferencing]]
* [[Courseware:IT_Plus_-_Public_Access_to_Conference_Resources#Bandwidth_and_CPU_consideration]]

Reference16r1:Concept App Service Conference Scaler

2026-03-13T13:18:53Z

Nwe: /* Restrictions / Known issues */

{{FIXME|reason=This product is in the beta phase and is not yet finished}}

[[Category:Concept|Apps]]

The Conference Scaler app is required to use multiple conference interfaces within one conference.

== Applies to ==
* Conference Scaler app

== Requirements ==
* innovaphone PBX
* innovaphone Application Platform
* V16r1
* At least one innovaphone Gateway with CONF/SCNF Interfaces
* The Devices API must be available for the settings plugin therefore the gateways have to be connected to devices

== Setting Plugins ==
=== Conference Scaler ===
With the Conference Scaler settings plugin the Conference Scaler app object can be created, edited and deleted on the PBX.
Furthermore, the conference interface to be used can be configured if the Devices API is available.
=== Conference ===
In the conference plugin, the menu item "Additional Features" appears for conferences. With this setting you can enable the use of the Conference Scaler within the conference object. Additionally, you can activate the use of PBX channels licenses.

== More information ==
=== Conference Scaler App (innovaphone-conferencescaler) ===
[[File:Conferencescaler.png|110px|conferencescaler.png/|conferencescaler.png/]]

The hidden app offers an interface via the Service API (com.innovaphone.conferencescaler), that is used by the conference object.

Parameters:
;URL: <pre>http://<ap.domain.tld>/<domain.tld>/<instance-name>/innovaphone-conferencescaler</pre>

== Features ==
* Allows the use of multiple conference interfaces within multiple conference
** This removes previous participant limits
** Adds flexibility to existing infrastructure
** The Conference Scaler app nominates one conference interface which will manage the control for voice streams. This conference interface will then receive and distribute the audio streams to all interfaces which are connected to the Conference Scaler app
** The Conference Scaler app will split the participants in a Round Robin maner, therefore the load will be evenly distributed to the interfaces
** If an interface has reached its limit for participants, the Conference Scaler app will then select a different and free Interface for a new incoming call

== Configuration ==
# Download the Conference Scaler app via App Store. (or use the Settings Plugin "app installer" to install the App + Instance automatically and skip to step 5)
# Install the Conference Scaler app on the App Platform.
# Create an instance in the Conference Scaler app on the App Platform.
# Make sure the Instance is running.
# Create via the settings plugin a new Conference Scaler app object.
# Select the Conference Interfaces to be used by the Conference Scaler app.
# Allow the use of the Conference Scaler app on the Conference Object settings.

== Restrictions / Known issues ==
* Currently, for technical reasons, the Conference Scaler app object requires video and app sharing licenses or a UC license, which should be configured in the PBX Advanced UI
* Using the Conference Scaler app generates an overhead depending on usage patterns, so the increase in the number of participants does not correspond to a multiple of x interfaces. (One participant will take up an estimation of 1,5x CPU-Load as the usual conference flow (because data is additionally send to a different interface). So for an interface with max. 100 Video participants this amount will decrease to a maximum participant limit of 66 (as 100/1,5=66). Therefore if you want to achieve a maximum of X video participants you need to take that information into account for calculating the needed interfaces)
* In case of failure of the App Platform the Conference Scaler app will not be available as well. Therefore it will not be possible to join or continue an ongoing conference.
* Room reservations are not needed anymore
* Video sharing between the conference interfaces is handled via multicast and this requires that the interfaces are on the same network (Multicast addresses differs between 239.0.0.0 - 239.255.255.255)
* Only SCNF interfaces are currently supported
* For now only Hardware Gateways are supported, no IPVAs are able to be used currently

== Troubleshooting ==
If you can reproduce the problem, take a screenshot (depending on the issue), save the browser logs and the app logs, and send them as an attachment to your support ticket.

== Related Articles ==
* [[Reference15r1:Concept_Conference_Web_Access]]
* [[Reference15r1:Concept_Conference]]
* [[Courseware:IT_Connect_-_08.1_Conferencing]]
* [[Courseware:IT_Plus_-_Public_Access_to_Conference_Resources#Bandwidth_and_CPU_consideration]]

Howto16r1:Firmware Upgrade V15r1 V16r1

2026-03-13T08:06:36Z

Nwe: /* New Apps */

== Applies To ==
This information applies to:

* All 16r1 capable innovaphone devices
: For a general overview of the upgrade process and a list of supported devices with 16r1, see [[Howto:Firmware Upgrade]]
== Licenses ==
In case of cloud or rental model, don't worry about licenses.

If the system is licensed on premise, you'll need to regenerate the license file for v16 in https://portal.innovaphone.com/ and load into the system before upgrade (The system needs to have the SSC up to date).

== Migration Policy ==
Before you begin, be sure that your whole installation is running the latest 15r1 service release. 

=== TechAssist Upgrade Helper ===
* Before you start, make sure that all TechAssist tests (you will receive the required tests in the last update in the previous major version) labelled <code>Pre Upgrade: xy</code> are positive, if available
* When you are finished, make sure that all TechAssist tests (you will receive new tests with the upgrade) are positive

=== New TLS Profile ===
Please note that we have changed our TLS profiles ([[Reference16r1:IP4/General/TLS]]). The new ''Normal'' setting, which is the default value, now only allows TLS 1.3 and TLS 1.2.

== Changes visible to the end customers ==
Listed here are changes that should be communicated by resellers to end users prior to a upgrade, as the change will be visible/audible in the behaviour of the application/device.

* ''Nothing''

== Manual steps needed after upgrade ==
If the installer is not used for a new installation, some new default settings are not set. Please evaluate per app whether you want to configure the new default settings manually.

=== Connector for Microsoft 365 ===
If you plan to use the new '''Contact Search''' feature of the Connector for Microsoft 365, you need to perform two manual Steps:
# Create the '''microsoft365-api''' app object by using the Settings template
# Assign the '''microsoft365-api''' app object to every user who should be able to use the new Contact Search feature. (Of cause, you can use a template for that)
 
For a more detailed guide, please refer to the how-to article: [[Howto16r1:Configure Contact Search by Connector for Microsoft365#Creating the PBX app object using the PBX Manager Plugin]]

=== Remote Control ===
In order to use the Admin Configuration Panel of the Settings App – AP Remote Control, it is necessary to grant access to the '''admin''' API, available in the App tab of the Remote Control App object.

=== Working ===
For badge counts to work, the Working Manager app object must have '''Websocket''' and '''PbxSignal''' enabled.
For the Connect integration of the Working app, '''Websocket''', '''Services''' and '''Connect'''(in the Apps tab) must be enabled in the Working User app.

== New Apps ==
New Apps will not be installed automatically by the upgrade. The installation description of new apps is usually in the concept article. Please rate per app whether you want to install/use the new app and configure it manually.

* App Polls: [[Reference16r1:Concept App Polls]]
* App Service Conference Transcriptions [[Reference16r1:Concept_App_Service_myApps_Assistant]]
* App Service Connector for Whatsapp: [[Reference16r1:Concept_App_Service_Connector_for_Whatsapp]]
* App Conference Scaler: [[Reference16r1:Concept_Conference_Scaler_App]]

== Removed ==
The following software is no longer included.

* IP110A (can still be used with 15r1 firmware on current PBX versions)
* IP240A (can still be used with 15r1 firmware on current PBX versions)
* CA on CF card feature

== Deprecated ==
The following software is based on legacy technology, with no further development and limited maintenance and support.

* ''Nothing''

== Previously deprecated and now no longer supported ==
The following software is based on legacy technology, with no further development and no more maintenance and support.

* ''Nothing''

==Known Problems==
===Long Update-duration===
When you update, it can be up to 10 minutes before you have access to your app platform again.

== Related Articles ==
*[[Howto:Firmware_Upgrade]]
* [[Howto15r1:Firmware_Upgrade_V14r2_V15r1]]
[[Category:Howto|{{PAGENAME}}]]

Reference16r1:Gateway/Interfaces

2026-03-13T07:04:33Z

Nwe:

{{Special:Prefixindex/Reference16r1:Gateway/Interfaces}}

The display of the gateway’s configurable '''interfaces''' is organized in columns:

* '''Interface:''' The name of the interface. Clicking this name opens a popup page, on which all settings can be made. The settings are described in more detail in the following chapter "[[#CGPN.2FCDPN_Mappings|Administration/Gateway/Interfaces/Interface (ISDN & virtual interfaces)]]".
* '''CGPN In, CDPN In, CGPN Out, CDPN Out:''' Precise details on CGPN In, CDPN In, CGPN Out and CDPN Out mappings are contained in the chapter entitled "[[#CGPN.2FCDPN_Mappings|Administration/Gateway/Interfaces/CGPN-CDPN Mappings]]" further down in the text.
* '''State:''' The current state of the interface at physical and at protocol level. Possible states are: ''Up'', ''Down''.
* '''Registration:''' If a terminal has successfully registered with an ISDN, SIP or virtual interface, then this is indicated in this column through specification of the IP address <Name of the VoIP interface:Call number:IP address>.

== Interface (ISDN, SIP & virtual interfaces) ==

Clicking the name of an interface in the Interface column opens a popup page, on which the interfaces can be individually configured. Like the PBX objects, this popup page also contains standard entry fields that occur, more or less, in all interfaces. These standard fields are:

* '''Name:''' The descriptive name of the interface.
* '''Disable:''' A checked check box disables the relevant interface.
* '''Tones:''' The standard calling tone for the relevant interface is set with the Tones list box.
* '''Interface Maps:''' The interface can be configured as a point-to-point connection (Point-to-Point), as a point-to-multipoint connection (Point-to-Multipoint) or manually (Manual) using CGPN/CDPN maps.
See description further down in the text.
* '''Registration:''' With the Registration list box, an H.323 registration or a SIP registration can be initiated for ISDN interfaces. The routes to be handled as incoming and outgoing calls on the relevant interface are automatically created here (see "[[Reference:Administration/Gateway/Routes|Administration/Gateway/Routes]]").

=== ISDN interfaces (PPP, TEL1-4, BRI1-4, PRI1-4) ===

After selection of an '''interface map''', the relevant section is displayed. If Point-to-Point is selected, the Interface Maps Point-to-Point section is displayed:

* '''Area Code:''' The number of the local area (for example, 7031). http://en.wikipedia.org/wiki/List_of_dialling_codes_in_Germany
* '''Subscriber Number:''' The local network number (for example, 73009).
* '''National Prefix:''' The national prefix (for example, 0).
* '''International Prefix:''' The international prefix (for example, 00).

If Trunk Point-to-Multipoint is selected, the Interface Maps Point-to-Multipoint section is displayed:
* '''MSN1-3 / Ext.:''' For every ISDN basic access, several call numbers can be configured. The innovaphone gateways support up to three multiple subscriber numbers (MSN1-3), followed by the extension (Ext.), which represents the extension to which the MSN is to be mapped.
* '''National Prefix:''' The national prefix (for example, 0).
* '''International Prefix:''' The international prefix (for example, 00).

==== Coder Preferences section: ====

After selection of a registration method, the '''Coder Preferences''' section is displayed together with the relevant Registration section.

The standard entry fields in the Coder Preferences section are:

* '''Model:''' The Model list box allows you to select the coder to be used. The coders available for selection are: ''G711A, G711u, G723-53, G729A, G726-32 and XPARENT''. If the remote VoIP device does not support the set coder, a commonly supported coder is used, unless the Exclusive check box was enabled.

Note: The codec ''Clearmode'' is chosen when the bearer capability is not Audio or Speech, that means if the B-channel has 'Unrestricted Digital Information’
means data will be sent via the B-channel - then the codec ''Clearmode'' will be in use (NB: ''Clearmode'' used to be called ''XPARENT'' in previous versions).

* '''Frame:''' Determines the packet size used in transmitting voice data (in ms). Larger packets cause a greater delay in voice data transmission, but cause less load on the network, since the overhead involved in transporting the packets in the network is lower. The higher the packet size used, the lower the bandwidth effectively used.

'''Encoding method | Packet size | Bandwidth'''
---------------------------------------------
G.711 | 30ms | 77kb
G.711 | 90ms | 68kb
---------------------------------------------
G.729 | 30ms | 21kb
G.729 | 90ms | 12kb

* '''Exclusive:''' A checked check box enforces the set encoding (Model), regardless of whether it is supported by the remote VoIP device.
* '''Silence Compression:''' A checked check box enables SC (Silence Compression). With SC, no data is transmitted during pauses in the conversation. This also allows bandwidth to be saved without quality loss.
* '''Enable T.38:''' A checked check box enables the T.38 Fax-over-IP protocol. If a fax machine was connected to the relevant interface, then this check box must be enabled; otherwise, fax transmissions are not handled.
* '''Enable PCM:''' A checked check box enables the PCM switch (Pulse Code Manipulation). Calls from one interface to another interface are then handled directly over the ISDN PCM bus, which in turn saves DSP channels. This entry field is optional and is displayed only in particular devices.
* '''No ICE''' disables ''Interactive Connection Establishment'' (see [[{{NAMESPACE}}:Concept ICE]])

'''Registration section:'''

All non-virtual interfaces additionally have the '''Registration''' section after selection of the registration method.

==== H.323 Registration section ====

The entry fields for an '''H.323 registration''' are:
* '''Gatekeeper Address (primary):''' The primary gatekeeper IP address at which the interface is to register. If the primary gatekeeper is located on the same device, the local IP address 127.0.0.1 can also be entered here.
* '''Gatekeeper Address (secondary):''' The secondary gatekeeper IP address at which the interface is to register, if registration with the primary gatekeeper fails. If the secondary gatekeeper is located on the same device, the local IP address 127.0.0.1 can likewise be entered here.
* '''Gatekeeper ID:''' It is also sufficient to specify only the Gatekeeper ID (see also the chapter entitled "[[Reference:Administration/Gateway/General|Administration/Gateway/General]]").
* '''Name:''' The unique, descriptive H.323 name of the interface or registration.
* '''Number:''' The unique E.164 call number of the interface or registration.
* '''Password / Retype:''' The security of the registration can be raised by specifying a password (Password). The password must be confirmed (Retype).
* '''Supplementary Services (with Feature Codes):''' A checked check box enables the use of additional features (Feature Codes). See description in the chapter entitled "[[Reference:Administration/Gateway/General|Administration/Gateway/General]]".
* '''Dynamic Group:''' A dynamic group can be added to the H.323 registration. Groups can be configured as static, dynamic-in or dynamic-out. For members of static groups, calls are always signaled. It works differently for members of dynamic groups, which register with or unregister from a group dynamically using a function key (Join Group). The difference between dynamic-in and dynamic-out lies in whether the object is to be contained in the relevant group as standard (in) or not (out).
See also description in the chapter entitled "[[Reference:Administration/PBX/Objects|Administration/PBX/Objects]]".
* '''Direct Dial:''' Using Direct Dial, a call setup to the specified call number is initiated as soon as the handset is picked up. A conceivable scenario would be a lift emergency telephone that is connected with the security control room, for example.
* '''Locked White List:''' Here, you can specify a comma-separated list of call numbers that may also be dialed in the case of a locked telephone (for example, emergency services numbers, like 110, 911).

==== SIP Registration section ====

The entry fields for a '''SIP registration''' are:
* '''Server Address (primary):''' The optional IP address of the SIP provider to where the SIP messages (REGISTER,INVITE,etc.) are to be sent. Only necessary if either the IP address cannot be obtained from the SIP URI's domain or a proxy server is to be used.
* '''Server Address (secondary):''' Backup IP address used if the SIP server on the primary IP address does not answer anymore.
* '''ID:''' Here you enter the registration ID followed by the SIP provider domain name (for example 8111111e0@sipgate.de).
* '''STUN Server:''' Only necessary if the SIP server is outside the private network.
* '''Username:''' Username for authorization (only if different from the registration ID).
* '''Password / Retype:''' The password for authorization must be specified here (Password) and confirmed (Retype).

=== FXO interfaces ===

Basically the same configuration is used as for ISDN interfaces.

To receive calling line identification a block-dial route must be used for incoming calls to delay the forwarding of the received call until after the calling line id was received. Please note that you must not use an '!' (exclamation mark) as part of the ''Number Out'' field of the route (see [[{{NAMESPACE}}:Gateway/Routes/Map|Number In/Out]]), as this disables the ''en-bloc'' feature.

=== SIP interfaces (SIP1-4) ===

In addition to the ISDN interfaces (PPP, TEL1-4, BRI1-4, PRI1-4) and virtual interfaces (TEST, TONE, HTTP), there are also four SIP interfaces (SIP1-4), which can be used to obtain a trunk line from a SIP provider, for example. For a description of the entry fields, please refer to the description of the SIP registration above. There are, however, three further entry fields:

* '''Name:''' A descriptive name for the interface.
* '''Disable:''' A switch to temporarily disable this interface without deleting the configuration.
* '''From Header:''' Interoperability option for outgoing calls. Controls the way the CGPN is transmitted to the SIP provider.
** '''AOR:''' The From header contains the fixed registration URI (AOR). The actual calling party number and name will be transmitted inside the ''P-Preferred-Identity'' header (RFC 3325).
** '''AOR with CGPN as display:''' The From header contains the fixed registration URI (AOR) with the calling party number as display string in front of the AOR.
** '''CGPN in user part of URI:''' The From header contains an URI with the calling party number as user part (left from @).
* '''Registration:''' Corresponds to the Registration entry field of the ISDN interfaces. After selection of H.323, the Registration for H.323 section is displayed, enabling registration of this SIP trunk interface with a local innovaphone PBX. After selection of SIP, the Registration for SIP section is displayed, enabling in turn registration with a local non-innovaphone SIP PBX.

To obtain a trunk line from a SIP provider, you must proceed as follows:

# Open one of the four SIP interfaces.
# Enter SIP Account data (ID, STUN server, Account, password).
# Under Registrations, link the SIP registration via H.323 to a PBX object of the Trunk type created beforehand (specification of the GK ID or GK address and the H.323 name or E.164 call number is sufficient).
# Confirm with OK.

A successful registration is displayed in the overview page Administration/Gateway/Interfaces as follows:
{| border=1
! style="background:#DCDCDC;"| State (IP of the SIP provider) !! style="background:#DCDCDC;"| Alias (PBX user object) !! style="background:#DCDCDC;"| Registration (IP of the PBX)
|- valign="top" align="center"
| For example, 217.10.79.9 (sipgate.de)
| H.323 name:E.164 no. SIPTrunk:8
| --> 127.0.0.1
|}

In the example above, the trunk line of the SIP carrier sipgate.de is picked up using the Trunk PBX object with the name SIPTrunk and the call number 8. The dialing of the call number 807031730090 therefore initiates a call at innovaphone AG via the configured SIP carrier.

=== Virtual interfaces (TEST, TONE, HTTP, SIG0/1) ===

The non-configurable, internal interface TEST is only usable as the destination for a call. If a call is received on this interface, the music on hold contained in the non-volatile memory is played. Incoming calls must be in G.729A or G.723 format; other formats are not supported. Suffix dialing digits are ignored.
The internal interface TONE is only usable as the destination for a call. If a call is received on this interface, it is connected and the configured dial tone (Tones) is played. This happens particularly with least-cost-routing scenarios, where the call can only be switched once some of the dialed digits have been analyzed. In the meantime, the dial tone is played via the TONE interface. Suffix dialing digits are ignored. The TONE interface can process several calls.
The non-configurable, internal interface HTTP is only usable as the destination for a call. If a call is received on this interface, music on hold, an announcement or some other spoken information is played from a Web server. The configuration only makes sense in combination with the innovaphone PBX.

The SIG0/SIG1 are virtual interfaces that often are hidden, only are displayed in newer versions and/or devices on test/license mode. This Interfaces are used mainly for SOAP applications be able to perform calls via the Gateway (ex: For automated call tests).

=== Conferencing interface (CONF and SCNF) ===
NB: this is a low-level interface. To implement application level conference rooms, please refer to [[{{NAMESPACE}}:PBX/Objects/Conference]] (v11r1 and up) and [[{{NAMESPACE}}:PBX/Objects/Call Broadcast Conference]].

The CONF interface is currently available on IP6010, IP3010, IP1060, IP0010, IP6000, IP800, IP305, IP3011, IP811 and IP1130. One or more CONF interfaces can be used to create a conferencing unit. Up to 60 (IPxx10), up to 10 (IP800) or up to 4 (IP305) subscribers can be in a single conference. Each call that ends up in a CONF interface takes one conference channel resource plus a DSP resource (there is an exception were DSP it's not used, when using PCM option for an ISDN connection however the incall commands will not work in this mode). A single CONF interface can host multiple conferences, which are identified by a unique number. Conference room ids must not overlap (that is, a room's id must not match the prefix of another id). A single conference cannot span across more than one interface. However, multiple interfaces can be stacked to provide more conferences than one interface's capacity would allow.

'''Note:'''
One DSP channel + One Conference channel is in use for each user in a conference call. We can see this values in the General Info screen of the Gateways

[[Image:Confchannels.png]]

'''Note:'''
Enable the interworking flag of the route to the CONF interface to enable the hold-/retrieve-notify support.

'''Note:'''
Some early IP6000 do not support this feature. Refer to the [[Howto:Multi-Party Conferencing with early IP6000|upgrade details]] if you consider upgrading your hardware.

The behavior of the interface is controlled by various call-setup and in-call commands and remote controls as follows.

==== Call-Setup Commands ====

The called party number in each call setup is interpreted as call-setup command. Also, additional digits received as info-elements are interpreted too. This is why calls must either be sent with ''sending complete'' property or the called party number must be terminated with a '#'. The ''sending complete'' property is activating by setting ''Force Enblock'' in the route toward the CONF interface.

===== Create a new room with a unique room number: *1 =====

This command creates a new conference room with a new, unique id. The syntax is

<code>*1</code>''<options>''<code>#</code> 
or 
<code>*1</code>''<nowiki><prefix></nowiki><nowiki><options></nowiki><code>#</code>''

Valid Options are

*<code>*1</code>''<nowiki><channel-max></nowiki>''
*<code>*2</code>''<nowiki><channel-reserve></nowiki>''
*<code>*3</code>''<nowiki><prefix></nowiki>''
*<code>*4</code>''<nowiki><match></nowiki>''
*<code>*5</code>''<nowiki><id-length></nowiki>''
*<code>*81</code>
*<code>*82</code>
*<code>*83</code>

''<nowiki><channel-max></nowiki>'' is used to set the maximum number of conference channels which shall be allowed in this interface at a time. The limit can be used for example to make sure the CONF interface does not consume all of the DSP channels on a gateway which is used as a trunk line interface too. NB: as of V9 release, you must specify the <code>*1</code>''<nowiki><channel-max></nowiki>'' option correctly. If you omit it, the CONF interface will accept more <code>*2</code>''<nowiki><channel-reserve></nowiki>'' reservations than it is capable to satisfy.

