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This field is used to enable the Emergency call routing for H.323 visiting users feature.
The Emergency call routing for H.323 visiting users feature support crisis alert for H.323 visiting users. Emergency calls made by H.323 visiting users from an enterprise network are routed to the nearest possible emergency response agencies (PSAPs) at the location visited by the user.
This field is available only for SIP signaling groups.
Valid entry |
Usage |
|---|---|
y |
The attendant endpoints are notified for the incoming emergency call. |
n |
The attendant endpoints are not notified for the incoming emergency call. The default value is n. |
Available only if Group Type is sip.
You must the LAR field to a value other than none. If you set the LAR field to none, the call drops.
Valid Entry |
Usage |
|---|---|
2 to 30 |
The time in seconds Communication Manager waits for a Ringing message from the far end of the trunk before trying for an alternate route to establish a call. The default value is 6 seconds. |
Available only if Group Type field is isdn-pri.
Valid Entry |
Usage |
|---|---|
y |
Enables associated signaling. |
n |
Enables non-facility associated signaling. |
Available only if Group Type is h.323 or sip.
Valid Entry |
Usage |
|---|---|
y |
Automatically remove from service the trunks assigned to this signaling group when IP transport performance falls below administered system-wide limits. |
n |
Signaling group trunks stay in service when IP transport performance falls below limits. |
Available only if Group Type is h.323 or sip.
Valid Entry |
Usage |
|---|---|
y |
Enables inter-connection between servers running Avaya Communication Manager. Unavailable if a RAS-Location Request (LRQ) is enabled.
Note:
When the near and far-end servers are running Avaya Communication Manager, this value must match the value administered for enabling the Layer 3 test. |
n |
Disables inter-connection between servers running Avaya Communication Manager. Used if the local or remote server is not running Avaya Communication Manager. |
Available only if Group Type is atm.
Valid Entry |
Usage |
|---|---|
T1 |
U.S., Canadian, and Japanese digital transmission format. |
E1 |
European digital transmission format. |
Valid Entry |
Usage |
|---|---|
y |
Allows the signaling group to administer Cluster ID. When it is set to y, the Far-end Node Name field is replaced with Cluster ID field. |
n |
Does not allow the signaling group to administer Cluster ID.. |
To control communications at layers 2 and 3 of the ISDN-PRI protocol, this field specifies what is on the far end of the link.
Available only if Group Type is atm.
Valid Entry |
Usage |
|---|---|
host |
The link connects Communication Manager to a computer. |
network |
The link connects Communication Manager to a local telephone company central office or any other public network switch. |
pbx |
The link is connected to another switch in a private network. |
The country protocol used by the local telephone company central office where the link terminates. This value must match the country protocol used by the far-end server. For connections to a public network, the network service provider can tell you which country protocol they are using.
Available only if Group Type is atm.
Valid Entry |
Usage |
|---|---|
1 to 25 |
The country code. |
etsi |
The network service provider uses the protocol of the European Telecommunications Standards Institute (ETSI). Used only if the Signaling Mode is isdn-pri. |
Sets the information transfer capability in a bearer capability IE of a setup message to speech or 3.1kHz. Available only if Group Type is atm or h.323.
Valid Entry |
Usage |
|---|---|
3.1kHz |
Provides 3.1kHz audio encoding in the information transfer capability. This is the default. |
speech |
Provides speech encoding in the information transfer capability. |
Allows or disallows direct audio connections between H.323 endpoints. Direct audio connections save bandwidth resources and improve sound quality of voice over IP (VoIP) transmissions. For SIP trunk groups, allows direct audio connections between SIP endpoints. Available only if Group Type is h.323 or sip.
For the SIP Resiliency feature, ensure that the Direct IP-IP Audio Connections field is set to Y.
The DTMF over IP field specifies the touchtone signals that are used for dual-tone multifrequency (DTMF) telephone signaling. The DTMF over IP field is available if the Group Type field is set to h.323 or sip.
Valid Entry |
Usage |
|---|---|
in-band |
All G711 and G729 calls pass DTMF in-band. DTMF digits encoded within existing RTP media stream for G.711/G.729 calls. G.723 is sent out-of-band. It also supports the SIP trunks. |
in-band-g711 |
Only G711 calls pass DTMF in-band. The system displays the in-band-g711 option if the Group Type field is set to h.323. |
out-of-band |
All IP calls pass DTMF out-of-band. For IP trunks, the digits are done with either Keypad IEs or H245 indications. This is the default for newly added H.323 signaling groups. The out-of-band option enables out of band signaling for the SIP Signaling group. When you select the out-of-band option, Communication Manager sends all outgoing DTMF messages as SIP INFO messages over a SIP signaling group. This option is interoperable with H.323 networks. You can connect an Avaya non-SIP endpoint or trunk to a voice mail system on the H.323 network that is integrated with Session Manager and Communication Manager through SIP. Examples of Avaya non-SIP endpoints or trunks are H.323, Analog, DCP, and Digital trunks. |
rtp-payload |
This is the method specified by RFC 2833. This is the default value for newly added SIP signaling groups. Support for SIP trunks requires the default entry of rtp-payload. |
Enables or disables a Layer 3 test. Communication Manager runs the Layer 3 test to verify that all connections known at the near-end are recognized at the far-end. Available only if Group Type is H.323 and SIP. This field is enabled by default and is in read-only mode.
