SIP Gateways

Telephony gateway data

Name

Select a unique Name for the SIP Gateway.

Gateway type

Use the drop down list to specify the interface of the telephone system. Only the gateways mentioned in this list are supported. Contact your XPhone Connect partner if you have any questions.

Dialling parameter

Select a Dialling parameter from the dropdown list. The dialling parameter must contain the main numbers that are set up in the PBX.

Deactivated

If desired, the SIP Gateway can be disabled by selecting the checkbox. All lines connected to this node are then inactive.

XPhone Call Controller (XCC) Gateway

SIP connection XCC <-> PBX

  • Enter the IP address and the Port of the server on which the XPhone call controller was installed.

  • Enter the IP address and the Port of the gateway (PBX).

    Attention

    If the port 5060 is already in use on the XPhone Connect Server (localhost) (e.g. for XCAPI switch-on), you must configure the SIP connections on a different port in the used PBX. If this is not possible, you must allocate another IP address to the XPhone Connect Server, which you can enter instead of localhost.

  • Select the Protocol used (default: UDP).

  • Select an option and/or CLIP no screening.

    • This parameter has an influence on how the signalled phone numbers are shown on voice end devices in combination with AnyDevice.

    • The CLIP No Screening feature must have been activated for the company by the landline provider and be supported by your voice communication system (PBX).

      Your selection determines how the Calling Line Identification (CLI) is transferred via SIP:

      • Remote Party ID (see SIP Extensions for Caller Identity and Privacy)

      • P-Asserted-Identity (see RFC 3325)

      • Off: The phone number is not transferred to a special SIP element

  • Define the Codecs to be used which are negotiated between XCC and PBX and check first which Codecs are supported by the connected PBX. The order of the Codecs is also top-down prioritisation, whereby the highest Codec has top priority.

    The following Codecs are available:

    • Opus (Default)

    • G.722

    • G.711 A-law (Europe)

    • G.711 µ-law (North america, Japan)

Ethernet Adapter for SIP-Logging

For the analysis of errors on SIP level, the network interface must be selected here, which is used for the communication to the PBX or SBC.

  • For SIP logging, select the adapter here that is used for the connection from the XPhone Server to the PBX/SBC.

  • Network Driver Interface Specification WAN Adapters (NDIS WAN) in the drop-down lists are virtual machine interfaces and can be ignored.

  • If several adapters are displayed and there is no clear assignment, it is possible to find out the adapter name (as used in the operating system) via the registry.

  • Navigate in the registry (Regedit.exe) to HKEY_LOCAL_MACHINE\SYSTEM\CurrentControlSet\Control\Network and search there for the value (NPF_{XXXXXXXX-XXXX-, etc..}) which is displayed in the selection list of the XPhone Server. Under this entry in the key “Connection” you will find the corresponding name of the adapter (e.g.: Ethernet 1).

Verify

After the SIP Gateway has been restarted, enable SIP Gateway logging and make a test call. Open the *.pcap files and check if these files have any content.

SIP connection XCC <-> XPhone Connect Server (AnyDevice/softphone)

  • Activate the connection between XPhone Call Controller and XPhone Connect Server to also be able to use the Softphone function with this SIP gateway.

  • Enter the IP address and the Port of the XPhone Call Controller.

  • Enter the IP address and the Port of the XPhone Connect Server.

    Attention

    The ports entered here must be free for use in the network.

  • The protocol used is TCP by default and cannot be changed.

  • Use a STUN server if you want to use Softphone Mobile or Softphone Desktop with payload separation.

    Attention

    If a STUN server is configured for the first time under System settings > Telephony & Meetings > Network, the XCC must be restarted afterwards under System settings > Telephony & Meetings > Telephony > SIP > XCC via Diagnosis. Otherwise, the XCC cannot determine its external IP and thus no media data connection can be established.

