The display of the gateway’s configurable interfaces is organised 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 "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 "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 "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 international code (for example, 49).
- Subscriber Number: The local network number (for example, 7031).
- 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 XPARENT 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 sendt via the B-channel - then the codec XPARENT will be in use.
- 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.
- SC: 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.
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 "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 "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 signalled. 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 "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 dialled 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 email@example.com).
- 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).
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:
(IP of the SIP provider)
(PBX user object)
(IP of the PBX)
|H.323 name:E.164 no.
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 dialling of the call number 807031730090 therefore initiates a call at innovaphone AG via the configured SIP carrier.
Virtual interfaces (TEST, TONE, HTTP)
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 dialling 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 dialled digits have been analysed. In the meantime, the dial tone is played via the TONE interface. Suffix dialling 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.
Conferencing interface (CONF)
This CONF interface is only available on IP6000, IP800 and IP305. It is used to create a conferencing unit. So that up to 60 (IP6000), up to 10 (IP800) or up to 4 (IP305) subscribers can be in a conference. If a call is received on this interface there must be dialed a conference Id (0-9) and a '#' as suffix. Every call what goes to this CONF interface takes one DSP resource.
*note* Only IP6000, IP800 and IP305 support local conferencing, one DSP is in use for each user in a conference call.
*note* Some early IP6000 do not support this feature. Refer to the upgrade details if you consider upgrading your hardware.
For every interface, it is possible to store so-called CGPN In, CDPN In, CGPN Out and CDPN Out mappings (Calling Party Number In, Called Party Number In, Calling Party Number Out, Called Party Number Out), enabling call numbers and call number formats to be adjusted for incoming and outgoing calls. The call number formats are as follows:
|Unknown:||Unspecified. Number called in outgoing calls.||u|
|Subscriber:||Call number in local network. Number called in incoming calls.||s|
|National:||Call number with area code. Calling number from home country.||n||0|
|International:||Call number with country code and area code. Calling number from abroad.||i||00|
Clicking the link + or a mapping already created (for example, n->0) opens a popup page, on which the setting for the CGPN In, CDPN In, CGPN Out and CDPN Out mappings can be made:
- CGPN In: Is used to process the calling number of incoming calls.
- CDPN In: Is used to process the called number of incoming calls.
- CGPN Out: Is used to process the calling number of outgoing calls.
- CDPN Out: Is used to process the called number of outgoing calls.
Each mapping can be specified for a particular call number type:
- Unknown: The mapping applies to unknown, external calls.
- ISDN: The mapping applies to external calls.
- Private: The mapping applies to internal calls.