The Layer 3 test should be enabled for an H.323 signaling group type when the Calls Share IP Signaling Connection is enabled and the far-end is Communication Manager. Requires administration of a Far-end Node Name.
The Layer 3 Test must be enabled for the call reconstruction.
The system displays this when the Group Type field is sip. Use this field to enable or disable Communication Manager to enforce SIPS URI for any incoming SIP message with security descriptions (used for SRTP negotiation) in SDP.
Valid Entry |
Usage |
|---|---|
y |
Enable Communication Manager to enforce SIPS URI for any incoming SIP message with security descriptions (used for SRTP negotiation) in SDP. Note: The system displays y as the default value for the Enforce SIPS URI for SRTP field. |
n |
Disable Communication Manager from enforcing SIPS URI for any incoming SIP message with security descriptions (used for SRTP negotiation) in SDP. For the SIP Resiliency feature, you must set this field to n if the SIP signaling is not supported. |
Available only if Group Type is isdn-pri and TSC Supplementary Service Protocol is set to c for ETSI.
Valid Entry |
Usage |
|---|---|
none |
Interface supports neither incoming nor outgoing ETSI CCBS. This is the default. |
incoming |
Interface supports only incoming ETSI CCBS. |
outgoing |
Interface supports only outgoing ETSI CCBS. |
both directions |
Interface supports incoming and outgoing ETSI CCBS.
Note:
When upgrading from a version of Communication Manager that is earlier than 5.1, this is the default. |
The name of the IP domain that is assigned to the far-end of the signaling group. For example, to route Session Manager calls within an enterprise, the domain assigned to the proxy server is used. For external SIP calling, the domain name could be that of the SIP service provider. Available only if Group Type is sip.
Valid Entry |
Usage |
|---|---|
Character string |
A maximum of 40 characters used to define the name of the IP domain for which the far-end proxy is responsible, if different than the near-end domain. |
blank |
Far-end domain is unspecified and the far-end IP address is used. Use if the domains are the same. |
Available only if Group Type is h.323 or sip.
Valid Entry |
Usage |
|---|---|
1 to 65535 |
Must match the number administered for Near-end Listen Port. Typically, this is the default of 5061 for SIP over TLS. |
blank |
Far-end listen port is unspecified. |
Available only if Group Type is h.323 or sip.
Valid Entry |
Usage |
|---|---|
1 to 2000 |
The number of the network region that is assigned to the far-end of the signaling group. The region is used to obtain the codec set used for negotiation of trunk bearer capability. |
blank |
Selects the region of the near-end node and the far-end network region is unspecified. |
Use this field to specify the node name for the far-end IP interface that is used for trunks assigned to the Signaling group. You must administer the node name on the Node Names screen.
The system displays the field only if the Group Type field is set to h.323 or sip.
The signaling group number.
The type of protocol used with the signaling group.
Valid Entry |
Usage |
|---|---|
h.323 |
h.323 protocols or SBS signaling trunks |
isdn-pri |
Integrated Service Digital Network Primary Rate Interface |
sip |
For SIP |
Enables or disables the requirement that the Communication Manager server uses the H.235 Annex H (now called H.235.5) protocol for authentication during registration. Available only if Group Type is h.323. Requires that RAS-Location Request (LRQ) is enabled.
Available only if DTMF over IP is set to out-of-band.
Valid Entry |
Usage |
|---|---|
80 to 350 blank |
Specifies the duration of DTMF tones sent in H.245-signal messages. The default is blank. |
Available if Group Type is h.323 and Direct IP-IP Audio Connections is enabled. Also available if Group Type is SIP and Direct IP-IP Audio Connections is enabled.
Valid Entry |
Usage |
|---|---|
y |
A call from an H.323 station over a trunk that uses this signaling group starts as a direct media call. The IP address and port of the H.323 station are sent as the media and media control channel addresses in the SETUP/INVITE message.
Note:
On an outgoing Direct Media call from an IP (H.323) telephone over this trunk, if an attempt is made to transfer the call, conference another party, or put the call on hold while the call is still in ringing state, the operation fails. |
n |
The IP address of the MEDPRO board is sent in the SETUP/INVITE message. This is the default. |
An eight-digit string that is compatible with the protocol used by the far-end switch or server. Sets the signal sent out over idle DS0 channels.
Available only if Group Type is atm.
Enables or disables accepting SIP requests that match the domain in the Far-End Domain. This field enables the signaling group to use the IMS signaling. Available only if Group Type is sip. If you are configuring Communication Manager as a feature server, set the IMS Enabled field to y. Setting this field to y only applies when Communication Manager is being configured as a feature server. If you are configuring Communication Manager as an evolution server, leave the IMS Enabled field set to n.