''<nowiki><channel-reserve></nowiki>'' is used to indicate the minimum number of conference channels which shall be available in the new conference room. Only 2 digits may follow <code>*2</code> so that a maximum of 99 channels is possible. If the <code>*2</code> option is used and 2 digits follow, the whole command is considered finished and the trailing <code>#</code> may thus be left out. Please note that due to this, the <code>*2</code> option should be the last option used.

''<nowiki><prefix></nowiki>'' is used as prefix for the conference id created. If more than one CONF interface is used, this prefix can be used to enforce unique conference ids across all interfaces. The prefix may be empty. It may be specified as value for the <code>*3</code> option, or - as a shorthand notation - directly following the initial <code>*1</code> command introducer.

''<nowiki><match></nowiki>'' is a digit string which is used to verify the room number (specified as ''prefix'') requested. If the requested room number does not begin with the specified ''match'', the CONF interface will reject the call with cause code ''no channel available''. This can be used to route calls for fixed room number to the appropriate CONF interface.

''<nowiki><id-length></nowiki>'' specifies the length of the random part of the created room number. It defaults to 6 if not specified.

<code>*81</code> is the ''disconnect control'' option. If the disconnect control option is enabled and the caller of the conference disconnects the conference call, all other calls in this conference room are automatically disconnected.

<code>*82</code> is the ''remote control connect'' option. If an user with the remote control connect option creates a conference room, the conference sends an alerting signal to incoming calls to this conference room instead of a connect signal first and waits for a remote control connect facility for each call before a call is connected.

<code>*83</code> is the ''multi-video off'' option. If a room is created with this option, no multi-video is offered to all calls to this room.

The new room is only created if the required channels can be provided and the maximum number of channels used by the interface is not exceeded. If the room can be created, it is joined, too. The room number (conference id) is returned as the connected number, including both the prefix and the random part. If the conference cannot be created, the call is disconnected with a ''No channel available'' cause code.

Example: <code>*110*124*26#</code> (or <code>*1*310*124*26#</code>) requests the creation of a new conference room with <code>10</code> as conference id prefix (room number). 6 channels are reserved for the conference (so there can be at least 6 participants) and the maximum amount of channels used by the CONF interface is set to 24.

Please note that the CONF interface does not store the <code>*1</code> channel limitation. So you need to make sure it is provided consistently on all calls to the CONF interface that create new conference rooms. Also, both limits (<code>*1</code> and <code>*2</code>) control the resources used by the interface itself. They do not ensure that the resources are not consumed by others when they are actually needed (for example, a physical interface such as ISDN may have used all available DSP channels for VoIP calls).

===== Create a new room with a given room number: *2 =====

This command is the same as above, except that no random number is appended to the ''<nowiki><prefix></nowiki>''. It is not possible to create the conference room if its id conflicts with a room that currently exists in this interface.

Example: <code>*210*124*26#</code> (or <code>*2*310*124*26#</code>) requests the creation of a new conference room with <code>10</code> as conference id (room number). 6 channels are reserved for the conference (so there can be at least 6 participants) and the maximum amount of channels used by the CONF interface is set to 24.

===== Create a new or join an existing room with a given room number: *3 =====

This command is the same as above, except that if there is no room with the given room number a new room is created. Otherwise the existing room is joined. No random number is appended, too.

Example: <code>*310*124*26#</code> (or <code>*3*310*124*26#</code>) requests the join to a conference room with <code>10</code> as conference id (room number) or - if the room does not exist yet - creation of a new one. 6 channels are reserved for the conference (so there can be at least 6 participants) and the maximum amount of channels used by the CONF interface is set to 24.

===== Join an existing room: 0-9 =====

If the called party number does not start with an asterisk <code>*</code>, the remaining digits are interpreted as conference room id and the call will join this conference if it exists.

The dialed digits are used to find an existing room. The first room number which matches is joined (for example, if the called party number is <code>10</code> and there is a conference room with id <code>1</code>). No sending complete or end marker is necessary if the room is found or if there is no such room. However, if the called party number matches only the head of an existing conference room id, the interface will wait for additional digits to decide unless a <code>#</code> is seen or ''sending complete'' is on. You may want to enable ''Force enblock'' in routes towards the CONF interface thus.

For commands which join a room, the call is rejected with a ''No channel available'' cause code if the room does not exist.

The old V8 behavior to create a new conference room instead of joining is only supported with the '*3' command. Now load balancing with multiple conference devices is possible.

==== In-Call commands ====

These commands can be used during calls and must be sent with DTMF tones.

===== Exclusive listen mode: *21# =====

All members except for the caller are muted.

===== Normal listen mode: *6# =====

All members are connected.

==== Innovaphone remote controls ====

An [[Reference:Remote Control Facility | innovaphone remote control facility ]] can be sent e.g. by a SOAP application. These remote controls are available:
* 0: ''Connect''
* 24: ''Receive on''
* 25: ''Receive off''
* 26: ''Exclusive listen mode''
* 27: ''Normal listen mode''

===== Connect (0) =====

Connects a conference call in alerting state made with the call-setup command option ''remote control connect'' (*82).

===== Receive on (24) =====

Switches on receiving incoming voice data of this conference member. This means all other conference members can listen to this conference member. It overwrites a before received ''exclusive listen mode''.

===== Receive off (25) =====

Switches off receiving incoming voice data of this conference member. The conference member is muted for all other conference members.

===== Exclusive listen mode (26) =====

All members except for this conference member are muted. It overwrites a before received ''receive on'' from other conference member calls.

===== Normal listen mode (27) =====

All members are connected. It overwrites a before received ''receive off'' from other conference member calls.

==== Call events ====

These events are recognized if the interworking option for the conference route is activated:

===== Hold event =====

If the call is hold by the member, the conference call is muted.

===== Retrieve event =====

If the call is retrieved again by the member, the conference call is activated.

==== Websocket Registration section ====
Registration to a websocket endpoint - in case of the CONF/SCNF Interface this will be the Registration to the Conference Scaler App (Available with 16r1)

URL: The address of the websocket endpoint

Username: Name used for the registration. If no name and no number is configured a name of the form <product-short-name>-<last 3 Bytes of Mac Address>-<Interface Id> (e.g. IP800-08-03-01-TEL2) is used instead.

Password: Password used for authentication if needed.

=== Conferencing interface (SCNF) ===

Same functionality as CONF, except only G.711 coders are supported.

===== Websocket Registration section=====
Registration to a websocket endpoint - in case of the CONF/SCNF Interface this will be the Registration to the Conference Scaler App (Available with 16r1)

URL: The address of the websocket endpoint

Username: Name used for the registration. If no name and no number is configured a name of the form <product-short-name>-<last 3 Bytes of Mac Address>-<Interface Id> (e.g. IP800-08-03-01-TEL2) is used instead.

Password: Password used for authentication if needed.

=== FAX interface ===

The FAX interface can be used to send or receive fax documents. Each call to this interface must be controlled with user-user-information messages by the [[{{NAMESPACE}}:Concept_App_Service_Fax|App Service Fax]].

==== Headline ====
There are some substitutions in the header line of outgoing faxes. The header line has a length of 108 signs.

* $d: day of month, two digits
* $j: day of month without leading zero
* $m: month, two digits
* $n: month without leading zero
* $Y: year, four digits
* $y: year, two digits
* $G: hour, 24 hours format, without leading zero
* $H: hour, 24 hours format, two digits
* $i: minutes, two digits
* $s: seconds, two digits
* $p: number of page
* $P: page count, if available, with leading '/'
* $I: subscriber id
* $R: remote subscriber id
* $$: the sign '$'
* $t: tabulator to the next 15 character grid
* $c: text is centered
Flags for the place holders:
* 0-9: field length, with tabulator jump forward to this position
* -: left aligned

The default header line for English is defined as:

$d/$m/$Y $G:$i <subscriber-name> $I$63t To:$-20R$tPage: $p$P$c

== CGPN/CDPN Mappings ==
{{CGPN-CDPN_Mappings}}

Reference16r1:Apps/PbxManager/Conference

2026-03-13T06:57:14Z

Nwe:

Configure your Conferences

== Add or edit a conference==
; Name
: Long Name of the PBX object

; SIP
: Name of the PBX object

; Number
: Number of the PBX object

; Allow web access
: If it is set, the web access of a room can be enabled in the conference app by users.

; International number
: External phone number of the PBX. This number is used for the construction of dial-in numbers within meeting invitations.

; Announcements
: Select the announcements to be played.
: Use [[{{NAMESPACE}}:PBX/Objects/Conference/Announcement_types|predefined announcement packages]] depending on language and place them in the local files app.
:
: For example: create a folder conference with a subfolder for the languages that you want to use, e.g. de for German, en for English, nl for Dutch etc. Be aware, that the (sub)folders should not contain capital characters!
: Upload the language audio files in the respective subfolder.
: Share the folder conference with the files key and the language audio files are enabled for selection.

; Conference interface
: Search for available conference devices and select appropriate one for use.
: Possibility to
:* set used amount for seats at dedicated conference device and
:* choose use of PBX-channels-license to operate interface with.

; Node and PBX
: The PBX and node configuration of the PBX object.

; Additional Features
* Conference Scaler
** If the Conference Scaler App is installed this Option will be listed here. Here you can enable the use of the Conference Scaler App - for this you must configure the Conference Scaler App in the Conference Scaler settings plugin.
*** Additionally, you can activate the use of PBX channels licenses

* Virtual background for web access
** All applications configured in the PBX that provide the virtual background support API are listed here. When an application is authorized for the conference object, the external user web access application loads the virtual background feature library.

; Rooms
: Add static room(s) to be used for conferences
:* Display name: Room name displayed as endpoint and as the conference app name.
:* Number: Room number. Appended to the conference number for a direct dial-in.
:* Reserved channels: Amount of conference room participants.
:* PIN: Access authentication to enter the room.
:* Meetings: Creating meetings by users can be allowed or not. It is also possible to use a room only with meetings.
:: Possible options are:
:# Not Allowed: The conference room is only a static room. Scheduled meetings are not possible and the menu of scheduled meeting is hidden in the Conference App.
:# Allowed: The conference room can be used as a static room and for scheduled meetings.
:# Only Meetings: The conference room can be used for scheduled meetings only.
:* Video coder: Supported video codec for video conference.

; Conference Operator
: In addition to static rooms, ad-hoc rooms can be created with a conference operator. This can be reached with its own number and can be secured with a PIN. Voice prompts guide the user through the menu.
:* Number: Number to reach the operator. If no number is configured, a call to the conference object is routed to the operator by default. The operator number must be different from any existing room number.
:* PIN: To restrict access to the operator a 4-digit PIN can be configured. If left empty no PIN is required.
:* Video coder: A video encoder must be selected. If unsure, use VP8.

; Configuration templates
: Each app which is provided by the conference object can be enabled for users with configuration templates.

= Related Articles =
* [[{{NAMESPACE}}:Concept_Conference]]
* [[Courseware:IT_Connect_-_08.1_Conferencing]]

[[Category:Concept Conference]]

Reference16r1:Apps/PbxManager/Conference

2026-03-13T06:56:39Z

Nwe:

Configure your Conferences

== Add or edit a conference==
; Name
: Long Name of the PBX object

; SIP
: Name of the PBX object

; Number
: Number of the PBX object

; Allow web access
: If it is set, the web access of a room can be enabled in the conference app by users.

; International number
: External phone number of the PBX. This number is used for the construction of dial-in numbers within meeting invitations.

; Announcements
: Select the announcements to be played.
: Use [[{{NAMESPACE}}:PBX/Objects/Conference/Announcement_types|predefined announcement packages]] depending on language and place them in the local files app.
:
: For example: create a folder conference with a subfolder for the languages that you want to use, e.g. de for German, en for English, nl for Dutch etc. Be aware, that the (sub)folders should not contain capital characters!
: Upload the language audio files in the respective subfolder.
: Share the folder conference with the files key and the language audio files are enabled for selection.

; Conference interface
: Search for available conference devices and select appropriate one for use.
: Possibility to
:* set used amount for seats at dedicated conference device and
:* choose use of PBX-channels-license to operate interface with.

; Node and PBX
: The PBX and node configuration of the PBX object.

; Additional Features
* Conference Scaler
** If the Conference Scaler App is installed this Option will be listed here. Here you can enable the use of the Conference Scaler App - for this you must configure the Conference Scaler App in the Conference Scaler settings plugin.
*** Select if you want to use PBX-Channels licenses

* Virtual background for web access
** All applications configured in the PBX that provide the virtual background support API are listed here. When an application is authorized for the conference object, the external user web access application loads the virtual background feature library.

; Rooms
: Add static room(s) to be used for conferences
:* Display name: Room name displayed as endpoint and as the conference app name.
:* Number: Room number. Appended to the conference number for a direct dial-in.
:* Reserved channels: Amount of conference room participants.
:* PIN: Access authentication to enter the room.
:* Meetings: Creating meetings by users can be allowed or not. It is also possible to use a room only with meetings.
:: Possible options are:
:# Not Allowed: The conference room is only a static room. Scheduled meetings are not possible and the menu of scheduled meeting is hidden in the Conference App.
:# Allowed: The conference room can be used as a static room and for scheduled meetings.
:# Only Meetings: The conference room can be used for scheduled meetings only.
:* Video coder: Supported video codec for video conference.

; Conference Operator
: In addition to static rooms, ad-hoc rooms can be created with a conference operator. This can be reached with its own number and can be secured with a PIN. Voice prompts guide the user through the menu.
:* Number: Number to reach the operator. If no number is configured, a call to the conference object is routed to the operator by default. The operator number must be different from any existing room number.
:* PIN: To restrict access to the operator a 4-digit PIN can be configured. If left empty no PIN is required.
:* Video coder: A video encoder must be selected. If unsure, use VP8.

; Configuration templates
: Each app which is provided by the conference object can be enabled for users with configuration templates.

= Related Articles =
* [[{{NAMESPACE}}:Concept_Conference]]
* [[Courseware:IT_Connect_-_08.1_Conferencing]]

[[Category:Concept Conference]]

Reference16r1:Apps/PbxManager/Conference

2026-03-13T06:51:05Z

Nwe: Created page with "Configure your Conferences == Add or edit a conference== ; Name : Long Name of the PBX object ; SIP : Name of the PBX object ; Number : Number of the PBX object ; Allow web access : If it is set, the web access of a room can be enabled in the conference app by users. ; International number : External phone number of the PBX. This number is used for the construction of dial-in numbers within meeting invitations. ; Announcements : Select the announcements to be playe..."

Configure your Conferences

== Add or edit a conference==
; Name
: Long Name of the PBX object

; SIP
: Name of the PBX object

; Number
: Number of the PBX object

; Allow web access
: If it is set, the web access of a room can be enabled in the conference app by users.

; International number
: External phone number of the PBX. This number is used for the construction of dial-in numbers within meeting invitations.

; Announcements
: Select the announcements to be played.
: Use [[{{NAMESPACE}}:PBX/Objects/Conference/Announcement_types|predefined announcement packages]] depending on language and place them in the local files app.
:
: For example: create a folder conference with a subfolder for the languages that you want to use, e.g. de for German, en for English, nl for Dutch etc. Be aware, that the (sub)folders should not contain capital characters!
: Upload the language audio files in the respective subfolder.
: Share the folder conference with the files key and the language audio files are enabled for selection.

; Conference interface
: Search for available conference devices and select appropriate one for use.
: Possibility to
:* set used amount for seats at dedicated conference device and
:* choose use of PBX-channels-license to operate interface with.

; Node and PBX
: The PBX and node configuration of the PBX object.

; Additional Features
* Conference Scaler
** If the Conference Scaler App is installed this Option will be listed here. Here you can enable the use of the Conference Scaler App - for this you must configure the Conference Scaler App in the Conference Scaler settings plugin.

* Virtual background for web access
** All applications configured in the PBX that provide the virtual background support API are listed here. When an application is authorized for the conference object, the external user web access application loads the virtual background feature library.

; Rooms
: Add static room(s) to be used for conferences
:* Display name: Room name displayed as endpoint and as the conference app name.
:* Number: Room number. Appended to the conference number for a direct dial-in.
:* Reserved channels: Amount of conference room participants.
:* PIN: Access authentication to enter the room.
:* Meetings: Creating meetings by users can be allowed or not. It is also possible to use a room only with meetings.
:: Possible options are:
:# Not Allowed: The conference room is only a static room. Scheduled meetings are not possible and the menu of scheduled meeting is hidden in the Conference App.
:# Allowed: The conference room can be used as a static room and for scheduled meetings.
:# Only Meetings: The conference room can be used for scheduled meetings only.
:* Video coder: Supported video codec for video conference.

; Conference Operator
: In addition to static rooms, ad-hoc rooms can be created with a conference operator. This can be reached with its own number and can be secured with a PIN. Voice prompts guide the user through the menu.
:* Number: Number to reach the operator. If no number is configured, a call to the conference object is routed to the operator by default. The operator number must be different from any existing room number.
:* PIN: To restrict access to the operator a 4-digit PIN can be configured. If left empty no PIN is required.
:* Video coder: A video encoder must be selected. If unsure, use VP8.

; Configuration templates
: Each app which is provided by the conference object can be enabled for users with configuration templates.

= Related Articles =
* [[{{NAMESPACE}}:Concept_Conference]]
* [[Courseware:IT_Connect_-_08.1_Conferencing]]

[[Category:Concept Conference]]

Reference16r1:Gateway/Interfaces

2026-03-13T06:35:30Z

Nwe:

{{Special:Prefixindex/Reference16r1:Gateway/Interfaces}}

{{FIXME|reason=This article is still work in progress}}

The display of the gateway’s configurable '''interfaces''' is organized in columns:

* '''Interface:''' The name of the interface. Clicking this name opens a popup page, on which all settings can be made. The settings are described in more detail in the following chapter "[[#CGPN.2FCDPN_Mappings|Administration/Gateway/Interfaces/Interface (ISDN & virtual interfaces)]]".
* '''CGPN In, CDPN In, CGPN Out, CDPN Out:''' Precise details on CGPN In, CDPN In, CGPN Out and CDPN Out mappings are contained in the chapter entitled "[[#CGPN.2FCDPN_Mappings|Administration/Gateway/Interfaces/CGPN-CDPN Mappings]]" further down in the text.
* '''State:''' The current state of the interface at physical and at protocol level. Possible states are: ''Up'', ''Down''.
* '''Registration:''' If a terminal has successfully registered with an ISDN, SIP or virtual interface, then this is indicated in this column through specification of the IP address <Name of the VoIP interface:Call number:IP address>.

== Interface (ISDN, SIP & virtual interfaces) ==

Clicking the name of an interface in the Interface column opens a popup page, on which the interfaces can be individually configured. Like the PBX objects, this popup page also contains standard entry fields that occur, more or less, in all interfaces. These standard fields are:

* '''Name:''' The descriptive name of the interface.
* '''Disable:''' A checked check box disables the relevant interface.
* '''Tones:''' The standard calling tone for the relevant interface is set with the Tones list box.
* '''Interface Maps:''' The interface can be configured as a point-to-point connection (Point-to-Point), as a point-to-multipoint connection (Point-to-Multipoint) or manually (Manual) using CGPN/CDPN maps.
See description further down in the text.
* '''Registration:''' With the Registration list box, an H.323 registration or a SIP registration can be initiated for ISDN interfaces. The routes to be handled as incoming and outgoing calls on the relevant interface are automatically created here (see "[[Reference:Administration/Gateway/Routes|Administration/Gateway/Routes]]").

=== ISDN interfaces (PPP, TEL1-4, BRI1-4, PRI1-4) ===

After selection of an '''interface map''', the relevant section is displayed. If Point-to-Point is selected, the Interface Maps Point-to-Point section is displayed:

* '''Area Code:''' The number of the local area (for example, 7031). http://en.wikipedia.org/wiki/List_of_dialling_codes_in_Germany
* '''Subscriber Number:''' The local network number (for example, 73009).
* '''National Prefix:''' The national prefix (for example, 0).
* '''International Prefix:''' The international prefix (for example, 00).

If Trunk Point-to-Multipoint is selected, the Interface Maps Point-to-Multipoint section is displayed:
* '''MSN1-3 / Ext.:''' For every ISDN basic access, several call numbers can be configured. The innovaphone gateways support up to three multiple subscriber numbers (MSN1-3), followed by the extension (Ext.), which represents the extension to which the MSN is to be mapped.
* '''National Prefix:''' The national prefix (for example, 0).
* '''International Prefix:''' The international prefix (for example, 00).

==== Coder Preferences section: ====

After selection of a registration method, the '''Coder Preferences''' section is displayed together with the relevant Registration section.

The standard entry fields in the Coder Preferences section are:

* '''Model:''' The Model list box allows you to select the coder to be used. The coders available for selection are: ''G711A, G711u, G723-53, G729A, G726-32 and XPARENT''. If the remote VoIP device does not support the set coder, a commonly supported coder is used, unless the Exclusive check box was enabled.

Note: The codec ''Clearmode'' is chosen when the bearer capability is not Audio or Speech, that means if the B-channel has 'Unrestricted Digital Information’
means data will be sent via the B-channel - then the codec ''Clearmode'' will be in use (NB: ''Clearmode'' used to be called ''XPARENT'' in previous versions).

* '''Frame:''' Determines the packet size used in transmitting voice data (in ms). Larger packets cause a greater delay in voice data transmission, but cause less load on the network, since the overhead involved in transporting the packets in the network is lower. The higher the packet size used, the lower the bandwidth effectively used.

'''Encoding method | Packet size | Bandwidth'''
---------------------------------------------
G.711 | 30ms | 77kb
G.711 | 90ms | 68kb
---------------------------------------------
G.729 | 30ms | 21kb
G.729 | 90ms | 12kb

* '''Exclusive:''' A checked check box enforces the set encoding (Model), regardless of whether it is supported by the remote VoIP device.
* '''Silence Compression:''' A checked check box enables SC (Silence Compression). With SC, no data is transmitted during pauses in the conversation. This also allows bandwidth to be saved without quality loss.
* '''Enable T.38:''' A checked check box enables the T.38 Fax-over-IP protocol. If a fax machine was connected to the relevant interface, then this check box must be enabled; otherwise, fax transmissions are not handled.
* '''Enable PCM:''' A checked check box enables the PCM switch (Pulse Code Manipulation). Calls from one interface to another interface are then handled directly over the ISDN PCM bus, which in turn saves DSP channels. This entry field is optional and is displayed only in particular devices.
* '''No ICE''' disables ''Interactive Connection Establishment'' (see [[{{NAMESPACE}}:Concept ICE]])

'''Registration section:'''

All non-virtual interfaces additionally have the '''Registration''' section after selection of the registration method.