LoopBack adapter for SIP logging

For the analysis of an error on SIP level, between the XPhone Server and the XPhoneCallController, a LoopBack Adapter is required, because both (XPS and XCC) are located on the same computer.

  • Select the LoopBack adapter for SIP logging here (e.g.: NPF_Loopback - Adapter for loopback traffic capture or NPCpl Loopback Adapter).

  • Network Driver Interface Specification WAN Adapters (NDIS WAN) in the drop-down lists are virtual machine interfaces and can be ignored.

Verify

After the SIP Gateway has been restarted, enable logging of the SIP Gateway and make a test call. Open the *.pcap files and check if these files have any content.

AnyDevice / Softphone

To enable all users to use a standardised number for forwarding to AnyDevice, the telephone system used must support communication of the forwarding user’s phone number in the SIP Diversion Header. This is then used to determine the forwarding number via a regular printout. To do this, activate this check box: Standardised telephone number ….

  • Telephone number for forwarding:

    Enter the telephone number for forwarding; this telephone number must be configured in the telephone system and firmly routed to the XCC. If a user later activates the AnyDevice function on his XPhone Connect Client, his extension is automatically forwarded to the telephone number entered here.

Attention

Note that the call number length of the AnyDevice number entered in this field must be taken into account in the CTI dialing parameter in the length of internal call number field. For call forwarding, the CTI dialing parameter under System settings > Dialing parameter used by the user’s line is generally applied and not the dialing parameter assigned to SIP Gateway. If you select too small a value in the length of internal number field, this will result in incorrect displays on the Connect Client and incorrect call forwarding.

Meeting service

  1. You can enable dial-in conferences here by activating the checkbox.

  2. Then, allocate an extension number that can be dialled externally and internally, and firmly allocate it to a language.

    If you want to configure another telephone number, click on the add-symbol; to delete a telephone number, click on the remove-symbol .

  3. The order of the telephone numbers corresponds to the list in the e-mail generated when a conference is set up. Keep in mind that appropriate dialling policies must be established in the PBX for these telephone numbers so that calls can be treated accordingly (see also: Telephone number plans and routing).

    The following languages are available for dial-in conferences:

    • German

    • English

    • French

    • Italian

  4. To apply all settings, click Save.

Hint

You can also find the telephone numbers entered here in the invitation e-mail for conferences. This telephone number is then shown to the participants to allow them to dial into a conference.

TeamDesk hotline

  1. Add a new line for a TeamDesk-Group.

  2. Select a queue for a group previously created in Teamdesk.

  3. Allocate a number in E.164 format which can be dialled externally and internally and assign it to a specific queue.

  4. Please note that the corresponding dialling rules for these numbers need to be created in the PBX to enable calls to be assigned to the corresponding queues.

  5. As an option, enter a comment on the queue.

  6. If you wish to configure another hotline, click the add-symbol; to delete a queue, click the remove- symbol after the corresponding line .

Advanced settings

Advanced settings

Caution

Only change these settings when advised to do so by product support.

Name

Description

Value

CreateLineVerify

Lines are verified before creation; i.e. invalid phone numbers are filtered. Type: Numerical decimal or hexadecimal (starts with 0x) value

0: Off 1: On (Default)

XccGatewayInPrefix

This parameter specifies the station number range at which AnyDevice participants are available from the voice gateway (PBX).

The numbers of the AnyDevice end devices in the XCC are identical to the extensions for the respective users in the voice gateway. The XccGatewayInPrefix is used to make the AnyDevices unique in the voice gateway number plan.

This parameter may only be used when the voice gateway does not provide any redirecting ID or the voice gateway displays (TAPI or CSTA) restrictions in the case of call deflection to forwarded end devices. See also: XccUseDirectRedirecting.

Concerns the AnyDevice feature - more information available under Phone number prefix

Type: Text

XccGlobalVirtualFollowMeUserNo

This parameter permits switching between a virtual port using the Follow-Me function to enable the “speaking” forwarding destination to be shown. Enter the number of a virtual port in your voice gateway here which is forwarded to the Follow-Me phone number on the XCC. Assign the virtual port a unique name (e.g. “Follow-Me”).

