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 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 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 firstname.lastname@example.org).
- 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)
The CONF interface is currently available on IP6010, IP3010, IP1060, IP810, IP0010, IP6000, IP800 and IP305 (see datasheet of your product for up-to-date information). 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 DSP resource (that is, the PCM mode for calls from a physical interface such as ISDN to the CONF interface is not supported currently). 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.
The behaviour of the interface is controlled by various call-setup and in-call commands as follows.
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 unqiue room number: *1
This command creates a new conference room with a new, unique id. The syntax is
Valid Options are
<channel-max> 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
*1<channel-max> option correctly. If you omit it, the CONF interface will accept more
*2<channel-reserve> reservations than it is capable to satisfy.
<channel-reserve> is used to indicate the minimum number of conference channels which shall be available in the new conference room. Only 2 digits may follow
*2 so that a maximum of 99 channels is possible. If the
*2 option is used and 2 digits follow, the whole command is considered finished and the trailing
# may thus be left out. Please note that due to this, the
*2 option should be the last option used.
<prefix> 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
*3 option, or - as a shorthand notation - directly following the initial
*1 command introducer.
<match> 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.
<id-length> specifies the length of the random part of the created room number. It defaults to 6 if not specified.
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.
*1*310*124*26#) requests the creation of a new conference room with
10 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
*1 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 (
*2) 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 <prefix>. It is not possible to create the conference room if its id conflicts with a room that currently exists in this interface.
*2*310*124*26#) requests the creation of a new conference room with
10 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.
*3*310*124*26#) requests the join to a conference room with
10 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
*, the remaining digits are interpreted as conference room id and the call will join this conference if it exists.
The dialled digits are used to find an existing room. The first room number which matches is joined (for example, if the called party number is
10 and there is a conference room with id
1). 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
# 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 behaviour 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.
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#
Normal mode, all members are un-muted.
These events are recognized if the interworking option for the conference route is activated:
If the call is hold by the member, the conference call is muted.
If the call is retrieved again by the member, the conference call is activated.
One DSP channel is in use for each user in a conference call. 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.