==== H.323 Registration section ====

The entry fields for an '''H.323 registration''' are:
* '''Gatekeeper Address (primary):''' The primary gatekeeper IP address at which the interface is to register. If the primary gatekeeper is located on the same device, the local IP address 127.0.0.1 can also be entered here.
* '''Gatekeeper Address (secondary):''' The secondary gatekeeper IP address at which the interface is to register, if registration with the primary gatekeeper fails. If the secondary gatekeeper is located on the same device, the local IP address 127.0.0.1 can likewise be entered here.
* '''Gatekeeper ID:''' It is also sufficient to specify only the Gatekeeper ID (see also the chapter entitled "[[Reference:Administration/Gateway/General|Administration/Gateway/General]]").
* '''Name:''' The unique, descriptive H.323 name of the interface or registration.
* '''Number:''' The unique E.164 call number of the interface or registration.
* '''Password / Retype:''' The security of the registration can be raised by specifying a password (Password). The password must be confirmed (Retype).
* '''Supplementary Services (with Feature Codes):''' A checked check box enables the use of additional features (Feature Codes). See description in the chapter entitled "[[Reference:Administration/Gateway/General|Administration/Gateway/General]]".
* '''Dynamic Group:''' A dynamic group can be added to the H.323 registration. Groups can be configured as static, dynamic-in or dynamic-out. For members of static groups, calls are always signaled. It works differently for members of dynamic groups, which register with or unregister from a group dynamically using a function key (Join Group). The difference between dynamic-in and dynamic-out lies in whether the object is to be contained in the relevant group as standard (in) or not (out).
See also description in the chapter entitled "[[Reference:Administration/PBX/Objects|Administration/PBX/Objects]]".
* '''Direct Dial:''' Using Direct Dial, a call setup to the specified call number is initiated as soon as the handset is picked up. A conceivable scenario would be a lift emergency telephone that is connected with the security control room, for example.
* '''Locked White List:''' Here, you can specify a comma-separated list of call numbers that may also be dialed in the case of a locked telephone (for example, emergency services numbers, like 110, 911).

==== SIP Registration section ====

The entry fields for a '''SIP registration''' are:
* '''Server Address (primary):''' The optional IP address of the SIP provider to where the SIP messages (REGISTER,INVITE,etc.) are to be sent. Only necessary if either the IP address cannot be obtained from the SIP URI's domain or a proxy server is to be used.
* '''Server Address (secondary):''' Backup IP address used if the SIP server on the primary IP address does not answer anymore.
* '''ID:''' Here you enter the registration ID followed by the SIP provider domain name (for example 8111111e0@sipgate.de).
* '''STUN Server:''' Only necessary if the SIP server is outside the private network.
* '''Username:''' Username for authorization (only if different from the registration ID).
* '''Password / Retype:''' The password for authorization must be specified here (Password) and confirmed (Retype).

=== FXO interfaces ===

Basically the same configuration is used as for ISDN interfaces.

To receive calling line identification a block-dial route must be used for incoming calls to delay the forwarding of the received call until after the calling line id was received. Please note that you must not use an '!' (exclamation mark) as part of the ''Number Out'' field of the route (see [[{{NAMESPACE}}:Gateway/Routes/Map|Number In/Out]]), as this disables the ''en-bloc'' feature.

=== SIP interfaces (SIP1-4) ===

In addition to the ISDN interfaces (PPP, TEL1-4, BRI1-4, PRI1-4) and virtual interfaces (TEST, TONE, HTTP), there are also four SIP interfaces (SIP1-4), which can be used to obtain a trunk line from a SIP provider, for example. For a description of the entry fields, please refer to the description of the SIP registration above. There are, however, three further entry fields:

* '''Name:''' A descriptive name for the interface.
* '''Disable:''' A switch to temporarily disable this interface without deleting the configuration.
* '''From Header:''' Interoperability option for outgoing calls. Controls the way the CGPN is transmitted to the SIP provider.
** '''AOR:''' The From header contains the fixed registration URI (AOR). The actual calling party number and name will be transmitted inside the ''P-Preferred-Identity'' header (RFC 3325).
** '''AOR with CGPN as display:''' The From header contains the fixed registration URI (AOR) with the calling party number as display string in front of the AOR.
** '''CGPN in user part of URI:''' The From header contains an URI with the calling party number as user part (left from @).
* '''Registration:''' Corresponds to the Registration entry field of the ISDN interfaces. After selection of H.323, the Registration for H.323 section is displayed, enabling registration of this SIP trunk interface with a local innovaphone PBX. After selection of SIP, the Registration for SIP section is displayed, enabling in turn registration with a local non-innovaphone SIP PBX.

To obtain a trunk line from a SIP provider, you must proceed as follows:

# Open one of the four SIP interfaces.
# Enter SIP Account data (ID, STUN server, Account, password).
# Under Registrations, link the SIP registration via H.323 to a PBX object of the Trunk type created beforehand (specification of the GK ID or GK address and the H.323 name or E.164 call number is sufficient).
# Confirm with OK.

A successful registration is displayed in the overview page Administration/Gateway/Interfaces as follows:
{| border=1
! style="background:#DCDCDC;"| State (IP of the SIP provider) !! style="background:#DCDCDC;"| Alias (PBX user object) !! style="background:#DCDCDC;"| Registration (IP of the PBX)
|- valign="top" align="center"
| For example, 217.10.79.9 (sipgate.de)
| H.323 name:E.164 no. SIPTrunk:8
| --> 127.0.0.1
|}

In the example above, the trunk line of the SIP carrier sipgate.de is picked up using the Trunk PBX object with the name SIPTrunk and the call number 8. The dialing of the call number 807031730090 therefore initiates a call at innovaphone AG via the configured SIP carrier.

=== Virtual interfaces (TEST, TONE, HTTP, SIG0/1) ===

The non-configurable, internal interface TEST is only usable as the destination for a call. If a call is received on this interface, the music on hold contained in the non-volatile memory is played. Incoming calls must be in G.729A or G.723 format; other formats are not supported. Suffix dialing digits are ignored.
The internal interface TONE is only usable as the destination for a call. If a call is received on this interface, it is connected and the configured dial tone (Tones) is played. This happens particularly with least-cost-routing scenarios, where the call can only be switched once some of the dialed digits have been analyzed. In the meantime, the dial tone is played via the TONE interface. Suffix dialing digits are ignored. The TONE interface can process several calls.
The non-configurable, internal interface HTTP is only usable as the destination for a call. If a call is received on this interface, music on hold, an announcement or some other spoken information is played from a Web server. The configuration only makes sense in combination with the innovaphone PBX.

The SIG0/SIG1 are virtual interfaces that often are hidden, only are displayed in newer versions and/or devices on test/license mode. This Interfaces are used mainly for SOAP applications be able to perform calls via the Gateway (ex: For automated call tests).

=== Conferencing interface (CONF and SCNF) ===
NB: this is a low-level interface. To implement application level conference rooms, please refer to [[{{NAMESPACE}}:PBX/Objects/Conference]] (v11r1 and up) and [[{{NAMESPACE}}:PBX/Objects/Call Broadcast Conference]].

The CONF interface is currently available on IP6010, IP3010, IP1060, IP0010, IP6000, IP800, IP305, IP3011, IP811 and IP1130. One or more CONF interfaces can be used to create a conferencing unit. Up to 60 (IPxx10), up to 10 (IP800) or up to 4 (IP305) subscribers can be in a single conference. Each call that ends up in a CONF interface takes one conference channel resource plus a DSP resource (there is an exception were DSP it's not used, when using PCM option for an ISDN connection however the incall commands will not work in this mode). A single CONF interface can host multiple conferences, which are identified by a unique number. Conference room ids must not overlap (that is, a room's id must not match the prefix of another id). A single conference cannot span across more than one interface. However, multiple interfaces can be stacked to provide more conferences than one interface's capacity would allow.

'''Note:'''
One DSP channel + One Conference channel is in use for each user in a conference call. We can see this values in the General Info screen of the Gateways

[[Image:Confchannels.png]]

'''Note:'''
Enable the interworking flag of the route to the CONF interface to enable the hold-/retrieve-notify support.

'''Note:'''
Some early IP6000 do not support this feature. Refer to the [[Howto:Multi-Party Conferencing with early IP6000|upgrade details]] if you consider upgrading your hardware.

The behavior of the interface is controlled by various call-setup and in-call commands and remote controls as follows.

==== Call-Setup Commands ====

The called party number in each call setup is interpreted as call-setup command. Also, additional digits received as info-elements are interpreted too. This is why calls must either be sent with ''sending complete'' property or the called party number must be terminated with a '#'. The ''sending complete'' property is activating by setting ''Force Enblock'' in the route toward the CONF interface.

===== Create a new room with a unique room number: *1 =====

This command creates a new conference room with a new, unique id. The syntax is

<code>*1</code>''<options>''<code>#</code> 
or 
<code>*1</code>''<nowiki><prefix></nowiki><nowiki><options></nowiki><code>#</code>''

Valid Options are

*<code>*1</code>''<nowiki><channel-max></nowiki>''
*<code>*2</code>''<nowiki><channel-reserve></nowiki>''
*<code>*3</code>''<nowiki><prefix></nowiki>''
*<code>*4</code>''<nowiki><match></nowiki>''
*<code>*5</code>''<nowiki><id-length></nowiki>''
*<code>*81</code>
*<code>*82</code>
*<code>*83</code>

''<nowiki><channel-max></nowiki>'' is used to set the maximum number of conference channels which shall be allowed in this interface at a time. The limit can be used for example to make sure the CONF interface does not consume all of the DSP channels on a gateway which is used as a trunk line interface too. NB: as of V9 release, you must specify the <code>*1</code>''<nowiki><channel-max></nowiki>'' option correctly. If you omit it, the CONF interface will accept more <code>*2</code>''<nowiki><channel-reserve></nowiki>'' reservations than it is capable to satisfy.

''<nowiki><channel-reserve></nowiki>'' is used to indicate the minimum number of conference channels which shall be available in the new conference room. Only 2 digits may follow <code>*2</code> so that a maximum of 99 channels is possible. If the <code>*2</code> option is used and 2 digits follow, the whole command is considered finished and the trailing <code>#</code> may thus be left out. Please note that due to this, the <code>*2</code> option should be the last option used.

''<nowiki><prefix></nowiki>'' is used as prefix for the conference id created. If more than one CONF interface is used, this prefix can be used to enforce unique conference ids across all interfaces. The prefix may be empty. It may be specified as value for the <code>*3</code> option, or - as a shorthand notation - directly following the initial <code>*1</code> command introducer.

''<nowiki><match></nowiki>'' is a digit string which is used to verify the room number (specified as ''prefix'') requested. If the requested room number does not begin with the specified ''match'', the CONF interface will reject the call with cause code ''no channel available''. This can be used to route calls for fixed room number to the appropriate CONF interface.

''<nowiki><id-length></nowiki>'' specifies the length of the random part of the created room number. It defaults to 6 if not specified.

<code>*81</code> is the ''disconnect control'' option. If the disconnect control option is enabled and the caller of the conference disconnects the conference call, all other calls in this conference room are automatically disconnected.

<code>*82</code> is the ''remote control connect'' option. If an user with the remote control connect option creates a conference room, the conference sends an alerting signal to incoming calls to this conference room instead of a connect signal first and waits for a remote control connect facility for each call before a call is connected.

<code>*83</code> is the ''multi-video off'' option. If a room is created with this option, no multi-video is offered to all calls to this room.

The new room is only created if the required channels can be provided and the maximum number of channels used by the interface is not exceeded. If the room can be created, it is joined, too. The room number (conference id) is returned as the connected number, including both the prefix and the random part. If the conference cannot be created, the call is disconnected with a ''No channel available'' cause code.

Example: <code>*110*124*26#</code> (or <code>*1*310*124*26#</code>) requests the creation of a new conference room with <code>10</code> as conference id prefix (room number). 6 channels are reserved for the conference (so there can be at least 6 participants) and the maximum amount of channels used by the CONF interface is set to 24.

Please note that the CONF interface does not store the <code>*1</code> channel limitation. So you need to make sure it is provided consistently on all calls to the CONF interface that create new conference rooms. Also, both limits (<code>*1</code> and <code>*2</code>) control the resources used by the interface itself. They do not ensure that the resources are not consumed by others when they are actually needed (for example, a physical interface such as ISDN may have used all available DSP channels for VoIP calls).

===== Create a new room with a given room number: *2 =====

This command is the same as above, except that no random number is appended to the ''<nowiki><prefix></nowiki>''. It is not possible to create the conference room if its id conflicts with a room that currently exists in this interface.

Example: <code>*210*124*26#</code> (or <code>*2*310*124*26#</code>) requests the creation of a new conference room with <code>10</code> as conference id (room number). 6 channels are reserved for the conference (so there can be at least 6 participants) and the maximum amount of channels used by the CONF interface is set to 24.

===== Create a new or join an existing room with a given room number: *3 =====

This command is the same as above, except that if there is no room with the given room number a new room is created. Otherwise the existing room is joined. No random number is appended, too.

Example: <code>*310*124*26#</code> (or <code>*3*310*124*26#</code>) requests the join to a conference room with <code>10</code> as conference id (room number) or - if the room does not exist yet - creation of a new one. 6 channels are reserved for the conference (so there can be at least 6 participants) and the maximum amount of channels used by the CONF interface is set to 24.

===== Join an existing room: 0-9 =====

If the called party number does not start with an asterisk <code>*</code>, the remaining digits are interpreted as conference room id and the call will join this conference if it exists.

The dialed digits are used to find an existing room. The first room number which matches is joined (for example, if the called party number is <code>10</code> and there is a conference room with id <code>1</code>). No sending complete or end marker is necessary if the room is found or if there is no such room. However, if the called party number matches only the head of an existing conference room id, the interface will wait for additional digits to decide unless a <code>#</code> is seen or ''sending complete'' is on. You may want to enable ''Force enblock'' in routes towards the CONF interface thus.

For commands which join a room, the call is rejected with a ''No channel available'' cause code if the room does not exist.

The old V8 behavior to create a new conference room instead of joining is only supported with the '*3' command. Now load balancing with multiple conference devices is possible.

==== In-Call commands ====

These commands can be used during calls and must be sent with DTMF tones.

===== Exclusive listen mode: *21# =====

All members except for the caller are muted.

===== Normal listen mode: *6# =====

All members are connected.

==== Innovaphone remote controls ====

An [[Reference:Remote Control Facility | innovaphone remote control facility ]] can be sent e.g. by a SOAP application. These remote controls are available:
* 0: ''Connect''
* 24: ''Receive on''
* 25: ''Receive off''
* 26: ''Exclusive listen mode''
* 27: ''Normal listen mode''

===== Connect (0) =====

Connects a conference call in alerting state made with the call-setup command option ''remote control connect'' (*82).

===== Receive on (24) =====

Switches on receiving incoming voice data of this conference member. This means all other conference members can listen to this conference member. It overwrites a before received ''exclusive listen mode''.

===== Receive off (25) =====

Switches off receiving incoming voice data of this conference member. The conference member is muted for all other conference members.

===== Exclusive listen mode (26) =====

All members except for this conference member are muted. It overwrites a before received ''receive on'' from other conference member calls.

===== Normal listen mode (27) =====

All members are connected. It overwrites a before received ''receive off'' from other conference member calls.

==== Call events ====

These events are recognized if the interworking option for the conference route is activated:

===== Hold event =====

If the call is hold by the member, the conference call is muted.

===== Retrieve event =====

If the call is retrieved again by the member, the conference call is activated.

==== Websocket Registration section ====
Registration to a websocket endpoint - in case of the CONF/SCNF Interface this will be the Registration to the Conference Scaler App (Available with 16r1)

URL: The address of the websocket endpoint

Username: Name used for the registration. If no name and no number is configured a name of the form <product-short-name>-<last 3 Bytes of Mac Address>-<Interface Id> (e.g. IP800-08-03-01-TEL2) is used instead.

Password: Password used for authentication if needed.

=== Conferencing interface (SCNF) ===

Same functionality as CONF, except only G.711 coders are supported.

===== Websocket Registration section=====
Registration to a websocket endpoint - in case of the CONF/SCNF Interface this will be the Registration to the Conference Scaler App (Available with 16r1)

URL: The address of the websocket endpoint

Username: Name used for the registration. If no name and no number is configured a name of the form <product-short-name>-<last 3 Bytes of Mac Address>-<Interface Id> (e.g. IP800-08-03-01-TEL2) is used instead.

Password: Password used for authentication if needed.

=== FAX interface ===

The FAX interface can be used to send or receive fax documents. Each call to this interface must be controlled with user-user-information messages by the [[{{NAMESPACE}}:Concept_App_Service_Fax|App Service Fax]].

==== Headline ====
There are some substitutions in the header line of outgoing faxes. The header line has a length of 108 signs.

* $d: day of month, two digits
* $j: day of month without leading zero
* $m: month, two digits
* $n: month without leading zero
* $Y: year, four digits
* $y: year, two digits
* $G: hour, 24 hours format, without leading zero
* $H: hour, 24 hours format, two digits
* $i: minutes, two digits
* $s: seconds, two digits
* $p: number of page
* $P: page count, if available, with leading '/'
* $I: subscriber id
* $R: remote subscriber id
* $$: the sign '$'
* $t: tabulator to the next 15 character grid
* $c: text is centered
Flags for the place holders:
* 0-9: field length, with tabulator jump forward to this position
* -: left aligned

The default header line for English is defined as:

$d/$m/$Y $G:$i <subscriber-name> $I$63t To:$-20R$tPage: $p$P$c

== CGPN/CDPN Mappings ==
{{CGPN-CDPN_Mappings}}

Reference16r1:Concept App Service Conference Scaler

2026-03-12T15:17:10Z

Nwe: /* Restrictions / Known issues */

{{FIXME|reason=This product is in the beta phase and is not yet finished}}

[[Category:Concept|Apps]]

The Conference Scaler app is required to use multiple conference interfaces within one conference.

== Applies to ==
* Conference Scaler app

== Requirements ==
* innovaphone PBX
* innovaphone Application Platform
* V16r1
* At least one innovaphone Gateway with CONF/SCNF Interfaces
* The Devices API must be available for the settings plugin therefore the gateways have to be connected to devices

== Setting Plugins ==
=== Conference Scaler ===
With the Conference Scaler settings plugin the Conference Scaler app object can be created, edited and deleted on the PBX.
Furthermore, the conference interface to be used can be configured if the Devices API is available.
=== Conference ===
In the conference plugin, the menu item "Additional Features" appears for conferences. With this setting you can enable the use of the Conference Scaler within the conference object. Additionally, you can activate the use of PBX channels licenses.

== More information ==
=== Conference Scaler App (innovaphone-conferencescaler) ===
[[File:Conferencescaler.png|110px|conferencescaler.png/|conferencescaler.png/]]

The hidden app offers an interface via the Service API (com.innovaphone.conferencescaler), that is used by the conference object.

Parameters:
;URL: <pre>http://<ap.domain.tld>/<domain.tld>/<instance-name>/innovaphone-conferencescaler</pre>

== Features ==
* Allows the use of multiple conference interfaces within multiple conference
** This removes previous participant limits
** Adds flexibility to existing infrastructure
** The Conference Scaler app nominates one conference interface which will manage the control for voice streams. This conference interface will then receive and distribute the audio streams to all interfaces which are connected to the Conference Scaler app
** The Conference Scaler app will split the participants in a Round Robin maner, therefore the load will be evenly distributed to the interfaces
** If an interface has reached its limit for participants, the Conference Scaler app will then select a different and free Interface for a new incoming call

== Configuration ==
# Download the Conference Scaler app via App Store. (or use the Settings Plugin "app installer" to install the App + Instance automatically and skip to step 5)
# Install the Conference Scaler app on the App Platform.
# Create an instance in the Conference Scaler app on the App Platform.
# Make sure the Instance is running.
# Create via the settings plugin a new Conference Scaler app object.
# Select the Conference Interfaces to be used by the Conference Scaler app.
# Allow the use of the Conference Scaler app on the Conference Object settings.

== Restrictions / Known issues ==
* Currently, for technical reasons, the Conference Scaler app object requires video and app sharing licenses or a UC license, which should be configured in the PBX Advanced UI
* Using the Conference Scaler app generates an overhead depending on usage patterns, so the increase in the number of participants does not correspond to a multiple of x interfaces. (One participant will take up an estimation of 1,5x CPU-Load as the usual conference flow (because data is additionally send to a different interface). So for an interface with max. 100 Video participants this amount will decrease to a maximum participant limit of 66 (as 100/1,5=66). Therefore if you want to achieve a maximum of X video participants you need to take that information into account for calculating the needed interfaces)
* In case of failure of the App Platform the Conference Scaler app will not be available as well. Therefore it will not be possible to join or continue an ongoing conference.
* Room reservations are not possible when using the Conference Scaler app
* Video sharing between the conference interfaces is handled via multicast and this requires that the interfaces are on the same network (Multicast addresses differs between 239.0.0.0 - 239.255.255.255)
* Only SCNF interfaces are currently supported
* For now only Hardware Gateways are supported, no IPVAs are able to be used currently

== Troubleshooting ==
If you can reproduce the problem, take a screenshot (depending on the issue), save the browser logs and the app logs, and send them as an attachment to your support ticket.

== Related Articles ==
* [[Reference15r1:Concept_Conference_Web_Access]]
* [[Reference15r1:Concept_Conference]]
* [[Courseware:IT_Connect_-_08.1_Conferencing]]
* [[Courseware:IT_Plus_-_Public_Access_to_Conference_Resources#Bandwidth_and_CPU_consideration]]

Reference16r1:Concept App Service Conference Scaler

2026-03-12T14:41:49Z

Nwe: /* Restrictions / Known issues */

{{FIXME|reason=This product is in the beta phase and is not yet finished}}

[[Category:Concept|Apps]]

The Conference Scaler app is required to use multiple conference interfaces within one conference.

== Applies to ==
* Conference Scaler app

== Requirements ==
* innovaphone PBX
* innovaphone Application Platform
* V16r1
* At least one innovaphone Gateway with CONF/SCNF Interfaces
* The Devices API must be available for the settings plugin therefore the gateways have to be connected to devices

== Setting Plugins ==
=== Conference Scaler ===
With the Conference Scaler settings plugin the Conference Scaler app object can be created, edited and deleted on the PBX.
Furthermore, the conference interface to be used can be configured if the Devices API is available.
=== Conference ===
In the conference plugin, the menu item "Additional Features" appears for conferences. With this setting you can enable the use of the Conference Scaler within the conference object. Additionally, you can activate the use of PBX channels licenses.

== More information ==
=== Conference Scaler App (innovaphone-conferencescaler) ===
[[File:Conferencescaler.png|110px|conferencescaler.png/]]

The hidden app offers an interface via the Service API (com.innovaphone.conferencescaler), that is used by the conference object.