Concerns the AnyDevice feature - more information available under Virtual Follow-Me-User

Type: Text

XccUseDirectRedirecting

This parameter is required if the redirect/deflect function is run via call control interface (e.g. TAPI/CSTA) on a workstation device where the Follow-Me transfer is activated.

A range of voice gateways do not permit a forwarded port for call deflection and acknowledge this request as an error.

If the call deflection destination is an AnyDevice device (e.g. where the Follow-Me function is active), the XPhone Connect Server can also reach the AnyDevice device via the XccGatewayInPrefix.

Concerns the AnyDevice feature - more information available under Redirect/deflect feature

The value should be 1 if this mode is to be activated.

XccProfileName

Defines the name for an SIP trunk

Type: Text

XccDiversionPattern

It is possible that certain PBX configurations cause the redirecting ID (SIP DIVERSION header) to be transferred for external in E.164 and for internal as an extension.

Concerns the AnyDevice feature - more information: Redirecting ID Pattern (SIP DIVERSION header)

([+]?\d+)\D+.*$ usually applies for any PBX - more informations on adjusting the value: Redirecting ID Pattern (SIP DIVERSION header)

GwSetting_caller-id-type

This parameter influences the display of the signaled phone number on voice terminals in connection with AnyDevice (Clip No Screening).

The Clip No Screening feature must have been activated for the company by the landline provider and be supported by your PBX.

The GwSetting_caller-id-type parameter determines how the Calling Line Identification (CLI) is transferred via SIP.

Concerns the AnyDevice feature - more information available under Clip No Screening

rpid: Caller info via Remote-Party-ID Header (Default) - see ietf

pid: Caller Info via “p-Asserted-Identity Header” (see RFC 3325)

none: no caller information

ShowForwardBusy

TRUE shows the call forwarding menu for Busy” in Client > Device management > Advanced…. The default value is set at TRUE. Automatic call forwarding can be set up for internal and external calls when the AnyDevice is busy.

TRUE

FALSE

ShowForwardNoAnswer

TRUE shows the call forwarding menu for By time in Client > Device management > Advanced…. The default value is set at TRUE. Automatic call forwarding can be set up for internal and external calls when calls are not picked up after a certain time.

TRUE

FALSE

XccDtmfPayloadType

Sets the correct payload type at the XCC for the transmission of the DTMF signals between PBX and XCC. See DTMF analysis and troubleshooting

Type: Text

Click Apply to save the selected parameter with the set values or Cancel to discard the entries. You will return to the Modify SIP Gateway dialog. You can now add further additional parameters, modify already existing ones via Edit or delete parameters with Remove.

In the Change SIP Gateway dialog, finally click Save to apply all settings. If you click Cancel, you will leave the settings dialog without applying the changes.

Parameter sets for lines (advanced settings)

The parameter records for lines are optional and only required in special cases. If desired or if your company structure requires this, additional parameters can be specified depending on the PBX.

Clicking on Add will first take you to the Create parameter set for lines dialogue. You must enter a unique name and, as an option, a description here. To enter the parameters now, click Add again, enter the required parameter and enter a value in the Add new advanced settings dialogue.

Hint

You can obtain parameters and values from product support. They will find out together with you, which parameters are useful for the corresponding circumstances and which values will result in a solution to the issue.

Now click Apply to save the selected parameter with the set values or Cancel to discard the entries. You will return to the Create parameter sets for lines dialog. You can add further additional parameters, modify already existing ones via Edit or delete parameters with Remove. Click Apply again to save the selected parameters with the set values or Cancel to discard the entries and return to the Change SIP gateway dialog.

In the Change SIP Gateway dialog, finally click Save to apply all settings. If you click Cancel, you will leave the settings dialog without applying the changes.

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