Parameters:
;URL: <pre>http://<ap.domain.tld>/<domain.tld>/<instance-name>/innovaphone-conferencescaler</pre>

== Features ==
* Allows the use of multiple conference interfaces within multiple conference
** This removes previous participant limits
** Adds flexibility to existing infrastructure
** The Conference Scaler app nominates one conference interface which will manage the control for voice streams. This conference interface will then receive and distribute the audio streams to all interfaces which are connected to the Conference Scaler app
** The Conference Scaler app will split the participants in a Round Robin maner, therefore the load will be evenly distributed to the interfaces
** If an interface has reached its limit for participants, the Conference Scaler app will then select a different and free Interface for a new incoming call

== Configuration ==
# Download the Conference Scaler app via App Store. (or use the Settings Plugin "app installer" to install the App + Instance automatically and skip to step 5)
# Install the Conference Scaler app on the App Platform.
# Create an instance in the Conference Scaler app on the App Platform.
# Make sure the Instance is running.
# Create via the settings plugin a new Conference Scaler app object.
# Select the Conference Interfaces to be used by the Conference Scaler app.
# Allow the use of the Conference Scaler app on the Conference Object settings.

== Restrictions / Known issues ==
* Currently, for technical reasons, the Conference Scaler app object requires video and app sharing licenses or a UC license, which should be configured in the PBX Advanced UI
* Using the Conference Scaler app generates an overhead depending on usage patterns, so the increase in the number of participants does not correspond to a multiple of x interfaces. (One participant will take up an estimation of 1,5x CPU-Load as the usual conference flow (because data is additionally send to a different interface). So for an interface with max. 100 Video participants this amount will decrease to a maximum participant limit of 66 (as 100/1,5=66). Therefore if you want to achieve a maximum of X video participants you need to take that information into account for calculating the needed interfaces)
* In case of failure of the App Platform the Conference Scaler app will not be available as well. Therefore it will not be possible to join or continue an ongoing conference.
* Room reservations are not possible when using the Conference Scaler app
* Video sharing between the conference interfaces is handled via multicast and this requires that the interfaces are on the same network (Multicast addresses differs between 239.0.0.0 - 239.255.255.255)
* Only SCNF interfaces are currently supported

== Troubleshooting ==
If you can reproduce the problem, take a screenshot (depending on the issue), save the browser logs and the app logs, and send them as an attachment to your support ticket.

== Related Articles ==
* [[Reference15r1:Concept_Conference_Web_Access]]
* [[Reference15r1:Concept_Conference]]
* [[Courseware:IT_Connect_-_08.1_Conferencing]]
* [[Courseware:IT_Plus_-_Public_Access_to_Conference_Resources#Bandwidth_and_CPU_consideration]]

Reference16r1:Gateway/Interfaces

2026-03-09T13:00:17Z

Nwe: Created page with "{{Special:Prefixindex/Reference16r1:Gateway/Interfaces}} {{FIXME|reason=This article is still work in progress}} The display of the gateway’s configurable '''interfaces''' is organized in columns: * '''Interface:''' The name of the interface. Clicking this name opens a popup page, on which all settings can be made. The settings are described in more detail in the following chapter "#CGPN.2FCDPN_Mappings|Administration/Gateway/Interfaces/Interface (ISDN & virtual i..."

{{Special:Prefixindex/Reference16r1:Gateway/Interfaces}}

{{FIXME|reason=This article is still work in progress}}

The display of the gateway’s configurable '''interfaces''' is organized in columns:

* '''Interface:''' The name of the interface. Clicking this name opens a popup page, on which all settings can be made. The settings are described in more detail in the following chapter "[[#CGPN.2FCDPN_Mappings|Administration/Gateway/Interfaces/Interface (ISDN & virtual interfaces)]]".
* '''CGPN In, CDPN In, CGPN Out, CDPN Out:''' Precise details on CGPN In, CDPN In, CGPN Out and CDPN Out mappings are contained in the chapter entitled "[[#CGPN.2FCDPN_Mappings|Administration/Gateway/Interfaces/CGPN-CDPN Mappings]]" further down in the text.
* '''State:''' The current state of the interface at physical and at protocol level. Possible states are: ''Up'', ''Down''.
* '''Registration:''' If a terminal has successfully registered with an ISDN, SIP or virtual interface, then this is indicated in this column through specification of the IP address <Name of the VoIP interface:Call number:IP address>.

== Interface (ISDN, SIP & virtual interfaces) ==

Clicking the name of an interface in the Interface column opens a popup page, on which the interfaces can be individually configured. Like the PBX objects, this popup page also contains standard entry fields that occur, more or less, in all interfaces. These standard fields are:

* '''Name:''' The descriptive name of the interface.
* '''Disable:''' A checked check box disables the relevant interface.
* '''Tones:''' The standard calling tone for the relevant interface is set with the Tones list box.
* '''Interface Maps:''' The interface can be configured as a point-to-point connection (Point-to-Point), as a point-to-multipoint connection (Point-to-Multipoint) or manually (Manual) using CGPN/CDPN maps.
See description further down in the text.
* '''Registration:''' With the Registration list box, an H.323 registration or a SIP registration can be initiated for ISDN interfaces. The routes to be handled as incoming and outgoing calls on the relevant interface are automatically created here (see "[[Reference:Administration/Gateway/Routes|Administration/Gateway/Routes]]").

=== ISDN interfaces (PPP, TEL1-4, BRI1-4, PRI1-4) ===

After selection of an '''interface map''', the relevant section is displayed. If Point-to-Point is selected, the Interface Maps Point-to-Point section is displayed:

* '''Area Code:''' The number of the local area (for example, 7031). http://en.wikipedia.org/wiki/List_of_dialling_codes_in_Germany
* '''Subscriber Number:''' The local network number (for example, 73009).
* '''National Prefix:''' The national prefix (for example, 0).
* '''International Prefix:''' The international prefix (for example, 00).

If Trunk Point-to-Multipoint is selected, the Interface Maps Point-to-Multipoint section is displayed:
* '''MSN1-3 / Ext.:''' For every ISDN basic access, several call numbers can be configured. The innovaphone gateways support up to three multiple subscriber numbers (MSN1-3), followed by the extension (Ext.), which represents the extension to which the MSN is to be mapped.
* '''National Prefix:''' The national prefix (for example, 0).
* '''International Prefix:''' The international prefix (for example, 00).

==== Coder Preferences section: ====

After selection of a registration method, the '''Coder Preferences''' section is displayed together with the relevant Registration section.

The standard entry fields in the Coder Preferences section are:

* '''Model:''' The Model list box allows you to select the coder to be used. The coders available for selection are: ''G711A, G711u, G723-53, G729A, G726-32 and XPARENT''. If the remote VoIP device does not support the set coder, a commonly supported coder is used, unless the Exclusive check box was enabled.

Note: The codec ''Clearmode'' is chosen when the bearer capability is not Audio or Speech, that means if the B-channel has 'Unrestricted Digital Information’
means data will be sent via the B-channel - then the codec ''Clearmode'' will be in use (NB: ''Clearmode'' used to be called ''XPARENT'' in previous versions).

* '''Frame:''' Determines the packet size used in transmitting voice data (in ms). Larger packets cause a greater delay in voice data transmission, but cause less load on the network, since the overhead involved in transporting the packets in the network is lower. The higher the packet size used, the lower the bandwidth effectively used.

'''Encoding method | Packet size | Bandwidth'''
---------------------------------------------
G.711 | 30ms | 77kb
G.711 | 90ms | 68kb
---------------------------------------------
G.729 | 30ms | 21kb
G.729 | 90ms | 12kb

* '''Exclusive:''' A checked check box enforces the set encoding (Model), regardless of whether it is supported by the remote VoIP device.
* '''Silence Compression:''' A checked check box enables SC (Silence Compression). With SC, no data is transmitted during pauses in the conversation. This also allows bandwidth to be saved without quality loss.
* '''Enable T.38:''' A checked check box enables the T.38 Fax-over-IP protocol. If a fax machine was connected to the relevant interface, then this check box must be enabled; otherwise, fax transmissions are not handled.
* '''Enable PCM:''' A checked check box enables the PCM switch (Pulse Code Manipulation). Calls from one interface to another interface are then handled directly over the ISDN PCM bus, which in turn saves DSP channels. This entry field is optional and is displayed only in particular devices.
* '''No ICE''' disables ''Interactive Connection Establishment'' (see [[{{NAMESPACE}}:Concept ICE]])

'''Registration section:'''

All non-virtual interfaces additionally have the '''Registration''' section after selection of the registration method.

==== H.323 Registration section ====

The entry fields for an '''H.323 registration''' are:
* '''Gatekeeper Address (primary):''' The primary gatekeeper IP address at which the interface is to register. If the primary gatekeeper is located on the same device, the local IP address 127.0.0.1 can also be entered here.
* '''Gatekeeper Address (secondary):''' The secondary gatekeeper IP address at which the interface is to register, if registration with the primary gatekeeper fails. If the secondary gatekeeper is located on the same device, the local IP address 127.0.0.1 can likewise be entered here.
* '''Gatekeeper ID:''' It is also sufficient to specify only the Gatekeeper ID (see also the chapter entitled "[[Reference:Administration/Gateway/General|Administration/Gateway/General]]").
* '''Name:''' The unique, descriptive H.323 name of the interface or registration.
* '''Number:''' The unique E.164 call number of the interface or registration.
* '''Password / Retype:''' The security of the registration can be raised by specifying a password (Password). The password must be confirmed (Retype).
* '''Supplementary Services (with Feature Codes):''' A checked check box enables the use of additional features (Feature Codes). See description in the chapter entitled "[[Reference:Administration/Gateway/General|Administration/Gateway/General]]".
* '''Dynamic Group:''' A dynamic group can be added to the H.323 registration. Groups can be configured as static, dynamic-in or dynamic-out. For members of static groups, calls are always signaled. It works differently for members of dynamic groups, which register with or unregister from a group dynamically using a function key (Join Group). The difference between dynamic-in and dynamic-out lies in whether the object is to be contained in the relevant group as standard (in) or not (out).
See also description in the chapter entitled "[[Reference:Administration/PBX/Objects|Administration/PBX/Objects]]".
* '''Direct Dial:''' Using Direct Dial, a call setup to the specified call number is initiated as soon as the handset is picked up. A conceivable scenario would be a lift emergency telephone that is connected with the security control room, for example.
* '''Locked White List:''' Here, you can specify a comma-separated list of call numbers that may also be dialed in the case of a locked telephone (for example, emergency services numbers, like 110, 911).

==== SIP Registration section ====

The entry fields for a '''SIP registration''' are:
* '''Server Address (primary):''' The optional IP address of the SIP provider to where the SIP messages (REGISTER,INVITE,etc.) are to be sent. Only necessary if either the IP address cannot be obtained from the SIP URI's domain or a proxy server is to be used.
* '''Server Address (secondary):''' Backup IP address used if the SIP server on the primary IP address does not answer anymore.
* '''ID:''' Here you enter the registration ID followed by the SIP provider domain name (for example 8111111e0@sipgate.de).
* '''STUN Server:''' Only necessary if the SIP server is outside the private network.
* '''Username:''' Username for authorization (only if different from the registration ID).
* '''Password / Retype:''' The password for authorization must be specified here (Password) and confirmed (Retype).

=== FXO interfaces ===

Basically the same configuration is used as for ISDN interfaces.

To receive calling line identification a block-dial route must be used for incoming calls to delay the forwarding of the received call until after the calling line id was received. Please note that you must not use an '!' (exclamation mark) as part of the ''Number Out'' field of the route (see [[{{NAMESPACE}}:Gateway/Routes/Map|Number In/Out]]), as this disables the ''en-bloc'' feature.

=== SIP interfaces (SIP1-4) ===

In addition to the ISDN interfaces (PPP, TEL1-4, BRI1-4, PRI1-4) and virtual interfaces (TEST, TONE, HTTP), there are also four SIP interfaces (SIP1-4), which can be used to obtain a trunk line from a SIP provider, for example. For a description of the entry fields, please refer to the description of the SIP registration above. There are, however, three further entry fields:

* '''Name:''' A descriptive name for the interface.
* '''Disable:''' A switch to temporarily disable this interface without deleting the configuration.
* '''From Header:''' Interoperability option for outgoing calls. Controls the way the CGPN is transmitted to the SIP provider.
** '''AOR:''' The From header contains the fixed registration URI (AOR). The actual calling party number and name will be transmitted inside the ''P-Preferred-Identity'' header (RFC 3325).
** '''AOR with CGPN as display:''' The From header contains the fixed registration URI (AOR) with the calling party number as display string in front of the AOR.
** '''CGPN in user part of URI:''' The From header contains an URI with the calling party number as user part (left from @).
* '''Registration:''' Corresponds to the Registration entry field of the ISDN interfaces. After selection of H.323, the Registration for H.323 section is displayed, enabling registration of this SIP trunk interface with a local innovaphone PBX. After selection of SIP, the Registration for SIP section is displayed, enabling in turn registration with a local non-innovaphone SIP PBX.

To obtain a trunk line from a SIP provider, you must proceed as follows:

# Open one of the four SIP interfaces.
# Enter SIP Account data (ID, STUN server, Account, password).
# Under Registrations, link the SIP registration via H.323 to a PBX object of the Trunk type created beforehand (specification of the GK ID or GK address and the H.323 name or E.164 call number is sufficient).
# Confirm with OK.

A successful registration is displayed in the overview page Administration/Gateway/Interfaces as follows:
{| border=1
! style="background:#DCDCDC;"| State (IP of the SIP provider) !! style="background:#DCDCDC;"| Alias (PBX user object) !! style="background:#DCDCDC;"| Registration (IP of the PBX)
|- valign="top" align="center"
| For example, 217.10.79.9 (sipgate.de)
| H.323 name:E.164 no. SIPTrunk:8
| --> 127.0.0.1
|}

In the example above, the trunk line of the SIP carrier sipgate.de is picked up using the Trunk PBX object with the name SIPTrunk and the call number 8. The dialing of the call number 807031730090 therefore initiates a call at innovaphone AG via the configured SIP carrier.

=== Virtual interfaces (TEST, TONE, HTTP, SIG0/1) ===

The non-configurable, internal interface TEST is only usable as the destination for a call. If a call is received on this interface, the music on hold contained in the non-volatile memory is played. Incoming calls must be in G.729A or G.723 format; other formats are not supported. Suffix dialing digits are ignored.
The internal interface TONE is only usable as the destination for a call. If a call is received on this interface, it is connected and the configured dial tone (Tones) is played. This happens particularly with least-cost-routing scenarios, where the call can only be switched once some of the dialed digits have been analyzed. In the meantime, the dial tone is played via the TONE interface. Suffix dialing digits are ignored. The TONE interface can process several calls.
The non-configurable, internal interface HTTP is only usable as the destination for a call. If a call is received on this interface, music on hold, an announcement or some other spoken information is played from a Web server. The configuration only makes sense in combination with the innovaphone PBX.

The SIG0/SIG1 are virtual interfaces that often are hidden, only are displayed in newer versions and/or devices on test/license mode. This Interfaces are used mainly for SOAP applications be able to perform calls via the Gateway (ex: For automated call tests).

=== Conferencing interface (CONF and SCNF) ===
NB: this is a low-level interface. To implement application level conference rooms, please refer to [[{{NAMESPACE}}:PBX/Objects/Conference]] (v11r1 and up) and [[{{NAMESPACE}}:PBX/Objects/Call Broadcast Conference]].

The CONF interface is currently available on IP6010, IP3010, IP1060, IP0010, IP6000, IP800, IP305, IP3011, IP811 and IP1130. One or more CONF interfaces can be used to create a conferencing unit. Up to 60 (IPxx10), up to 10 (IP800) or up to 4 (IP305) subscribers can be in a single conference. Each call that ends up in a CONF interface takes one conference channel resource plus a DSP resource (there is an exception were DSP it's not used, when using PCM option for an ISDN connection however the incall commands will not work in this mode). A single CONF interface can host multiple conferences, which are identified by a unique number. Conference room ids must not overlap (that is, a room's id must not match the prefix of another id). A single conference cannot span across more than one interface. However, multiple interfaces can be stacked to provide more conferences than one interface's capacity would allow.

'''Note:'''
One DSP channel + One Conference channel is in use for each user in a conference call. We can see this values in the General Info screen of the Gateways

[[Image:Confchannels.png]]

'''Note:'''
Enable the interworking flag of the route to the CONF interface to enable the hold-/retrieve-notify support.

'''Note:'''
Some early IP6000 do not support this feature. Refer to the [[Howto:Multi-Party Conferencing with early IP6000|upgrade details]] if you consider upgrading your hardware.

The behavior of the interface is controlled by various call-setup and in-call commands and remote controls as follows.

==== Call-Setup Commands ====

The called party number in each call setup is interpreted as call-setup command. Also, additional digits received as info-elements are interpreted too. This is why calls must either be sent with ''sending complete'' property or the called party number must be terminated with a '#'. The ''sending complete'' property is activating by setting ''Force Enblock'' in the route toward the CONF interface.

===== Create a new room with a unique room number: *1 =====

This command creates a new conference room with a new, unique id. The syntax is

<code>*1</code>''<options>''<code>#</code> 
or 
<code>*1</code>''<nowiki><prefix></nowiki><nowiki><options></nowiki><code>#</code>''

Valid Options are

*<code>*1</code>''<nowiki><channel-max></nowiki>''
*<code>*2</code>''<nowiki><channel-reserve></nowiki>''
*<code>*3</code>''<nowiki><prefix></nowiki>''
*<code>*4</code>''<nowiki><match></nowiki>''
*<code>*5</code>''<nowiki><id-length></nowiki>''
*<code>*81</code>
*<code>*82</code>
*<code>*83</code>

''<nowiki><channel-max></nowiki>'' is used to set the maximum number of conference channels which shall be allowed in this interface at a time. The limit can be used for example to make sure the CONF interface does not consume all of the DSP channels on a gateway which is used as a trunk line interface too. NB: as of V9 release, you must specify the <code>*1</code>''<nowiki><channel-max></nowiki>'' option correctly. If you omit it, the CONF interface will accept more <code>*2</code>''<nowiki><channel-reserve></nowiki>'' reservations than it is capable to satisfy.

''<nowiki><channel-reserve></nowiki>'' is used to indicate the minimum number of conference channels which shall be available in the new conference room. Only 2 digits may follow <code>*2</code> so that a maximum of 99 channels is possible. If the <code>*2</code> option is used and 2 digits follow, the whole command is considered finished and the trailing <code>#</code> may thus be left out. Please note that due to this, the <code>*2</code> option should be the last option used.

''<nowiki><prefix></nowiki>'' is used as prefix for the conference id created. If more than one CONF interface is used, this prefix can be used to enforce unique conference ids across all interfaces. The prefix may be empty. It may be specified as value for the <code>*3</code> option, or - as a shorthand notation - directly following the initial <code>*1</code> command introducer.

''<nowiki><match></nowiki>'' is a digit string which is used to verify the room number (specified as ''prefix'') requested. If the requested room number does not begin with the specified ''match'', the CONF interface will reject the call with cause code ''no channel available''. This can be used to route calls for fixed room number to the appropriate CONF interface.

''<nowiki><id-length></nowiki>'' specifies the length of the random part of the created room number. It defaults to 6 if not specified.

<code>*81</code> is the ''disconnect control'' option. If the disconnect control option is enabled and the caller of the conference disconnects the conference call, all other calls in this conference room are automatically disconnected.

<code>*82</code> is the ''remote control connect'' option. If an user with the remote control connect option creates a conference room, the conference sends an alerting signal to incoming calls to this conference room instead of a connect signal first and waits for a remote control connect facility for each call before a call is connected.

<code>*83</code> is the ''multi-video off'' option. If a room is created with this option, no multi-video is offered to all calls to this room.

The new room is only created if the required channels can be provided and the maximum number of channels used by the interface is not exceeded. If the room can be created, it is joined, too. The room number (conference id) is returned as the connected number, including both the prefix and the random part. If the conference cannot be created, the call is disconnected with a ''No channel available'' cause code.

Example: <code>*110*124*26#</code> (or <code>*1*310*124*26#</code>) requests the creation of a new conference room with <code>10</code> as conference id prefix (room number). 6 channels are reserved for the conference (so there can be at least 6 participants) and the maximum amount of channels used by the CONF interface is set to 24.

Please note that the CONF interface does not store the <code>*1</code> channel limitation. So you need to make sure it is provided consistently on all calls to the CONF interface that create new conference rooms. Also, both limits (<code>*1</code> and <code>*2</code>) control the resources used by the interface itself. They do not ensure that the resources are not consumed by others when they are actually needed (for example, a physical interface such as ISDN may have used all available DSP channels for VoIP calls).

===== Create a new room with a given room number: *2 =====

This command is the same as above, except that no random number is appended to the ''<nowiki><prefix></nowiki>''. It is not possible to create the conference room if its id conflicts with a room that currently exists in this interface.

Example: <code>*210*124*26#</code> (or <code>*2*310*124*26#</code>) requests the creation of a new conference room with <code>10</code> as conference id (room number). 6 channels are reserved for the conference (so there can be at least 6 participants) and the maximum amount of channels used by the CONF interface is set to 24.

===== Create a new or join an existing room with a given room number: *3 =====

This command is the same as above, except that if there is no room with the given room number a new room is created. Otherwise the existing room is joined. No random number is appended, too.

Example: <code>*310*124*26#</code> (or <code>*3*310*124*26#</code>) requests the join to a conference room with <code>10</code> as conference id (room number) or - if the room does not exist yet - creation of a new one. 6 channels are reserved for the conference (so there can be at least 6 participants) and the maximum amount of channels used by the CONF interface is set to 24.

===== Join an existing room: 0-9 =====

If the called party number does not start with an asterisk <code>*</code>, the remaining digits are interpreted as conference room id and the call will join this conference if it exists.

The dialed digits are used to find an existing room. The first room number which matches is joined (for example, if the called party number is <code>10</code> and there is a conference room with id <code>1</code>). No sending complete or end marker is necessary if the room is found or if there is no such room. However, if the called party number matches only the head of an existing conference room id, the interface will wait for additional digits to decide unless a <code>#</code> is seen or ''sending complete'' is on. You may want to enable ''Force enblock'' in routes towards the CONF interface thus.

For commands which join a room, the call is rejected with a ''No channel available'' cause code if the room does not exist.

The old V8 behavior to create a new conference room instead of joining is only supported with the '*3' command. Now load balancing with multiple conference devices is possible.

==== In-Call commands ====

These commands can be used during calls and must be sent with DTMF tones.

===== Exclusive listen mode: *21# =====

All members except for the caller are muted.

===== Normal listen mode: *6# =====

All members are connected.

==== Innovaphone remote controls ====

An [[Reference:Remote Control Facility | innovaphone remote control facility ]] can be sent e.g. by a SOAP application. These remote controls are available:
* 0: ''Connect''
* 24: ''Receive on''
* 25: ''Receive off''
* 26: ''Exclusive listen mode''
* 27: ''Normal listen mode''

===== Connect (0) =====

Connects a conference call in alerting state made with the call-setup command option ''remote control connect'' (*82).

===== Receive on (24) =====

Switches on receiving incoming voice data of this conference member. This means all other conference members can listen to this conference member. It overwrites a before received ''exclusive listen mode''.

===== Receive off (25) =====

Switches off receiving incoming voice data of this conference member. The conference member is muted for all other conference members.

===== Exclusive listen mode (26) =====

All members except for this conference member are muted. It overwrites a before received ''receive on'' from other conference member calls.

===== Normal listen mode (27) =====

All members are connected. It overwrites a before received ''receive off'' from other conference member calls.

==== Call events ====

These events are recognized if the interworking option for the conference route is activated:

===== Hold event =====

If the call is hold by the member, the conference call is muted.

===== Retrieve event =====

If the call is retrieved again by the member, the conference call is activated.

===== Websocket Registration section=====
Registration to a websocket endpoint - in case of the CONF/SCNF Interface this will be the Registration to the Conference Scaler App (Available with 16r1)

URL: The address of the websocket endpoint

Username: Name used for the registration. If no name and no number is configured a name of the form <product-short-name>-<last 3 Bytes of Mac Address>-<Interface Id> (e.g. IP800-08-03-01-TEL2) is used instead.

Password: Password used for authentication if needed.

=== Conferencing interface (SCNF) ===

Same functionality as CONF, except only G.711 coders are supported.

===== Websocket Registration section=====
Registration to a websocket endpoint - in case of the CONF/SCNF Interface this will be the Registration to the Conference Scaler App (Available with 16r1)

URL: The address of the websocket endpoint

Username: Name used for the registration. If no name and no number is configured a name of the form <product-short-name>-<last 3 Bytes of Mac Address>-<Interface Id> (e.g. IP800-08-03-01-TEL2) is used instead.

Password: Password used for authentication if needed.

=== FAX interface ===

The FAX interface can be used to send or receive fax documents. Each call to this interface must be controlled with user-user-information messages by the [[{{NAMESPACE}}:Concept_App_Service_Fax|App Service Fax]].

==== Headline ====
There are some substitutions in the header line of outgoing faxes. The header line has a length of 108 signs.

* $d: day of month, two digits
* $j: day of month without leading zero
* $m: month, two digits
* $n: month without leading zero
* $Y: year, four digits
* $y: year, two digits
* $G: hour, 24 hours format, without leading zero
* $H: hour, 24 hours format, two digits
* $i: minutes, two digits
* $s: seconds, two digits
* $p: number of page
* $P: page count, if available, with leading '/'
* $I: subscriber id
* $R: remote subscriber id
* $$: the sign '$'
* $t: tabulator to the next 15 character grid
* $c: text is centered
Flags for the place holders:
* 0-9: field length, with tabulator jump forward to this position
* -: left aligned

The default header line for English is defined as:

$d/$m/$Y $G:$i <subscriber-name> $I$63t To:$-20R$tPage: $p$P$c

== CGPN/CDPN Mappings ==
{{CGPN-CDPN_Mappings}}

Reference16r1:Apps/PbxManager/App Conference Scaler

2026-03-09T12:07:10Z

Nwe: /* Add an App */

This Settings Plugin contains 1 section:

* Add, edit, or delete Conference Scaler app objects on the PBX and configure devices for use by the Conference Scaler app instance.

== Add an App ==
; Name
: The name displayed for the App Object.

; SIP
: The sip from the App Object, which must be unique.

; Conference interface
: Here you can select the conference interfaces that can be used with the Conference Scaler app instance. These will be configured accordingly. The Devices API must be available for this to work (only Interfaces which are available at the Devices App will be shown here).

Howto13r3:Step-by-Step Standby registration through Reverse Proxy

2025-10-02T11:28:22Z

Nwe:

[[Category:Step-by-Step|Standby registration through Reverse Proxy]]

{{HOWTOMOD13r3 Translation Info}}

This article explains how to set up an H323/TLS standby registration through the Reverse Proxy. If the master PBX fails, phones should be able to register with the standby PBX.

We already discussed how to set up a standby PBX in another [[:Howto13r3:Step-by-Step_Setting_up_a_Standby_PBX| step-by-step guide.]] So if your knowledge is a bit fuzzy, please take a look at this article. Also, this article builds on our IT Plus topic [[Courseware:IT_Plus_-_Public_Access_to_Conference_Resources#Registering_a_phone_through_the_reverse_proxy| Public_Access_to_Conference_Resources]], so if you don't know how to set up a reverse proxy or how to register a phone through a reverse proxy, please use your IT Plus subscription and do the topic on class.innovaphone.com.

[[image:h323-through-RP-overview.png]]

== Purpose ==
A standby PBX provides redundancy in your system in case your master PBX fails. This should be the case if the standby PBX is cloud-based or behind a reverse proxy. Phones should be able to register with the standby PBX if the master PBX fails without any manual action.

=== Features ===
Phones should remain usable even if the master PBX fails.

=== Limitations ===
* Internet/connection to Reverse Proxy is mandatory
* This feature is only implemented in H323

== Requirements ==
* To enable standby functionality, a standby PBX must have the same amount of standby licenses as port licenses. Please note that the standby PBX receives the necessary licenses as soon as it registers with the master PBX by replication. You don't need to upload the standby licenses manually to the standby PBX! In case you use an IPVA as a Standby PBX, make sure that you also have IPVA-licenses available. For more information on licensing, please check our [https://www.innovaphone.com/en/services/licenses.html license guide]
*The platform used as a standby PBX needs to be able to handle as many registrations and object definition as the master PBX does. However, it does not need to be the same type of device. E.g. an IP0011 may serve as a standby for an IP3011 in certain scenarios.

== Things to know before you begin ==
* DNS name of the reverse proxy (or reverse Proxies) has to be known. Master and standby PBX may be behind the same or different reverse proxies.
* Master PBX and Standby PBX must have the same PBX name, but must have different DNS names.
* Port 1300 is used for H.323/TLS, make sure the reverse proxy is reachable on this port from the internet
* The system name of the PBX

== Configuration ==
=== Standby PBX ===
Create a standby PBX as it is explained in our [[:Howto13r3:Step-by-Step_Setting_up_a_Standby_PBX| step-by-step guide.]]

=== Reverse Proxy ===
*The reverse proxy has to listen to Port 1300 for H.323/TLS requests.
*You need to create a hostname entry like ''innovaphone.com/hq'' which forwards H.323 traffic to the master PBX (destination port 1720 and TLS port 1300)
*You need to create a hostname entry like ''innovaphone.com/hq-standby'' which forwards H.323 traffic to the standby PBX (destination port 1720 and TLS port 1300)

[[image:h323-through-RP-1.png]]

=== Phone ===
*Configure a primary and secondary gatekeeper on the phone.
*Configure a Gatekeeper identifier like innovaphone.com/hq''':'''hq-standby.

[[image:h323-through-RP-2.png]]

The ''':''' is crucial because it separates the gatekeeper identifiers used in the regular and standby cases. In normal operation, the phone uses the primary gatekeeper address and the gatekeeper identifier ''innovaphone.com/hq'' to register with the master PBX. The reverse proxy analyzes the gatekeeper identifier and sends the registration request to the master PBX.
If the phone cannot register with the master PBX (e.g. power outage), it uses the secondary gatekeeper and the gatekeeper identifier ''innovaphone.com/hq-standby''. The reverse proxy will forward this registration request to the standby PBX based on your hostname entry.

== Verification ==
Unplug the master PBX, after a timeout the phone will change its gatekeeper identifier and attempt to register with the standby PBX.

== Related Articles ==
[[{{NAMESPACE}}:Step-by-Step_Setting_up_a_Standby_PBX]]

[[Courseware:IT_Plus_-_Public_Access_to_Conference_Resources#Registering_a_phone_through_the_reverse_proxy|Course13:IT_Plus_-_Public_Access_to_Conference_Resources]]

[[Courseware:IT_Advanced_-_06_Public_Access_to_PBX_Resources_(theory)_-_optional]]

Howto13r2:Step-by-Step Simple DECT Installation

2025-10-02T11:28:12Z

Nwe:

[[Category:Step-by-Step|Simple DECT Installation]]

{{HOWTOMOD13r3 Translation Info}}

In this article we will explain how to set up a DECT system without any theoretical background. In this step-by-step tutorial we will use only one IP DECT master and one radio. Since both modules can be operated on the same IP1202/IP1203, you only need one base station. If you need more than one base station for your customer setup, you will see how easy it is to add as many base stations as you want.

[[Image:Simple-DECT-overview.png]]

== Purpose ==

DECT is used to transmit voice over the air. This means you can move freely around the house or office while talking on the phone without the need of a cable. In this guide we will explain all steps necessary to link your IP1202/IP1203, which are models of our DECT base station, to your PBX. This article does not go into full detail. If you want to know more, you are welcome to do our Advanced training.

=== Features ===
*cordless communication

=== Limitations ===
* This guide only explains how to set up DECT on one location.
* This article does not cover roaming.
* If calls get dropped due to a lack of coverage, you need to buy (an) extra base station(s) to increase the coverage area.

== Requirements ==
* innovaphone PBX
* IP1202 or IP1203 (All steps in this guide are the same for both models)
* 1 x SARI (50-00060-019)
* 1 x DECT handset like an IP64 or IP65

== Things to know before you begin ==
*You need to know the PBX Password, which was created by the ''Install'' during the initial configuration of your PBX.

== Configuration ==
=== Create a DECT System object ===
The first step is to create a DECT System object in the PBX. This is done on the Advanced UI -> PBX/Objects. Select a '' DECT System'' object in the drop down menu and click on ''new''.

[[Image:Simple-DECT-sys-object.png]]

Give this object a ''Long name''. Please note that this name will be displayed on all DECT handsets. As a result, please select the name carefully.

[[Image:Simple-DECT-sys-object-2.png]]

=== User configuration ===

Each user must have a hardware ID that corresponds to his name.

[[Image:Simple-DECT-user-config.png]]

On the DECT tab of the user object, you must configure the following options:
* You must set the ''Gateway'' option (screenshot below) to the same name as the name of the DECT system object you created in the previous step.
* If you already know the ''IPEI'' of the DECT handset, you can enter it here. This is not mandatory, but you can skip a step later if you already enter the IPEI here.
* Please define a user ''AC''. This user AC will be used later for the authentication of the DECT handset at the DECT system.

[[Image:Simple-DECT-user-config-2.png]]

=== LDAP Server ===
As the next step we will set up the LDAP replication between PBX and IP-DECT Master so that all necessary objects can be replicated to the IP-DECT Master. To do this, we need to create new LDAP credentials on the LDAP server of our PBX. This requires you to set up a username and password on Services/LDAP/Server. As a rule of thumb, configure a username that consists of a domain and a user part. Once you have completed the IT Advanced training, the reason for this will be obvious to you.
* Please configure a username like ''host.domain.net\dect''
* Feel free to configure a password of your choice
* Activate the checkbox ''Write access''.

[[Image:Simple-DECT-LDAP-server.png]]

=== DECT System ===
Before we begin, please reset the device to factory settings by pressing the reset button until the LED turns orange. As a next step enter the IP address of your IP1202/IP1203 into your browser. The Install page is displayed. We recommend to ''add the gateway'' to your ''Devices'' app as you learned in [[Courseware:IT_Connect_-_06.0_Managing_Devices#Adding_a_Gateway|your IT Connect training.]]

[[Image:Simple-DECT-Install.png]]

Afterwards, please set the ''Mode'' of your IP1202/IP1203 to ''Active (Master)''. You can find this option on DECT/Config/Master. You have to reset the device afterwards.

[[Image:Simple-DECT-mode.png]]

As a next step, please go to DECT/Config/System and configure the following options:
* System Name: The system name must match the name of the DECT system object that we created in the first step.
* Password: This password has to match the PBX password.
* Subscriptions: Due to security reasons please select ''With activation''

Press ok, afterwards a reset is required.

[[Image:Simple-DECT-system.png]]

=== DECT IP-DECT master ===
As next step we will configure the IP-DECT Master. Please go to DECT/Config/Master
* Enable PARI function: Please select this checkmark
* Protocol: Please select H323/TLS for security reasons
* Gatekeeper: Please enter the IP address or DNS name of your PBX

Afterwards a reset is required again.

[[Image:Simple-DECT-master-config.png]]

After the device is restarted, an authentication code is suggested under DECT/Config/System. As long as you do not press Ok again, the suggested authentication code will change every time you refresh the page. You need to write an authentication code in the configuration, so either just press Ok to accept the suggested code or choose an authentication code according to your liking.

Furthermore you will be able to enter a SARI now. Please go to DECT/Config/SARI and copy the SARI you have purchased. You only need one SARI, even though you have multiple base stations.

=== LDAP client ===
In the next step we will complete the LDAP replication. To do this, please go to Services/LDAP/Replicator on your IP1202/IP1203. Set the following options, all other options should remain as they are:
* Enable: Please select this option
* Server: Please enter the IP or DNS Name of your PBX.
* TLS: Please select the TLS checkmark
* DECT Gateway Name: Enter the name of the DECT system object we created in the first step (Create a DECT System object)
* User: Please enter the username of the LDAP Server. This name should be configured like ''host.domain.net\dect''
* Password: The password must match the password we configured when we were setting up the LDAP server

[[Image:Simple-DECT-replicator.png]]

=== Radio ===
To add a radio to your DECT system, please go to DECT/Devices Overview/Radios. A list of all uninitialized radios in your network will be displayed. You should see at least one radio that belongs to the same device you are configuring. Just click on ''Add'' and then on ''OK''. If you have more than one DECT base station in your network, you just add them the same way. If you need more than a few base stations, refer to [[Courseware:IT_Plus_-_IP-DECT|our IP-DECT book]] and please attend our IT Advanced Training.

[[Image:Simple-DECT-Radios.png]]

After a few seconds you will see the radio again, but this time it has a static registration, but is not yet synchronized.

[[Image:Simple-DECT-Radios-2.png]]

=== Air Sync ===
To establish a synchronization between all base stations, one of the base stations must act as Air Sync Master. To do this, please go to DECT/Config/Air Sync and set the ''Sync mode'' to Master.

[[Image:Simple-DECT-air-sync.png]]

After that, the LED lights up constantly blue.

=== Subscription ===

When you put your DECT handset into operation for the first time, you will be asked for the subscription parameters. Please set the following options. If your device does not ask you for the subscription parameters, you can navigate to Menu/Connections/System/Subscribe.

* System name: Please enter the system name here. The system name is the name of the DECT system object created in the first step. (Create a DECT System object)
* PARK: Please enter the PARK here. You will find the PARK on your IP-DECT Master DECT/User
* AC: Please enter the User AC here if you entered the IPEI already in the PBX. If this not the case, you have to enter the System authentication here. You can find the system authentication code under DECT/Config/System/Authentication code
* Protection: Set to ''No''.

After you entered all the required options, your subscription will still fail. The reason is that we selected the Subscription mode ''With Activation'' while we configured the DECT system (DECT/Config/System). Although it would have made life much easier if we had selected the subscription mode ''Allowed'', ''Allowed'' should not be used in productive systems as it is considered a security risk.

If you want to subscribe a handset when the subscription mode ''With Activation'' is active, you have to click on the IPEI before you start the subscription. You will find the IPEI under IP-DECT/User if you have entered the IPEI on the PBX user object in the previous step.

[[Image:Simple-DECT-dect-users-ipei.png]]

If the IPEI is not known, you must enter the IPEI of the handset as Anonymous handset under DECT/Anonymous. If you try to subscribe the handset, you must also press the IPEI in this case.

[[Image:Simple-DECT-dect-anonymous.png]]

=== Registration ===
If the IPEI is already known in your IP DECT master, the IP DECT master will automatically register the handset with the PBX. If this is not the case, you must assign the handset to a user object. This is done by calling this number:
*<Master-ID>*<User-Number>*<Access-Code>#

* Master ID: Since we did not change the master ID, the master ID is 0.
* User Number: The user number of the user to whom you want to assign the DECT phone.
* Access-Code: You must configure the user AC here. You can find the user AC on the DECT tab of the user object in the PBX.

An example for a number to be dialed would be this: *0*13*1234#

This number consists of 0 for the master ID, 13 for the user number and 1234 for the user access code.

After successful registration, you will see the DECT system name and the user name on the handset display.

== Verification ==
* You can call any number now.

== Known issues ==
== Related Articles ==
*[[Courseware:IT_Plus_-_IP-DECT]]
*[[Reference:IP1202 DECT System]]

Howto13r3:Step-by-Step Shared Fax App ressources in the Cloud

2025-10-02T11:27:59Z

Nwe:

[[Category:Step-by-Step|Shared Fax App ressources in the Cloud]]
[[Category:myApps_Cloud]]

{{HOWTOMOD13r3 Translation Info}}

== Purpose ==
This article is useful for Cloud providers. An IPVA doesn't have DSPs, because of this you may want your Cloud Customers to use a physical device for coding/decoding Fax.
In this article I will show you how you can configure a PBX and share external Fax resources for your Cloud Devices.

=== Features ===
* Easy way to share FAX resources from a physical Device
* Cloud Customers will have access to Audiofax (i.e. G.711A) Fax

== Requirements ==
* innovaphone PBX (cloud based IPVA)
* innovaphone Fax gateway with Audio Fax ability (see column "Audio Fax Ch" in [[Howto:How_to_implement_large_PBXs#Technical_data_and_recommended_number_of_users_supported|this table]] for feasible devices)
* The FAX Interface registration requires one Port and one FAX License on the FAX Gateway.
* Every PBX registration requires one Port License on the FAX Gateway.
* The FAX Object on the Customer PBX requires one Port License.

[[File:Fax_setup.jpg]]

=Pre-Configuration of the FAX Gateway=

==PBX Configuration==
*Go to PBX/Config and Choose "Master" as PBX Mode
*Enter the following parameters: System Name, PBX Name, Reverse Proxy Addresses
*Important: Enter "For Fax" in "Route Root-Node External Calls to", we will create this Object later, you can also name it something else but it's important that the Long Name matches

==Setting up the FAX Registration and the Fax Interface==
*First, go to Gateway/Interfaces
*Now open the FAX Interface and enter following parameters: Name, Protocol: H.323/TLS, Gatekeeper Address: 127.0.0.1
*Tick following checkboxes below: Enable T.38, Audio FAX Support
*Now, register the FAX Interface to a Gateway Object
**Head to the Section PBX/Objects
**Create a new "Gateway" Object
**Enter the following parameters: Long Name (this name is important for the step above in "PBX Configuration"), Display Name, Name, Password (the matching Password for the FAX Interface)
**As Hardware ID you enter your FAX Gateway "Mac Address-Interface", e.g. "12ABC7426CDD-FAX"
** Make sure to tick "Fax License" in the "Gateway" section
[[Image:for_fax.png|For Fax GW Object]]
[[Image:for_fax_license.png|For Fax GW Object]]

=Configuration of the Cloud customers PBX=

==Configure the first Gateway Interface==

*1. First, open the first Gateway interface on Gateway/GK/GW1
*2. Enter the following parameters (Name: "4 Fax"; Protocol: "H323/TLS"; Mode: "Register as Gateway"; Adress "DNS name of the Cloud PBX"; Gatekeeper Identifier: "######.com")
*3. Tick following parameters (Enable T.38: checked; Audio FAX Support: checked)
[[Image:4fax.png]]

==Creating the FAX Object and registering it to your recently configured Gateway Interface==
*Open PBX/Objects and create a new "Fax Object"
*Enter the following parameters (Long Name: "Fax"; Name: "Fax"; Hide from LDAP "checked"; Critical "checked")
*Also edit the Hardware ID of the Fax object to following format: "IPVAMAC-Interface", for example: "0A4C8BF9DD21-GW1"
**"TLS Only" and "Reverse Proxy" must be checked
*Enter the right innovaphone FAX URL in section "Fax"
[[Image:fax_object.png]]

==Configuration and registering of the second Gateway Interface==
*First, open the second Gateway interface on Gateway/GK/GW2
*Enter the following parameters (Name: "to Fax GW"; Protocol: H323/TLS; Mode: Register as Gateway; Address: "Your Gateway address"; GK-Identifier: "Your Gatekeeper address";)
*Tick the following parameters: (Enable T.38: checked; Audio FAX Support: checked)
[[Image:GW1_int.png|GW1 Interface]]

==Configuration of two routes==
===First route===
*At first, open the route Tables under /Gateway/Routes/
*Create a new Route by pressing the Arrow next to the "From" Table header
[[Image:routetable.png]]
*For the first route from GW1->GW2 you will need a prefix in the following format: "*" + "INSTANCENUMBER" e.g. "*XXXXX". (Without quotes), you can also use the Customernumber but this Number must be identical to the "name" of the Fax Gateway User (which we will create later)
*Tick the "GW1 4Fax" check
*Select "GW2 to FAX GW" at the Dropdown menu on the right
[[Image:fax_route1.png]]

===Second route===
*Create a new Route by pressing the Arrow next to the "From" Table header
*For the second route tick the Checkbox "GW2 to FAX GW" and in the same Dropdown menu as before select "GW1 4Fax".
[[Image:fax_route2.png]]
*The routes should then look like this:
[[Image:fax_route.png]]
 

==Adjustment of the SIP-Trunk==
*First, open the Customer PBX
*Open the right SIP-Trunk at Gateway/SIP
*Check if the "Exclusive" check is ticked, If not then tick this check.
[[Image:fax_sip.png]]
 

=Configuration on your Fax Gateway for all Customers=
*Open PBX/Objects
*Create a new "User" Object
*Edit following parameters:
**Description: "Customer Domain"
**Long Name: Instancenumber/customernumber
**Display Name: Instancenumber/customernumber
**Name: Instancenumber/customernumber
**Password: Choose a secure Password
**Hardware-ID: "IPVAMAC-GW2" e.g. "ABCDEF012345-GW2"
**Reverse-Proxy: checked
*If you have configured it successfully, your Users should be shown in the "Registration" page at PBX/Registrations
[[Image:successful_registration.png]]
*Repeat this step with every new Cloud customer that you want to share FAX resources with.

== Verification ==

===Sending a FAX to another User through the FAX App===

Open the FAX App and send a Fax. Address your FAX App Number + extension of User e.g. your FAX App has the Number "70" and your User has the Extension "4" -> then send a Fax to the Number "704"

Howto14r2:Step-by-Step Shared Conference ressources in the Cloud

2025-10-02T11:27:48Z

Nwe:

=Description=

[[Category:Step-by-Step|Shared Conference ressources in the Cloud]]
[[Category:myApps_Cloud]]

{{HOWTOMOD13r3 Translation Info}}

This document shows how to set up an innovaphone PBX as External Conferencing Resource and explains the necessary configuration steps.

==Description==
An innovaphone PBX can act as External Conferencing Resource, especially useful when used in conjunction with IPVA cloud instances. We call it '''Conferencing Gateway''' from this onward.

[[Image:conf_setup.jpg]]

==Requirements==
* innovaphone PBX
* The Conf Interface registration requires one Port License
* Every PBX registration requires one Port License on the PBX
* App(innovaphone-pbx-conferencing) License

=Generic Pre-Configuration=

The following configuration steps are done on the Conferencing Gateway.

===Setting up the Conferencing Gateway===

Go to ''PBX/Config'' and choose ''Master'' as PBX Mode and enter the following parameters:
*System Name / PBX Name / Reverse Proxy Addresses: Set accordingly to your setup
*"Route Root-Node External Calls to": In this example, we will enter "Conf". You can choose the name as you wish but note we will need it later

[[File:Pbxconf.jpg]]

===Setting up the Conferencing Interface===
Go to ''Gateway/Interfaces'' and click on the CONF Interface and enter the following parameters:
*Name: ''Conf''
*Protocol: ''H.323/TLS''
*Gatekeeper Address: ''127.0.0.1''

[[Image:conf_int2.png]]

===Registering the Interface===

Now we will register the Conferencing Interface to a PBX Object.
*Head to the Section ''PBX/Objects''
*Create a new Gateway Object
*Enter the following parameters:
**Long Name: ''Conf'' (this name is important for the step above in "PBX Configuration")
**Display Name: ''Conf''
**Name: ''conf''
**As ''Hardware ID'' you enter your Conf Gateways "MAC Address-Interface", e.g. ''12ABC7426CDD-CONF''

[[Image:conf_gateway.png]]

=Customer-Specific Configuration=

Now we will configure the options which are specific to the customer. Steps in the first section are done on the Conferencing Gateway while the steps in the second section are done on the customers PBX/IPVA. In this article we will use "kunde1" as reference.

===Configuration of the Conferencing PBX===

*Go to ''PBX/Objects'' and create a new Gateway Object
*Enter the following parameters:
**Long Name: ''Kunde1''
**Display Name: ''Kunde1''
**Name: ''kunde1''
**Password
*Edit following parameters in the Devices section:
**Hardware ID: ''kunde1''
**Name: ''kunde1''
**Config VOIP: ''Enable''

[[Image:Kunde1.png]]

After pressing on ''Apply'' you can see a new section has appeared on the top.
*Click on that section and choose ''H.323/TLS'' in the drop-down menu
*Enter the following parameters:
**Customers Gatekeeper ID
**Customers Gatekeeper Address

[[Image:config_voip.png]]

===Configuration of the Customers PBX/IPVA===

We will now create the Conference Object on the customers PBX. In this article we will use "Conference" and a free number "80" as reference.
*Open PBX/Objects and add a new Conference Object
*Enter the following parameters:
**Long Name: ''Conference''
**Display Name: ''Conference''
**Name: ''conference''
**Number: ''80''
**Hardware ID of the Conf object: Here you will enter the name of the Gateway that we just created for the Customer. ''kunde1'' in this instance
**Reverse Proxy: ''Enable'' if the Customers PBX is accessed through an RP

[[File:Conf conf.png]]
 

*Head to the section Devices
*Enter the Hardware ID
*If the registration was successful you should see an IP-address to the right

[[File:Conf devices.png]]

==Verification==

After creating the Conference Object, you will notice that two routes have been configured automatically on the customers PBX.

[[File:Confroutes.png]]

Create a static room and verify if it works.
*Open your Conference Object and move to the Section ''Options''
*Change Room Number length to ''2''
*A new Section will appear called ''Rooms'', move to that Section
*Create a new Room
*Call that Room and see if you successfully entered the Room and everything works fine.

Howto13r3:Step-by-Step Setting up a Standby PBX

2025-10-02T11:27:32Z

Nwe:

[[Category:Step-by-Step|Setting up a Standby PBX]]
{{HOWTOMOD13r3 Translation Info}}

This guide provides step-by-step instructions for configuring a standby PBX only, which is used to enhance the availability of your system.

This guide does not describe all aspects that might be necessary in your scenario, it only covers the setup of the standby PBX. Check for more details the related articles the courseware about redundancy.

== Purpose ==
A standby PBX provides redundancy in your system architecture in case the master PBX fails or the connection to the master PBX got lost.

In the standby role, registrations arriving at the standby PBX will be diverted to the master PBX.

When there is no connection between standby PBX and master PBX or if the master PBX fails, the standby PBX will switch its role from "Standby" to "Master" and accepts registrations from the clients.

As soon as the master PBX is up and running again or the connection is restored, the master PBX takes over it's master role again and the standby will be in the standby role again.

There is a timeout mechanism to make sure that this switching roles only happens after the timeout has been expired, this to prevent repeatedly switching in case of short connection errors.

[[image:set-up-standby-PBX-overview.png|set-up-standby-pbx-overview.png/]]

=== Features ===
* A standby PBX ensures the availability of your system in case the master PBX in your system architecture fails or the connection to the master PBX fails.

== Requirements ==
* The platform used as a standby PBX needs to be able to handle as much registrations and object definition as the the master PBX does. However, it does not need to be the same type of device. E.g. an IP0011 may serve as a standby for an IP3011 in certain scenarios.
* To enable standby functionality, a standby PBX must have the same amount of standby licenses as port licenses. Please note that the standby PBX receives the necessary licenses as soon as it registers with the master PBX by replication. You don't need to upload the standby licenses manually to the standby PBX! In case you use an IPVA as a Standby PBX, make sure that you also have IVA-licenses available.

== Things to know before you begin ==
* You need to know the PBX password of your system
* You need to know the IP address of your master PBX

== Configuration ==
Please enter the IP address or DNS name of your designated standby PBX into a browser and log in to the advanced user interface of this device.
=== PBX Mode ===
Go to PBX/config/General and set the PBX mode to ''Standby''

[[image:set-up-standby-PBX-1.png|set-up-standby-pbx-1.png/]]

=== General Settings ===
Almost all properties of the general page of the standby PBX must be set identically to those of the master PBX. Please note that the standby PBX must have a unique ''DNS name''.

[[image:set-up-standby-PBX-2.png|set-up-standby-pbx-2.png/]]

Do not forget to press OK to save the settings.

=== PBX Password ===
Go to PBX/Config/Security and set the PBX password identical to the ''PBX password'' of the master PBX.

[[image:set-up-standby-PBX-3.png|set-up-standby-pbx-3.png/]]

=== Registration ===
Please go back to PBX/Config/General and set the ''registration'' protocol to H.323/TLS. Furthermore please set the ''Master'' property to the IP address of the master PBX. (No DNS Name allowed)

=== Object Replication ===
After registration we want to replicate all objects to the standby PBX. This can be achieved by ticking the ''Replicate from Master'' checkbox and the ''use TLS'' checkbox.

[[image:set-up-standby-PBX-4.png|set-up-standby-pbx-4.png/]]

=== Client configuration ===
Make sure each phone or interface is configured with a secondary gatekeeper, as this allows the client to send its registration requests to the standby PBX if the master fails.

== Verification ==
There are multiple ways to verify if your configuration works as expected.

*You should see an entry called ''_STANDBY_'' in the list of registrations on the master PBX.

[[image:set-up-standby-PBX-5.png|set-up-standby-pbx-5.png/]]

*You should see an entry called ''_ACTIVE_'' in the list of registrations on the standby PBX.

[[image:set-up-standby-PBX-6.png|set-up-standby-pbx-6.png/]]

*You should see all expected licenses in the Licenses area (PBX/Config/General) on your standby PBX.

*All Objects of the master PBX should be replicated to the standby PBX.

*You can test your configuration by turning off the master PBX. The standby PBX should take over all registrations

== Known issues ==

== Related Articles ==
[[Courseware:IT_Advanced_-_02_PBX_-_initial_Configuration]]

[[Courseware:Advanced_-_Standby_PBX]]

[[{{NAMESPACE}}:Step-by-Step Standby registration through Reverse Proxy]]

Howto13r3:Step-by-step Set up a group pickup

2025-10-02T11:27:21Z

Nwe:

[[Category:Step-by-Step|Set up a group pickup]]

{{HOWTOMOD13r3 Translation Info}}

This article explains how to create a group pickup function key without group indications. Group indications should be avoided whenever possible because they increase load on the PBX. You should rather use dialog infos.

[[Image:Group-pickup-overview.png]]

== Purpose ==
A user can monitor all members of a group with a function key. This allows the user to pick a call destined for any user in the group. This feature is not restricted to one PBX. The user can monitor a group on another PBX.

=== Features ===
* Alert multiple users at once
* This feature allows users to be alerted across PBXs. Although it is a different group on the other PBX, if it has the same name, communication is possible.
* Less CPU load on the PBX due to the use of dialog infos instead of group indications

=== Limitations ===
* Groups are limited to 2000 members. Groups only exist on a single PBX. If another PBX has a group with the same name, it is a different group.

== Requirements ==
* Innovaphone PBX
* Hardware phones such as an IP111, IP112, IP222 or IP232 or softphone

== Things to know before you begin ==
* You can only pick calls from user objects. A call cannot be picked from a waiting queue or call broadcast object
* The visibility rights ''calls'' and ''calls with number'' are crucial. Either has to be allowed to pick a call from a user

== Configuration ==
In this scenario we will create a group called ''pickup'' and we will create a function key of type partner on an IP phone so that the user can pick a call from any member of the group.
To use the pick up function on the softphone/phone app, you can skip the first two steps. Only visibility rights are required.

To make it simple as simple as possible we will use the names of the User we used during the IT Connect Training.

=== Create a group ===
We will create a group on the PBX. A group is created by the ''Groups'' PBX Manager plugin.
* Just enter the name you want, e.g. ''pickup''
* Search for each member of the group and then click on the name.
* The option ''A'' means active. Only active members of the group can pick the call.

[[Image:Group-pickup-1.png]]

=== Create a function key ===
The easiest way to create a function key for a group of users is a config template in the PBX Manager.
* Expand the ''Function keys'' section
* Select the placement for the function key on the phone display
* Select Partner as type of the function key
* Enter a name to be displayed, e.g Group pickup
* Enter the name of the group prepended by a dot (.), e.g .pickup

[[Image:Group-pickup-2.png]]

=== Visibility rights ===
To pick a call, visibility rights must be granted. ''Calls'' or ''Calls with Number'' must be allowed from the user receiving the call.
You can enable either of these options in the config template( Visibility column in the Advanced UI -> PBX/Objects).

[[Image:Group-pickup-3.png]]

It's important to understand that a group membership is not enough to enable pickup. Visibility rights are mandatory. The ''Group Default Visibility'' (Advanced UI -> PBX/General) can be used to add visibility rights to an active group membership. By default, ''Calls'' or ''Calls with Number'' is enabled for the ''Group Default Visibility'', which makes your life easier because by default an active group membership would be enough for pickup. If you would disable both options, the pickup will not work anymore.
[[Image:Group-pickup-4.png]]

=== Pickup from softphone ===
As long as the visibility right is set, all you need to do, is add the user you want to pick from, as a favorite.

[[Image:Group-pickup-5.png]]

== Verification ==
Call a monitored user (e.g. Lisa Svensson).

*You will receive a notification to pick the call in the sotphone
[[Image:Group-pickup-6.png]]

*Your group pickup function key is highlighted with the option that a call can be picked.

== Known issues ==
*There is no audible signal in myApps/softphone to indicate that a call is ready to be picked.

== Related articles ==
[[Courseware:IT_Advanced_-_More_on_advanced_PBX_object_properties_and_behavior#Group_pickup]]

[[Reference11r2:Concept_Group_Pickup_across_PBXs]]

Howto13r3:Step-by-Step set up a conference diagnostics app

2025-10-02T11:27:01Z

Nwe:

[[Category:Step-by-Step|set up a conference diagnostics app]]
[[Category:Concept_Conference]]

{{HOWTOMOD13r3 Translation Info}}

This document explains how to get a conference diagnostics app up and running.

[[image:set-up-conf-diagnostics-overview.png]]
== Purpose ==
The conference diagnostics app allows administrators to get an overview of all conference rooms of a single conference interface. Administrators can see all conference rooms and participants, as well as their video channels and ICE candidates.

=== Features ===
* A running conference diagnostics app assists you in troubleshooting issues related to the conference interface.

== Requirements ==
* innovaphone PBX version 13r3 and up
* A CONF/SCNF interface for conferences for the number of participants you plan. You can determine which device is suitable for this purpose on [[Howto:How_to_implement_large_PBXs#Technical_data_and_recommended_number_of_users_supported|this wiki page.]]

== Things to know before you begin ==
* You need to be able to access the advanced user interface of the PBX as well as the device running the conferencing interface. We recommend including both devices in your ''Devices'' app and getting access via ''Devices''.
* No App Service is required for this app. The app gets its information directly from the CONF/SCNF interface itself.

== Configuration ==
In this scenario, we want all users who inherit their settings through the ''Config Admin'' template to be able to monitor all conferences within the conference interface.

=== Configuration of the Conference interface ===
The first step is to go to Interfaces/CONF or Interfaces/SCNF on your device running the conference interface and set an ''App Password''. You can choose any password you want. Additionally, copy the app URL to your clipboard.

[[image:set-up-conf-diagnostics1.png]]

=== Creating an app object ===
As a second step we will create a new object in the PBX (PBX/Objects). Please select an ''App'' object from the drop-down menu and then click ''new''.

[[image:set-up-conf-diagnostics2.png]]

Please configure the following parameters in the General tab:

*Long Name: This name is displayed to all users, so we named the app ''Conference Diagnostics'' in our example. Keep in mind that this name has to be unique in the system. You can change this name later on without breaking anything.
*Name: Like the long name, the name must also be unique. Do not use spaces for the name. In our scenario we used the name conf-diagnostics.
*Password: Please configure the same password as you did in the first step

[[image:set-up-conf-diagnostics3.png]]

Please configure the URL in the ''App'' tab of the object:

*The URL has to start with: '''https://'''
*The URL is followed by the IP address or DNS Name of the Box running the conference interface: '''conf-box.dvl-vgr.net'''
*Copy the URL of the app from the clipboard as the rest of the URL(without the preceding dot). For example, '''/CONF/innovaphone-conference-diagnostics'''. Note that the path is different depending on whether you are using a SCNF or CONF.

Please remember to press '''Ok''' afterwards.

[[image:set-up-conf-diagnostics4.png]]

=== Distribute the app to all administrators ===

Please open the Templates PBX manager plugin and add the ''conf-diagnostics'' App to the Apps section of the Config Admin Template.

[[image:set-up-conf-diagnostics5.png]]

== Verification ==

Open the All Apps section in the administrator's myApps client and open the ''Conference Diagnostics'' app. You should now see every call to a conference room on the conference interface.

Example of the App

[[image:example-conf-diagnostics-App.png]]

Example of the opened App and explanation of the details

[[image:example-conf-diagnostics.png]]

== Known issues ==

== Related Articles ==
* [[Reference13r3:Concept_Conference_Web_Access]]
* [[Reference13r3:Concept_Conference]]
* [[Courseware:IT_Connect_-_08.1_Conferencing]]

Howto15r1:Step-by-Step Restore a system

2025-10-02T11:26:34Z

Nwe:

{{HOWTOMOD13r3 Translation Info}}


[[Category:Step-by-Step|Restore a system]]

This article explains all the necessary steps to recover a system once it has crashed and you cannot access the App Platform or the PBX.

[[image: restore-system-overview.png]]

== Purpose ==
In the event of a disaster, you will want to restore your system as quickly as possible. If a factory reset is your only option, here are some important steps to take afterward.

=== Features ===
*Restore a system to normal operation

=== Limitations ===
* There is no automated process to restore app services except the ''Install''.

== Requirements ==
* Innovaphone App Platform
* any device running a PBX in V13 or up
* You will need the PBX configuration as a complete file (not with standard password) and the ''.dump files'' of every App service instance which was created by the Devices App backup process, as we explain it [[Courseware:IT_Connect_-_06.0_Managing_Devices#Backup|in our IT Connect training.]]

[[image: restore-system-1.png]]

Hint: Please note that you cannot access the backup files instance if the App Platform fails. Therefore, we recommend saving the backup on an external system.

== Things to know before you begin ==
* To restore the System you need the ''Administrator password'' which was created during the initial installation. This password could have been overwritten by the system password in devices.

[[image: restore-system-2.png]]
== Configuration ==
The first step depends on your setup.
* If you have a local Application platform, you need to start the ''Install'' by selecting ''Master PBX to start a new system with local app platform''
* If your design requires you to have an external App platform, you need to have a virgin AP already in place. This means that an AP with only an AP Manager and Webserver, with default password ''pwd'', should be in place. You can download an App Platform image for a supported hypervisor in our store(http://store.innovaphone.com/release/download.htm). If you have an external App Platform you start the ''Install'' by selecting ''Master PBX to start a new system with external App Platform''

*Then, you go through the installation process. The configuration set up during installation is not important because we will overwrite it with our backup anyway, but we recommend setting the domain correctly as this will save you a click later on.

Please save the emergency passwords of this install as well, as we will use it (the Administrator Password and the PBX Password)!

===Restore PBX===
As a next step, we want to restore the PBX, therefore we will need to access the Advanced UI of the PBX.
* You do that by entering https://<ip-of-PBX>/admin.xml?xsl=admin.xsl
* If a password is required, please use ''admin'' as username and the ''Administrator password'' saved from the ''Install''.
* Go to General/Admin and set the password to what it was before the crash.

[[image: restore-system-3.png]]

* Go to Maintenacnce/Upload/Config and select the PBX config from your backups.
* Click Upload

Hint : keep in mind that by restoring on another hardware, may involve change in the configuration ''(e.g: local gateway registration with mac address)'' and licenses need to be uploaded as well. In case of an IPVA, we recommend to assign the same MAC address as before.

===Restore Manager===

Now we are going to restore the configuration of the ''App Platform Manager''.
* Click on Manager
* Then ''Restore''
* Afterwards click on ''Upload''
* Select the ''manager-0.dump'' file
* Upload it
* Wait until it is successfully restored.

[[image: restore-system-4.png]]

===Restore all Apps===
Fortunately, the ''Install'' process has already set up many App Services for us. For the ones that are not, you will need to install them one by one from the App Store in the top right corner. Just search for the app, select the version you want to install, and then press Install. You do not need to set up an App Instance because restoring the backup will do it for us.

* To restore the contents of an app, click on the app service
* Then ''Restore''
* Afterwards click on ''Upload''
* Select the ''xxxxx-0.dump'' file
* Upload it
* Click on OK to overwrite the existing instance.
* Start the Instance

[[image: restore-system-5.png]]

Hint : Special care for the Files App service, it contains 2 instances ''files'' and ''backup-files'', do not forget to restore both instances
== Verification ==
Start your myApps client and try any app you want. It should work as before

== Known issues ==

== Related Articles ==
[[Courseware:IT_Connect_-_06.0_Managing_Devices]]

Howto13r3:Step-by-Step Offline Firmware/App Store on HTTP Server

2025-10-02T11:26:23Z

Nwe:

[[Category:Step-by-Step|Offline Firmware/App Store on HTTP Server]]

{{HOWTOMOD13r3 Translation Info}}

Setting up your own firmware/app distribution doesn't have to be complicated. There are easy-to-use HTTP servers on the market that can help.

[[image:sbs-appstore-http-overview.png]]

== Purpose ==
Some customers do not allow outgoing HTTP traffic to the innovaphone App Store. In such cases, firmware files have to be hosted on a local web server. Thanks to a software like Simple HTTP Server, you can host them on your PC.

=== Features ===
* Deploy update job to the devices in an isolated environment
* The administrator can choose with firmware versions is hosted

=== Limitations ===
* Since the environment is isolated, you must download the firmware files from the innovaphone App Store beforehand.

== Requirements ==
* innovaphone PBX & App Platform
* 3rd party HTTP Server
* A PC to download a HTTP Server.

== Things to know before you begin ==

== Configuration ==
This is a brief summary of our scenario:

*The customer does not allow outgoing HTTP traffic to the innovaphone app store.
*Firmware updates should be deployed to all devices and apps.

=== Preparation===
* Download the desired firmware of gateways, phones and apps to your notebook before connecting to the isolated network. It is crucial to download all firmware as ''download package'' [[Courseware:IT_Connect_-_06.0_Managing_Devices#Update|as described in the training.]] We will need the ''apps.json'' and ''firmware.json'' files, which are automatically created when you download the firmware as ''download package''.

*These are the necessary steps:
** open http://store.innovaphone.com/release/download.htm
** click on ''Preselection''
** click on ''innovaphone AG apps & software/13rx''
** click on ''firmware/13rx''
** click on ''linux'' and select the newest version
** click on ''Apply selection''
** click on ''Download package''

*After downloading the package, you need to unzip it

===HTTP Server ===
You can use any HTTP server you like. For the purpose of this article, we used [https://apps.microsoft.com/store/detail/simple-http-server/9NT5T97KHPQG Simple HTTP Server from Firefly]. This application can be downloaded from the Microsoft Store, is free, and has a fairly simple interface.

#In the Simple HTTP Server application, click Select Server Folder and select the folder you just unzipped.
#Enable the Simple HTTP Server by sliding the switch. Simple HTTP Server opens a port on your PC.

[[image:sbs-appstore-1.png]]

== Verification ==

==== Using the HTTP Server during Install ====
During install the Server path has to altered to point to the folder hosted on your PC.

[[image:sbs-appstore-2.png]]

It is important to note that the path must point to the folder where your apps.json and firmware.json files are located. Therefore, you might need to add the folder name to the path as shown in the screenshot.

[[image:sbs-appstore-3.png]]

==== Using the HTTP Server for an Update====
Of course, you can also use your local app store for an update. Just open your devices app and select ''Update'' in your domain. You have to change the paths to point to your firmware.json and apps.json files.

[[image:sbs-appstore-4.png]]

==== Using the HTTP Server for installing Apps====
If you want to install new apps, you can change the App Store URL in the AP Manager to point to your local App store.

[[image:sbs-appstore-5.png]]

== Known issues ==

== Related Articles ==
*[[Courseware:IT_Connect_-_06.0_Managing_Devices#Update]]
*[[Reference13r2:Concept_App_Service_Devices]]
*[[Howto13r1:Step-by-Step_Offline_Firmware/App_Store_on_SSD]]

Howto13r3:Step-by-Step Media Relay connection for third party phone

2025-10-02T11:26:05Z

Nwe:

[[Category:Step-by-Step|Media Relay connection for third party phone]]

{{HOWTOMOD13r3 Translation Info}}

This article defines the recommended configuration to connect third party phones also via network boundaries. The configuration ensures media connectivity and compatibility to WebRTC without relying on ICE and DTLS on the phone.

[[Image:Media-relay-endpoints-overview.png|media-relay-endpoints-overview.png/]]

== Purpose ==
A common problem is that third-party SIP phones do not support ICE or DTLS protocols. As a result audio connections across NAT boundaries are often a challenge. To solve this problem, you can enable Media Relay, but the Media Relay endpoint address must be a public IP address so that the external SIP phone can send its audio to this address.

Furthermore, WebRTC endpoints require ICE and DTLS to establish an audio connection. Since not all SIP phones support these features, enabling Media Relay for these 3rd party phones solves this issue.

=== Features ===
*Send Audio traffic across NAT boundaries without using the ICE mechanism.

== Requirements ==
*The firmware has to be at least 13r3
*innovaphone PBX
*3rd party SIP phone
*TURN Server
*External endpoints must be connected via the innovaphone Reverse Proxy

== Things to know before you begin ==
*The public IP address of the network (External IP of the Firewall or NAT Router)
*You must have access to the firewall or NAT router to be able to configure port forwardings
*You could route the RTP directly to the PBX, without using a TURN server. This is not recommended as it would allow attacks on your PBX.
*If your third party SIP Phone supports ICE this solution doesn't apply. No need to use Media Relay.

== Configuration ==
In this scenario, we are going to configure the above picture to allow an external third party SIP device to send its audio to an internal destination. This Wiki article does not explain how to register a SIP phone to an innovaphone PBX via a reverse proxy. If you need help to accomplish this, please read the instructions in our [[Courseware:IT_Advanced_-_07_Public_access_to_PBX_resources_(practice)|Advanced Training Part 2]] materials.

Please register your SIP device to your PBX via the reverse proxy.

=== Configuration on the User Object ===
Please set the option ''Media Relay'' on the hardware ID of the user object on which your SIP device is registered. Do not set the Media Relay option globally in the PBX (PBX/Config/General)! This option is no longer required as you can enable Media Relay for each hardware ID individually.

[[Image:Media-relay-endpoints-1.png|media-relay-endpoints-1.png/]]

=== Configuration of the PBX ===
Under PBX/Config/General of your Advanced UI you will find a configuration option called ''Media Relay Endpoints''. Leave the ''Firewall public IP'' configuration empty, as this option is only necessary if you would route the traffic directly to the PBX. As mentioned above, this is not recommended. Instead, use a TURN server that you would use for your myApps client anyway.

To enable the use of a TURN server you must to activate the checkbox right next to the ''Media Relay Endpoints'' configuration. This option allows the PBX to send the TURN server IP address as the "connection address" in the SDP for all devices that are registered via the reverse proxy and use Media Relay. (see first step)

[[Image:Media-relay-endpoints-2.png|media-relay-endpoints-2.png/]]

In order to send the correct (public) IP address, you need to configure the option ''TURN Public Address'' in your TURN server settings(IP4/NAT/General).

[[Image:Media-relay-endpoints-4.png|media-relay-endpoints-4.png/]]

=== Configuration-check of the Reverse Proxy (RP) ===
Although this step-by-step article is focussing on relaying the voice correctly for external SIP-phones, there is one extra config-check which must be considered for a good and lasting RTP-stream.

In the Advanced UI of the Reverse Proxy (RP) you can/must configure the '''Public NAT router address''' under Services/Reverse-Proxy.
Here you can configure the Public/External IP address of the Reverse Proxy if the Reverse Proxy is not acting/configured as NAT Router.
This config-field is used to adjust/change the 'Record-Route'-Headers in the SIP-signalling towards the external SIP-device, to force future requests in the dialog to be routed through the proxy.

[[Image:RP-Public_nat_router_address.png|rp-public_nat_router_address.png/]]

===RTP Range Configuration===
If you want to restrict connections to a specific port range, you can create an RTP port range on your TURN server. Keep in mind that this range will be used for every call from any device that is using this TURN server. This applies to both internal and external devices, so you should not restrict the number of ports too much. The RTP port range is configured on IP4/General/Settings of your TURN server. Set the ''First UDP-RTP Port'' and then the ''Number of Ports''.

[[Image:Media-relay-endpoints-3.png|media-relay-endpoints-3.png/]]

===Firewall Configuration===
You need to create port forwardings on your firewall. A port forwarding for your RTP/UDP ports must be configured towards the TURN server. E.g If you configured a RTP port Range 16384 to 32767, a port forwarding for those exact ports have to be configured on your firewall.

== Verification ==

Please call any extension number in your PBX. You should be able to hear and talk to the other party on the call.

== Known issues ==
By default, calling a Voicemail has no audio and while calling a conference there is also no audio after the PIN has been dialed. However there is a workaround by forcing these calls over an extra Gateway with Media-Relay activated.

== Known Limitations ==
Currently Video relay it's not supported with this setup, only Audio relay.

== Related Articles ==
*[[Reference13r3:Concept Third Party Phones]]

Howto14r2:Step-by-Step innovaphone Services over Reverse proxy with offline PBX installation

2025-10-02T11:25:50Z

Nwe:

If you have difficulty understanding the written language, we recommend to use https://deepl.com for translation.If installed, you can also use the translation function of your browser by right-clicking.

This article provides detailed step-by-step instructions for situations where your PBX is offline due to security measures, but you still need to access essential services like Push, Store, and Provisioning. By following these steps, you can ensure that your network remains secure while continuing to use these services without direct internet access to the PBX.

In this article, the various networking devices mentioned are all by us, innovaphone. However, it is not necessary to use only our devices. In fact, we advise against it if you have a large system with a high traffic load, as our devices are not designed to handle massive amounts of data. In small or lab environments, however, there should be no issues. Under General Configuration you can find a written general Configuration.

== Applies To ==

* PBX Versions 14r2 and greater

== Problem Details ==
The situation is that the PBX cannot be directly connected to the Internet for security reasons. The problem that arises is that our services, which require an Internet connection, such as Push, Store, and Provisioning, cannot be used. This article will provide you with a comprehensive, step-by-step guide on how to configure your setup so you can still utilize these services while maintaining the security of your PBX system.

== System Requirements ==

* PBX
* innovaphone Reverse Proxy
* Admin Access to the PBX Systems
* IP Addresses of the PBX Systems

== General Configuration ==
To resolve the issue of your PBX being offline and unable to reach online services, one solution is to route the connection through a reverse proxy. The general steps to achieve this are as follows:

* On a DNS server (preferably one that's already in use and distributed via DHCP to other devices), create entries for push. -, store. -, and config.innovaphone.com. These entries should point to your reverse proxy.
* This ensures that any time these domains are requested, the traffic is routed through your reverse proxy.
* On your reverse proxy, configure the appropriate entries (like below) so that the traffic can be routed to the Internet, resolving the connectivity issue.
* Keep in mind that the Reverse Proxy has to use a different DNS-Server in order to route the Requests, otherwise you would build a loop and not go to the Internet

== Applying to Push Services ==

=== PBX ===
Enter a local DNS entry in the PBX under "Services -> DNS -> Hosts-> under "New Resource Record" and select "A" from the drop-down menu.

* "Name": push.innovaphone.com
* "IP addresse:" Ip addresse of the Reverse Proxy

then press OK

[[image:Ese_DNS_Push2.png|ese_dns_push2.png/|ese_dns_push2.png/]]

=== Reverse Proxy ===
In the reverse-Proxy are two entry to make

* 1. Under DNS Suffix on "new"
** ID: 1
** Suffix: .innovaphone.com

[[image:ese_03.png|ese_03.png/|ese_03.png/]]

* 2. Under Hosts on "new"
** 1.Name: push.innovaphone.com
** 2."http://<host> -> out:" @push#1
** 3."http://<host> -> TLS:" 443

[[image:Ese_RP_Push.png|ese_rp_push.png/|ese_rp_push.png/]]

== Applying to Provisioning service ==
Since the Provisioning service is called through the Devices's, Your AP-Platform will need to be able to resolve "config.innovaphone.com". You'll also have to configure the DNS Server of your AP-Platform so that it uses your local PBX as a DNS Server. One way to apply it, that the DNS server is the PBX, is to make the entry in your DHCP server

[[image:Ese_server_config.png|ese_server_config.png/|ese_server_config.png/]]

The Ports 443 and 80 have to be configured in the RP's Host entry, depending on your network you may have to configure your Firewall.

=== PBX ===
1. Another local DNS entry in the PBX under "Services -> DNS -> Hosts-> under "New Resource Record" and select "A" from the drop-down menu.

* "Name": config.innovaphone.com
* "IP addresse:" Ip addresse of the Reverse Proxy

then press OK

[[image:Ese_DNS_Provi2.png|ese_dns_provi2.png/|ese_dns_provi2.png/]]

=== Reverse Proxy ===
In the reverse-Proxy are two entries to make, if a Suffix entry is already configured you can just use the same entry, and skip this part.

1. Under DNS Suffix on "new"

* ID: 1
* Suffix: .innovaphone.com

[[image:ese_03.png|ese_03.png/|ese_03.png/]]

2. Another Reverse Proxy entry, for Provisioning.

[[image:Ese_RP_Provi.png|ese_rp_provi.png/|ese_rp_provi.png/]]

== Applying to Store service ==

=== PBX ===
The Problem with Store service is that "store.innovaphone.com" is called by the AP-Platform, so it doesn't natively use the local entries.

To fix this, check the "Enable DNS-Server" box, add the "store.innovaphone.com" PBX Local-DNS entry and hit "OK".

[[image:Ese_DNS_Store2.png|ese_dns_store2.png/|ese_dns_store2.png/]]

You'll also have to configure the IP entry of your AP-Platform so that it uses your local PBX as a DNS Server.

One way to apply it, that the DNS server is the PBX, is to make the entry in your DHCP server

[[image:Ese_server_config.png|ese_server_config.png/|ese_server_config.png/]]

=== Reverse Proxy ===
In the reverse-Proxy there are two entries to make, if a Suffix entry is already configured you can just use the same entry, and skip another Suffix entry.

1. Under DNS Suffix on "new"

* ID: 1
* Suffix: .innovaphone.com

[[image:ese_03.png|ese_03.png/|ese_03.png/]]

2. Another RP entry pointing to store.innovaphone.com

[[image:Ese_store_RP.png|ese_store_rp.png/|ese_store_rp.png/]]

== Troubleshooting ==

=== Services don't connect ===

* Ensure that the IP address of the RP is also reachable locally in the local PBX. You can check this under "Maintenance -> Diagnostics -> Ping -> Ping." If the entered IP address is not reachable, the Services will not work.
* Ensure that the Host entry is correctly recorded.
** Check if the first "." is specified in the suffix
** And ensure that there are no dots in @push#1

=== Trouble with Provisioning ===
One issue that may occur is if a code is generated, but the provisioning does not work after entering it on the phone. This could be due to the port configurations, the Devices gets the codes from config.innovaphone.com over :443, and the Phones send them back over :80, so be sure that in the RP's Host entry of "config.innovaphone.com" the ports are not missing and configured correctly.

Also, make sure the Phones are able to resolve "config.innovaphone.com" to a Reverse Proxy that can route out to the internet.

== Verification ==

=== Push ===
To test if push is working, log into a user account via mobile phone and completely close MyApps. To completely close MyApps, open the burger menu on your home screen and hit "Exit." Now, call the logged-in user and see if the call gets pushed through. If so, the push now works over your reverse proxy.

=== Store ===
To test the Store service, open your AP Manager and go to the App Store button on the upper-corner of the right-hand side, Check if you are able to open the app store and try downloading and running any new APP

=== Provisioning ===
To check Provisioning Try adding a hard-phone to your PBX
[[Category:Step-by-Step|innovaphone Services over Reverse proxy with offline PBX installation]]

Howto13r3:Step-by-Step Hot Desking

2025-10-02T11:25:27Z

Nwe:

{{HOWTOMOD13r3 Translation Info}}

{{datasheet_header|Hotdesking|Step by Step innovaphone Hotdesking Installation}}
<div class="datasheets-einspaltig">



[[Category:Step-by-Step|Hot Desking]]

Hot desking should allow employees to request any desk of their choice. Therefore, the desk phone should be switched to the respective user profile at the push of a button. This document explains the necessary configuration steps. Additionally, please take the security considerations seriously.

[[image:hot-desking-overview.png]]

== Purpose ==
Users should be able to claim any desk they want. A default user that allows emergency calls should be registered on the phone by default. In addition, a hot-desking function key should allow enabling a user profile on the phone.

=== Features ===
* Easy way to activate a user profile on a desk phone with the click of a button
* Users must enter their password in addition to their username or number to authenticate themselves

=== Limitations ===
* Configuration is lost after the reboot of a device
* There is no automatic logout. The user profile will remain active until the hot desking button is pressed again or the device is rebooted.

== Requirements ==
* Innovaphone PBX version 13r3 and up
* The default user (''Hot Desk Base'' user) requires one port license
* innovaphone desk phone providing function keys (IP11x, IP2x2)

== Things to know before you begin ==
* Think of a strong password. If secure passwords are required, a password generator may be used (e.g. https://www.lastpass.com/features/password-generator).
* The admin user who performs configuration and provisioning must have access to the phone app, devices, and users, as well as their APIs and his apps tab

== Configuration ==
In this scenario, we will take all the necessary steps to configure a default user on all Hot Desking phones. Each phone will have a function key that allows each user to log into their personal profile. Furthermore, this base user will only be allowed to call emergency numbers.

===Create a config template - PBX Manager ===
First we recommend to create a new template. Please go to the ''Templates'' PBX manager Plugin and add a new template.

[[image:hot-desking-1.png]]

Assign a name to this template and configure the following options in the ''settings'' section

* Call filter: select ''intern''
* Call Forward filter: select ''intern''
* Store phone config on PBX: ''Enable''

[[image:hot-desking-2.png]]

Please be aware that this template should not inherit a configuration from another template. A license or access to an app is not requirement.

===Create a ''Hot Desk Base'' User - Users Admin app===
Open your ''Users Admin'' app and add two new user. The first user will be used as a default user on all Hot Desking phones. In this example, we will name the user ''Hot Desk Base'' but you can choose the name as you wish

* Hide from LDAP: ''Enable''
* Username: ''hot-desk-base''
* Password: Give a strong password
* ID: ''Hot Desk Base''
* Extension: Please configure a extension number outside your numbering plan. In this example we set the Extension number to ''#10''
* Template: Please select the template you created in the last step

[[image:hot-desking-3.png]]

Afterwards please create another user which is used during the provisioning for a short period of time.

* Hide from LDAP: ''Enable''
* Username: ''temp''
* Password: Give a strong password
* ID: ''temp''
* Extension: Please configure a extension number outside your numbering plan. In this example we set the Extension number to ''#20''
* Template: Please select the template you created in the last step
[[image:hot-desking-4.png]]

===Create Hot Desking Hardware ID ===

As a next step please open ''Register phones'' in the top menu of the users admin app and click on ''+ Hotdesking registration''.

Select the phone app and click on next.

Select your ''hot-desk-base'' user and any user which should be able to do hot desking. Afterwards press next and then finish.

[[image:hot-desking-8.png]]

===Create category - Devices app ===
Go to your Devices app and select ''Categories''. Afterwards click on ''Add category''. Assign a name like ''PBX name Hotdesking'' and enable the check mark ''Provisioning category for device configuration deployment''

[[image:hot-desking-5.png]]

===Devices Configuration - Devices App===

As a next step select your domain in our devices app and click on ''Device configuration''.

=====Assign category to Phone configuration =====

Open your ''Phone device configuration'' and add the category you just created to your Phone device configuration by selecting the category and then click the + symbol. This way the device will receive all necessary options like gatekeeper ID or primary gatekeeper address from this device configuration.
Afterwards don't forget to press OK.

[[image:hot-desking-6.png]]

=====Create Expert configuration =====

As a final step of your configuration please click on ''Define device configuration'' and select ''Expert configuration''
* Please assign a descriptive name like ''Hot Desking''
* Select the category you just created from the drop down menu and click the + symbol
* Please create an update script to register your phone with your ''Hot Desk Base'' user and configure a ''Hot Desking'' function key. You can use the following example, but please change ''<your secure password>'' to the ''Hot Desk Base'' users password you created earlier

config add PHONE SIG /h323 hot-desk-base /gk-pwd <your secure password>
vars create PHONE/USER-CFG/00000 p %3cuser%3e%3cf+id='0'+label='hot-desking'%3e%3chotdesk+label='log+out'+pbx='1'/%3e%3c/f%3e%3c/user%3e
mod cmd UP1 check iresetn hot-desk
config write
config activate

[[image:hot-desking-7.png]]

== Verification ==

===Provision a phone to ''Hot Desk Base'' user - Users Admin app===

Create a provisioning code through for the temp user in the users admin app. Enter this provisioning code into a phone that has previously been reset to factory defaults. The phone will be provisioned and will be registered on the ''Hot Desk Base'' user.

[[image:hot-desking-9.png]]

===Use Hot-Desking function key - Phone===

Through the expert configuration script, the ''Hot Desk Base'' user received a Hot Desking function key. By pressing the key, the user can input his name or number and his personal password. The phone creates a registration on the PBX for this user. Note that users must have a hardware ID on their user object that matches their name. Furthermore if the registration needs to pass a reverse proxy, you will need to activate the corresponding flag or the registration will fail.

===Tutorial in a Video===

Here you can find this tutorial in a Video to help you even more.

<youtube>https://www.youtube.com/watch?v=RHow_H7mIes</youtube>

=====Security Considerations =====

A common requirement is that users have problems entering their personal password through the phone UI. In this case, we recommend using OAuth2 Authentication [[Reference13r3:Concept_OAuth2_Windows_Authentication]] for myApps. This way, you can reuse the user's personal password since myApps uses the Windows Domain password for authentication. Please be aware that using a simple password is a security concern because a malicious attacker could guess it. As result, please make sure you configure IP Filter and set No of Regs w/o Pwd. to 0.

== Known issues ==

== Related Articles ==
You may also consider to use the [[Howto:Hotdesk_-_MediaRunway_-_Partner_App|MediaRunway Hotdesk partner application]] for simple phone assignment via the Windows myApps Client when changing workstations.
</div>
{{Template:Datasheet_footer|EN}}

Howto14r2:Step-by-Step extended 3 way conference

2025-10-02T10:59:45Z

Nwe:

[[Category:Step-by-Step|extended 3 way conference]]

{{HOWTOMOD13r3 Translation Info}}

=='''Pre-information'''==
'''In this Article we will show you how you can extend your 3-way conference to an Innovaphone Conference Room. For instance, if you have an ongoing 3 way call and you expect another participant to join, you can simply transfer all participants as well as new incoming calls to an Innovaphone Conference Room.'''

===What do you need?===

* myApps Conference Room
* Conference Licenses
* More than 3 Participants
* myApps Native Client with Softphone in use
* myApps at least V13r3

===Purpose===

* On a Desk Phone, you have the option to concatenate a 3-party conferencing call to have more than 3 participants
* On the Softphone you don't have the option to add another participant, so transferring is necessary

===Procedure===

* If the Conference Room does not have a PIN follow '''Scenario A'''

* If the Conference Room has a PIN, you generally have the following options:
** Deactivate the room PIN temporarily or give the PIN to your participants - they will have to type it in to successfully join. Follow '''Scenario A'''
** If you are comfortable to enter the PIN for the participants follow '''Scenario B'''
** Else if you don't want to enter the PIN follow '''Scenario C'''

== '''Scenario A: Transfer to a Conference Room without a PIN''' ==

=== You are part of a three-party conference call and want to transfer the participates to a MyApps Conference Room. ===
This method also works when the room has a PIN set but the transferred participants have to know and enter the room PIN manually. Keep in mind when sharing PINs to participants that this will allow them to join this conference room at any given time. So you might want to change the PIN afterwards.
* First you open the call information on your Softphone for the ongoing call

[[File:call-info-softphone.png|/Call-info-softphone.png|/Call-info-softphone.png]]

* Now you select the participants you would like to transfer to the conferencing room and click the ''Transfer Button''

[[File:softphone-panel.png|/Softphone-panel.png|/Softphone-panel.png]]

* After that, your ''Users Window'' appears
* Now you select the conferencing room that you would like to transfer your participant to and click on ''Transfer to'' ''Button''

[[File:room-search.png|/Room-search.png|/Room-search.png]]

* Do that procedure with every participant that you would like to transfer to the conferencing room
* At last join the conference yourself
Now you have successfully transferred the participants to the MyApps Conference Room.

== '''Scenario B: Basic Transfer to a Conference Room with a PIN''' ==

=== You are part of a three-party conference call while you get an incoming call. You want wo transfer everybody to a MyApps Conference Room. ===
You transfer participants by calling the conference room yourself and then connect the participants to the room. This method has the significant advantage that the participants don't have to know and enter the PIN themselves. Using the Softphone App you might have to dial in the room PIN multiple times though.
* First take the incoming call. The conference party members get held
* Now move to your phonebook and call the conference room. If necessary put in the PIN to the MyApps Conference Call. You can also use the Conference App to dial in to the room and skip the PIN prompt
[[File:ConfRoom.jpg|alt=ConfRoom|border|/ConfRoom.jpg|/ConfRoom.jpg]]

* Go back to your calls and choose the participant you would like to move to the conference room and select the option ''"Connect with''..."
[[File:ConnectWithRoom.jpg|alt=ConnectWithRoom|border|/ConnectWithRoom.jpg|/ConnectWithRoom.jpg]]

* Repeat these steps for every participant that you would like to transfer to the conferencing room
Now you have successfully transferred the participants to the MyApps Conference Room.

== '''Scenario C: Advanced Transfer to a Conference Room with a PIN''' ==

=== You are part of a three-party conference call and want to transfer participants to a conference room completely without the need to enter the PIN number. ===

=== Configuration ===
Before you can transfer someone from a Call to a MyApps conference room, you have to make some pre-configurations to make it possible to transfer someone without entering the conference room pin.
 

* First we create a new Trunk Line object and give the following parameters -> ''Long Name, Name, Number'' and ''Password''
[[File:Trunkobject2.jpg|alt=Trunk Object configuration|border|/Trunkobject2.jpg|/Trunkobject2.jpg]]
 

Now we register the newly created Trunk Line object to your gateway:
* Go to ''Gateway/GK'' on your PBX where you would like to register the trunk line to

* Select the gateway interface where you would like to register your trunk line to (We choose ''GW1'' for this example)
* Enter the following parameters:
** ''Name''
** ''Protocol''
** ''Mode:'' Register as Gateway
** Address: (has to be the Address where the Trunk Line object was created)
** ''Password'': (has to be identical to the password that you configured on the Trunk Line object)
** ''Alias List/Name:'' trunk (has to be identical to the Devices Hardware Id of the Trunk Line object)
[[File:Trunkgateway.jpg|alt=Gateway configuration|border|/Trunkgateway.jpg|/Trunkgateway.jpg]]
 

Note: Alternatively you can use your hardware ID to register the gateway interface to the Trunk Line object. Leave empty ''Password'' and ''Alias List/Name'' in the gateway interface configuration and in the Trunk Line object set ''Hardware ID'' to ''009033AABBCC-GW1'' (replace ''AABBCC'' according to your hardware ID).
 
As next, we create a new route which goes to the gateway interface that we just created (GW1):
* Go to ''Gateway/Routes'' and click on the small icon in the upper left to add a new route
* Optional: Add a description
* In the left choose the gateway interface as ''source interface''
* Also set it as ''destination interface'' in the drop-down menu on the right
[[File:TrunkRoute2.jpg|alt=Route back configuration|border|/TrunkRoute2.jpg|/TrunkRoute2.jpg]]
 

Now we add the dialing rule to our route.
*Choose an ''input number'' (this is the number that you'll have to dial along with your trunk line number to reach that route. In this example it's ''2'')
* Set the ''output number'' (this number has to be the number of your conference room. In this example it's ''00301'')
With this configuration your participants still have to enter the PIN of your MyApps Conference Room, so we also have to add a DTMF dialing to the output number so that the PIN will be dialed automatically. (Example: my room number is ''00301'', my Room PIN is ''1234'', this means the output number is ''00301^1234'').
* As discussed add ^ followed by the PIN to the output number
[[File:TrunkRoute.jpg|alt=Route configuration|border|/TrunkRoute.jpg|/TrunkRoute.jpg]]
 

Our new route should now appear like this in the routes overview: 
[[File:RouteOverview.jpg|alt=Route Overview|border]]
 

Now you have successfully created a Route to our MyApps Conference Room.
 

=== Testing the configuration: ===
Call the trunk line number followed by the number of your input route. For example:
''63'' (which is the trunk line number) and ''2'' (which is the input route number) -> This means we have to call ''632''.
The call should be directly routed to your MyApps Conference Room.
 

=== Transferring calls: ===

* First you open the call information on your Softphone for the ongoing call
[[File:call-info-softphone.png|/Call-info-softphone.png|/Call-info-softphone.png]]
 

* Now you select the participants you would like to Transfer to the conferencing room and click the ''Transfer Button''
[[File:softphone-panel.png|/Softphone-panel.png|/Softphone-panel.png]]
 

* After that, your users window appears
* Now type ''632'' and click on ''Transfer to''
[[File:transferToRoute.jpg|/TransferToRoute.jpg|/TransferToRoute.jpg]]
 

* Do that procedure with every participant that you would like to transfer to the conferencing room
* At last join the conference yourself
 
Now you have successfully transferred your participants to the MyApps Conference Room.
 

=== Optional Configuration to make Transfer even easier/faster ===

Now you can transfer participants to your conference room using the route number, in this example ''632''. As a finishing touch it might be worth it to add a dummy user which forwards all calls to the route. Now remembering "qc" or "Quick Conference" is all it takes to transfer participants to our MyApps Conference Room.
 
* Create a User with the following parameters: ''Long Name, Display Name, Name'' and ''Number''
[[File:Quickconfuser.jpg|alt=quickconfuser|border|/Quickconfuser.jpg|/Quickconfuser.jpg]]
 

* Now add a CFU to that User towards the trunk line number of your MyApps Conference Room. In this case ''632''
[[File:Cfu.jpg|alt=Cfu|border|/Cfu.jpg|/Cfu.jpg]]
 

Now you have successfully created a dummy User to directly transfer into a MyApps Conference Room without the need of entering the PIN.

== Known issues ==
* Currently with 14r2 transferring a Participant to a conference room does not support Video. We are working on a fix for so that video is also working again.

Howto14r2:Step-by-Step Distribute a custom device certificate

2025-10-02T10:59:31Z

Nwe:

[[Category:Step-by-Step|Distribute a custom device certificate]]

{{HOWTOMOD13r3 Translation Info}}

This article describes a method to roll out a custom certificate to innovaphone devices.

[[image:device-certificate-overview.png]]

== Purpose ==
Some customers run their own public key infrastructure (PKI) and want to use their own certificate (eg a wildcard certificate like *.company.com). This way you can distribute this certificate to all innovaphone devices.

=== Features ===
* A convenient way to distribute a custom device certificate
* Reboot is not necessary

=== Limitations ===
* The length of the public key should not exceed 2048 bits. This is to limit the CPU consumption on our devices, see [[:Reference11r1:Certificate_management|Certificate management]] for details.

== Requirements ==
* Devices App
* Innovaphone PBX
* Firmware should be at least v14r2sr4
* You need a complete certificate chain containing the private key. We recommend to use a PEM encoded Text file [[:Reference11r1:Certificate_management#Uploading_a_certificate_chain_together_with_the_private_key|as explained here.]]
* Wireshark

== Things to know before you begin ==
* The certificate device configuration in your Devices app only maintains your trust list. As a result it will not distribute the device certificate.
* The pre-installed certificate signed by the Inno-CA remains in the Flash when you upload a new certificate. If you delete the new certificate, the pre-installed certificate will reappear.

== Configuration ==
=== Create Expert configuration ===
* Open your ''Devices'' App->''<your Domain>''-''>Device Configuration''-''>Define device configuration''->''Expert''
* Assign a ''Description'' e.g Device certificate
* Assign the provisioning category to this device configuration that should receive the new device certificate

[[image:device-certificate-1.png]]
=== Get VARS ===
* Open your Wireshark
* Drag and drop the PEM file into your Wireshark
* Your Wireshark will only display a few packets

[[image:device-certificate-2.png]]
* Click on the first packet
* Do a right click on the section starting with Certificate
* Select Copy and then Copy as Hex stream
* Create a first line in your Expert configuration starting with ''vars create X509/CERTIFICATE/00000 pbln'' and paste the Hex stream from wireshark to the end of the line

[[image:device-certificate-3.png]]

* Then repeat the same procedure for each certificate in the certificate chain, but increase the index by 1. e.g. ''vars create X509/CERTIFICATE/00001''

[[image:device-certificate-4.png]]

The private key has to be copied as well.
* Open the last packet in Wireshark and select the BER section. Copy the section as Hex stream as well.
* Create a line ''vars create X509/KEY pbxln'' and paste the Hex stream from the BER packet to the end of the line.

[[image:device-certificate-5.png]]

=== Finish Expert configuration ===
* The last line in our script is: ''mod cmd X509 /servercert-update''
* In the end the expert configuration should look like this:

[[image:device-certificate-6.png]]

*As soon as you save the configuration, the device certificate will be pushed.

== Verification ==
Look at the Advanced UI of the device. You should see a new Device certificate on General/Certificates
== Known issues ==
===High CPU load ===
We only recommend to use a certificate that uses 2048 bit public key length.

== Related Articles ==
*[[Courseware:IT_Advanced_-_09_Custom_certificates]]
*[[Reference11r1:Certificate_management]]

Howto15r1:Step-by-Step Custom Onboarding screen for new Users

2025-10-02T10:59:18Z

Nwe:

[[Category:Step-by-Step|Custom Onboarding screen for new Users]]

{{HOWTOMOD13r3 Translation Info}}

This article explains how to set up a custom onboarding screen in myApps for new Users. This onboarding screen will be displayed once when starting myApps for the first time and in the burger menu at "intro".

[[Image:Onboarding_stepbystep.png]]

== Purpose ==
An individual onboarding screen can be configured and used for the first myApps Login of an User or in the burger menu. By Default, in a new installation, an introduction Video by innovaphone is used to introduce the new myApps User, but it's also possible to use a custom introduction Page or Video (embedded inside a Web page).

=== Features ===
* Custom onboarding screen and Website for new myApps users
* Viewable at all time through the burger menu
* Give new Users an Overview of myApps
* Include anything that might be needed for new Users (Websites, App explanations, How to use myApps,...)

=== Limitations ===

== Requirements ==
* At least PBX Version 15r1
* Custom Web page which is accessable for all myApps Users (internal and/or also external Users)
* Must work inside an IFrame.
* Should follow the language given in the URL parameters, defaulting to "en" if no contents are available for the language. ( See passed URL Parameters: https://wiki.innovaphone.com/index.php?title=Reference15r1:Concept_myApps#Onboarding )
* Should follow the collor scheme ("dark" or "light") given in the URL Parameters. ( See passed URL Parameters: https://wiki.innovaphone.com/index.php?title=Reference15r1:Concept_myApps#Onboarding )
* Must be responsive to work in different orientations (landscape or portrait) and on different screen sizes (Desktop and Smartphone).
* Must reserve a space (width 306px, height 102px) for an overlay button that is displayed by myApps in the bottom right corner.

== Things to know before you begin ==
* This article does not explain how to set up the Web page, this Article will only show the configuration steps which are needed to use a custom Web page as onboarding screen for myApps.

== Configuration ==
In this scenario we will replace the (by Default) configured onboarding URL to use a custom Web Page. For demonstration purposes the Website I will use is the innovaphone Homepage.

=== Set up a Website ===

First, you need to create a Website with all important infomation that should be included. Additionally you can have a Video on that Website, it's up to you what should be included.
Make sure that the following Rules are applied to that Website:

* Accessable for all myApps users
* Support for iFrame
* Should follow the passed Paramenters (see above)

=== Set up this Website as the onboarding Website ===

* Open the Admin UI of your PBX
* Go to PBX - Config - myApps
* Replace or enter (if empty) the Onboarding URL by your own Website URL.

[[Image:Onboarding_url_pbx.png]]

== Verification ==
In the burger menu of myApps should be a new Button called "Intro".

[[Image:Onboarding intro.png]]

* To test if your Website works you can open "Intro" at any time to see the onboarding screen. 
* Brand new myApps Users who never opened myApps before will get this Screen after the first successful Login. 

[[Image:Onboarding_intro_video.png|900px]]

* With a click on the right below textbox "Let's go", the onboarding screen will close.

== Known issues ==
== Related Articles ==
* [[Reference15r1:Concept_myApps]]
* [[Reference15r1:PBX/Config/myApps]]

Howto13r2:Step-by-Step Create a Tutorials App

2025-10-02T10:58:55Z

Nwe:

The goal of the Tutorials app is to provide a quick way to get help when questions arise.

[[Image: tutorials-concept.png]]

== Purpose ==

The Tutorials app provides a quick reference for learning how to use myApps. innovaphone provides video tutorials on how to use common features. Alternatively you can create your own tutorials app containing videos embedded on your website.

=== Features ===

*Provide tutorials for users
*A shortcut to the tutorials app is available in the burger menu of myApps

=== Limitations ===

* Users need access to the tutorials video website. If you want to use videos provided by innovaphone, users will need Internet access.
* Videos provided by innovaphone are available in H.264 and webm. The used video player has to be able to play either of those formats.

== Requirements ==
*innovaphone PBX
*Website hosting the tutorials.(You can use innovaphone's tutorials webpage)
*Used firmware has to be at least 13r2

== Things to know before you begin ==
*The URL to the website hosting the tutorials

== Configuration ==
=== Create app object ===
The first step is to create an app object for your tutorial app. This app object must have a name and a long name. These names can be chosen freely, but remember that users will see the long name of the app in his myApps client.

[[Image:tutorials-name.png]]

In the App tab of the object, you need to define the URL pointing to the website hosting the tutorial content. You can use innovaphone tutorials by configuring this URL:https://www.innovaphone.com/myapps/tutorial.htm .

[[Image:tutorials-url.png]]

=== Grant access to the tuorials app ===

The next step is to allow the user to use the Tutorials app. This can be done either on the "Apps" tab of the user object or via a config template.

[[image:tutorials-template.png]]

=== Create a shortcut in the burger menu of the myApps client ===

As mentioned before you can create an entry in the Burger menu of the myApps client to have a quick link to the tutorials page.

[[Image:tutorials-burger-menu.png]]

You have to add the name (h323id) of your tutorial object as tutorial app under PBX->Config->myApps.

[[Image:tutorials-myApps.png]]

== Verification ==
*Search for the tutorials app in the ''All Apps area'' of your myApps client and press the icon. A new window (IFRAME) will open which redirects you to the tutorials web page
*Go to the burger menu of your myApps client. A shortcut to the tutorials app is available.

== Known issues ==
== Related Articles ==
* [[Reference13r2:Concept_myApps]]
[[Category:Step-by-Step|Create a Tutorials App]]

Howto14r2:Step-by-Step Connector for Microsoft365

2025-10-02T10:58:32Z

Nwe:

[[Category:Step-by-Step|Connector]]

{{HOWTOMOD13r3 Translation Info}}

The Connector for Microsoft365 synchronises Microsoft Teams presences with the innovaphone PBX and back. In this Step-by-Step I will show you how to configure it.

[[File:M365-function2.jpg]]

== Purpose ==
The Presence of a User will be synchroniced From Teams to the PBX and vice versa. This way the Users presence is always up-to-date for both Systems.

=== Features ===
* Synchronisation of the presence between Teams and PBX and vice versa

=== Limitations ===
* Line states set by the PBX does not block calls in Teams
* Maximum number of supported Users per communication User
* Synchronisation Delay due to limitations in the Graph-API
* Communication Users with MFA are not supported as technical communication users for the Connector
* Subscription Timeout

For more information regarding the Limitations please see: https://wiki.innovaphone.com/index.php?title=Reference14r2:Concept_App_Service_Connector_for_Microsoft_365#Known_Limitation

== Requirements ==
* innovaphone PBX
* innovaphone Application Platform
* App(Connector for Microsoft 365)
* PBX-App(innovaphone-microsoft365) license per user - order no. 02-00050-009
* account in Azure Portal of Microsoft (for each of the technical communication users, no permission role needed)
** Each communication user must have a Teams license applied, more infos can be found here
** Each communication user can be used to subscribe the presence of up to 650 users. (If you want to synchronize more than 650 users, you will need a multitude of communication users)
** Must not have multi factor authentication activated
* Must have access from the internet to your App Platform
** This can be done by using a reverse proxy or other firewall
* The public endpoint must have a valid, public signed certificate (in order to make a trusted SSL connection from the Azure cloud to the Application Platform possible)
** A valid certificate is required in all involved network entities - at least in the App Platform and if used in the Reverse Proxy; to ensure transmission of MS365 HTTPS POST requests to the app service in order to send notifications.
* Admin account for Azure Portal (only necessary for granting needed permission for registered app during setup)

== Configuration ==
In this scenario, we will take all the neccessary steps to configure the Connector for Microsoft365 App and the Azure Portal.

=== Create an App for syncing Teams to PBX ===
* First you have to add an app registration in the Azure Portal of Microsoft
** Assign a Name to this App

[[Image:App_Registration_Connector_for_Microsoft365.png|thumb|none|600px|app_registration_connector_for_microsoft365.png/|app_registration_connector_for_microsoft365.png/]]

Switch to the Authentication Tab on the left side
* Enable "Allow public client flows"

[[Image:Authentication_Connector_for_Microsoft365.png|thumb|none|600px|authentication_connector_for_microsoft365.png/|authentication_connector_for_microsoft365.png/]]

Switch to api permissions on the left side

[[Image:Azure_Select_Api-Permission.png|thumb|none|600px|azure_select_api-permission.png/|azure_select_api-permission.png/]]

* Configure delegated permissions
* Add the following permissions
** User.Read.All
** Presence.Read.All

[[Image:Azure_Select_Api-Permission_Delegated.png|thumb|none|600px|azure_select_api-permission_delegated.png/|azure_select_api-permission_delegated.png/]]

* Grant access to the api permissions, if not possible you have to ask an admin

[[Image:APIPermission_Connector_for_Microsoft365.png|thumb|none|600px|apipermission_connector_for_microsoft365.png/|apipermission_connector_for_microsoft365.png/]]

Now configure the following attributes for the Connector for Microsoft365

* Client ID (already given after successful registration of the App) as shown in the picture
* Tenant ID (already given after successful registration of the App) as shown in the picture
* User
* Password

[[Image:Overview_Connector_for_Microsoft365.png|thumb|none|600px|overview_connector_for_microsoft365.png/|overview_connector_for_microsoft365.png/]]

====Create an App for syncing PBX to Teams====

* Create a new App registration for the synchronisation from the PBX to Teams
* The Client ID and Tenant ID has been created automatically. These values are needed to configure the Connector for Microsoft365 App.

[[Image:Overview_Connector_for_Microsoft365_Sync_to_Teams.png|thumb|none|600px|overview_connector_for_microsoft365_sync_to_teams.png/|overview_connector_for_microsoft365_sync_to_teams.png/]]

* Open Certificates & Secrets on the left
* Create a new Client secret
* Save the Value as you need it later

[[Image:Authentication_Connector_for_Microsoft365_Sync_to_Teams.png|thumb|none|600px|authentication_connector_for_microsoft365_sync_to_teams.png/|authentication_connector_for_microsoft365_sync_to_teams.png/]]

* Open api permissions on the left

[[Image:Azure_Select_Api-Permission.png|thumb|none|600px|azure_select_api-permission.png/|azure_select_api-permission.png/]]

* configure application permission and add the following permission
** Presence.ReadWrite.All

[[Image:Azure_Select_Api-Permission_Application.png|thumb|none|600px|azure_select_api-permission_application.png/|azure_select_api-permission_application.png/]]

* Grant access to the api permissions, if not possible you have to ask an admin

[[Image:APIPermission_Connector_for_Microsoft365_Sync_to_Teams.png|thumb|none|600px|apipermission_connector_for_microsoft365_sync_to_teams.png/|apipermission_connector_for_microsoft365_sync_to_teams.png/]]

===Installing and configuring App Platform and PBX===
Install the following App from the App Store on your App Platform:

* Connector for Microsoft365

Now create a Instance and add an App through the PBX-Manager Plugin. Also assign this Admin App to a User/Template.

Note: If you don't know how this works, have a look here: https://wiki.innovaphone.com/index.php?title=Course14:IT_Advanced_-_04_Setting_up_the_Application_Platform#Adding_an_App_Service_instance

===Synchronization from Teams to the PBX===
* First you will need to configure the inbound sync from Teams to PBX
* Open Configuration
**Master PBX - Set the name of your Master PBX ('''Not full DNS name, really just only the PBX name''') Good: [pbx] Bad: [pbx.domain.tld]
**ClientIDSynctoPbx - Insert the Application ID (Client ID) from Azure Portal (from the preparation created Teams to PBX app)
**TenantSynctoPbx - Insert the Directory ID (Tenant) from Azure Portal (from the preparation created Teams to PBX app)
**Notification URL - You need to specify the address Microsoft can send presence updates to.
*** You need to make sure that you define a URL where you can reach your App Platform from the public internet <code>public.dns</code>
*** Next you need the domain you have configured in the app instance before (3.2.2) <code>your.domain</code>
*** Next you need the name of the instance you have configured before (3.2.2) <code>microsoft365</code>
*** The URL will always be terminated by <code>subscriptions</code>
*** <code>https://public.dns/your.domain/microsoft365/subscriptions</code>
*** Save your changes
*** [[Image:Microsoft365_admin_app_2_14r2.png|thumb|none|600px|microsoft365_admin_app_2_14r2.png/|microsoft365_admin_app_2_14r2.png/]]

* Open Manage Teams accounts
* Click on Add Teams account
** Enter username (Azure Portal account in email format)
** Enter password
** Click on the check mark
** Make sure to add an Azure Portal communication user for each 650 user you want to subscribe
** [[Image:Microsoft365_admin_app_2_14r2_user.png|thumb|none|600px|microsoft365_admin_app_2_14r2_user.png/|microsoft365_admin_app_2_14r2_user.png/]]
* After successful configuration and subscription, the Microsoft services will connect to the specified ''Notification URL'' for presence and line state updates.
* For this to work, it is important to make sure that the specified ''Notification URL'' is reachable from the Microsoft services, meaning from the public internet.
'''It can take up to 10 Minutes until all check marks are green, and the sync is working'''

===Synchronization from the PBX to Teams===
* For the outbound synchronization you select from PBX to Teams in the admin app
**ClientIDSynctoTeams - Insert the Application ID (Client ID) from Azure Portal (from the in preparation created PBX to Teams app)
**TenantSyncto Teams - Insert the Directory ID (Tenand) from Azure Portal (from the in preparation created PBX to Teams app)
**ClientSecretSynctoTeams - Insert the shared secret (from the in preparation created PBX to Teams app)

[[Image:Microsoft365_admin_app_3_14r2.png|thumb|none|600px|microsoft365_admin_app_3_14r2.png/|microsoft365_admin_app_3_14r2.png/]]

* For the sync direction from PBX to Teams, the app itself needs visibility permissions (presence, on-the-phone) for the users who should be synced to Teams. The configuration tag is visibility for each user object.

== Verification ==

===Change the presence of a User===

* Change the presence of a User in Microsoft Teams from Available to Busy
** Check if the myApps presence also changes to Busy

* Change the presence of a User in myApps from Available to Busy
** Check if the Microsoft Teams presence also changes

== Known issues ==

==Related Articles==
[[Reference14r2:Concept App Service Connector for Microsoft 365|Concept App Service Connector for Microsoft 365]]

Howto15r1:Step-by-Step connect to shared services

2025-10-02T10:58:07Z

Nwe:

{{HOWTOMOD13r3 Translation Info}}



[[Category:Step-by-Step|Connect]]

With version 15, innovaphone is publishing apps as shared services that can be integrated into your myApps client. This article explains the necessary steps.

[[image:shared-service-overview.png]]
== Purpose ==
At the time of writing, you have the option of connecting to two services that innovaphone provides as a shared service.
* Community Connect: You can connect to our Connect instance of our [[Howto:Community_Platform|community platform.]]
* Quotations Calculator: You can connect to our Quotations Calculator to submit projects and calculate prices.

=== Features ===
*Integrate all published shared services to any myApps client.

== Requirements ==
* Your system (PBX and AP) has to be at least version 15.
* You need to set up an H323 Federation as we explain in [[Howto13r3:Step-by-Step_Open_H.323_Federation|this wiki article.]]
** You can skip the DNS configuration step, because innovaphone provides all necessary SRV and A records on our DNS server.

* Your PBX needs to have a valid certificate.
** This means that your device certificate was signed by a well known Root CA.
** The CN or DNS parameter of the certificate must match the domain (system name) of the box.
***Alternatively, the SRV record target (DNS name) for H.323 federation must be present as a DNS entry in the certificate. (e.g. needed for Let's Encrypt created certificates)

== Configuration ==
Please read the requirements before you proceed.
* Set up an H323 Federation.
* A valid certificate is mandatory.
===== Shared Services config in the PBX =====

Go to PBX/Config/myApps and check that Directory Service URL in the Section Shared Services is pointing to ''https://shared-services.innovaphone.com''

[[image:shared-service-1.png]]
===== Settings Plugin =====
Go to the Shared Services plugin in your Settings App and click on Add Service.
* Press the question mark icon. A website will open. (At the time of writing this website is only available in German)
* Select the App you want to Install and click on ''Jetzt installieren''.
* Configure an ''App Long Name'' (e.g. Community Connect). This Long Name has to be unique in your PBX. This is the name displayed for all users.
* Configure an ''App Name'' (e.g. community-connect). This name has to be unique in your PBX as well. Do not use any special characters or spaces in this name. Use only lower case letters as explained in the [[Reference15r1:PBX/Objects#General_Object_Properties| name section of this wiki article.]]
* Click on OK

[[image:shared-service-2.png]]

===== Distribute Apps via Template =====
Open your Templates settings plugin and select the config template you want to use to distribute the App to users. In the Apps section, click on the app you have created and then click OK.

== Verification ==
Go to the All Apps area in your myApps client and open the app you created. You will see the content of the app. In the case of the Community Connect app, you will see the latest discussions.

== Known issues ==
=== Wrong Destination ===
If you receive this error message in your settings plugin, your H323 Federation is not configured correctly. Please check the configuration steps of our [[Howto13r3:Step-by-Step_Open_H.323_Federation|wiki article]].

=== URL missing ===
If you receive this error message in your settings plugin, your certificate was not accepted by us. Please check the requirements above.

== Related Articles ==
[[Howto13r3:Step-by-Step_Open_H.323_Federation]] 
[[Howto:Community_Platform]]