The system displays this on the Signaling Group screen when the Group Type field is sip.
Valid entries |
Usage |
|---|---|
allow |
Communication Manager software connects the call and allows the SIP trunks to remain in the looparound connection. If the trunk group controlled by this SIP signaling group is used for IGAR calls, set the value allow. |
eliminate |
Communication Manager software connects the call and eliminates the SIP trunks from the looparound connection. The default value is eliminated. |
Valid Entry |
Usage |
|---|---|
n |
Disables the SIP direct media feature. The default value is n. |
y |
Enables the SIP direct media feature and SIP-H.323 direct media feature. |
If Initial IP- IP Direct Media? is set to y in the SIP Signaling Group then Initial INVITE with SDP for secure calls? must also be set to y in the system parameter features forms.
For the SIP Resiliency feature, ensure that the Initial IP- IP Direct Media? and the Initial INVITE with SDP for secure calls? field must be set to y.
Controls how your server negotiates glare with the far-end switch. Available only for a far-end PBX connection.
Valid entry |
Usage |
|---|---|
U.S. private network applications |
|
network |
The server overrides the other end when glare occurs. For servers connected to a host computer. |
user |
The server releases the contested circuit and looks for another when glare occurs. For servers connected to a public network. |
Private networks, including QSIG networks, applications outside the U.S. |
|
peer-master |
The switch overrides the other end when glare occurs. |
peer-slave |
The switch releases the contested circuit and looks for another when glare occurs. |
The companding algorithm expected by the system at the far end.
Valid Entry |
Usage |
|---|---|
a-law |
Algorithm expected at the far-end for E1 service. |
mu-law |
Algorithm expected at the far-end for T1 service. |
Determines what message Communication Manager sends when an incoming ISDN trunk call is routed over a non-ISDN trunk group.
Valid Entry |
Usage |
|---|---|
PROGress |
Requests the public network to cut through the B-channel and let the caller hear tones such as ringback or busy tone provided over the non-ISDN trunk. Normally-selected value. |
ALERTing |
Causes the public network in many countries to play ringback tone to the caller. This value is used only if the DS1 is connected to the public network, and it is determined that callers hear silence (rather than ringback or busy tone) when a call incoming over the DS1 interworks to a non-ISDN trunk. |
Enables or disables IP video capability for this signaling group. Available only if the signaling group type h.323 and sip.
Specifies how long to hold the call state information in the event of an IP network failure or disruption. Communication Manager preserves calls and starts this timer at the onset of network disruption (signaling socket failure). If the signaling channel recovers before the timer expires, all call state information is preserved and the signaling channel is recovered. If the signaling channel does not recover before the timer expires, the system:
Raises an alarm against the signaling channel
Maintains all connections with the signaling channel
Discards all call state information about the signaling channel
Valid Entry |
Usage |
|---|---|
1 to 180 |
The number of seconds to delay the reaction of the call controller to a link bounce. Default is 90. |
The system displays this only when the Group Type field on the Signaling Group screen is h.323 and Multiple Locations field on the Optional Feature screen is y.
Valid Entry |
Usage |
|---|---|
1 to 2000 |
(Depending on your server configuration, see Avaya Aura® Communication Manager System Capacities Table.) Determines the location-based digit analysis which is applied to the destination digits dialed by the incoming trunk call. |
blank |
The location of the Processor Ethernet (PE) is taken for routing the call. By default, the value is blank. |
Enables or disables RAS-Location Request (LRQ) messages. Available only if Group Type is h.323.
Valid Entry |
Usage |
|---|---|
y |
Allows IP trunk availability to be determined on a per call basis. An LRQ message is sent to the far-end gatekeeper prior to each call over the IP trunk. The far-end gatekeeper responds with a RAS-Location Confirm (LCF) message, and the call proceeds. Required if H.235 Annex H Required is enabled. Calls Share IP Signaling Connection must be disabled. |
n |
For far-end servers running Communication Manager, unless H.235 Annex H Required is enabled. |
Available only if Group Type is atm, h.323, or isdn-pri.
Valid Entry |
Usage |
|---|---|
0 to 619 |
The maximum number of simultaneous call-associated Temporary Signaling Connections that can exist in the signaling group. Typically this is the number of ISDN-PRI trunk group members controlled by this signaling group. |
Available only if Group Type is atm, h.323, or isdn-pri.
Valid Entry |
Usage |
|---|---|
0 to 256 |
The maximum number of simultaneous non-call-associated Temporary Signaling Connections. The TSCs carry signaling for features not associated with a specific call, for example, signals to turn on Leave Word Calling. |
Enables or disables encryption for trunk calls assigned to this signaling group. If encryption for the signaling group is disabled, then trunk calls using this signaling group do not get encrypted regardless of IP Codec Set administration. Available only if Media Encryption is enabled in Communication Manager and the Group Type is h.323.
A name that identifies the signaling group. Accepts up to 15 alphanumeric characters.
Avaya BRI deskphones support only ASCII characters because non-ASCII characters, such as Eurofont and Kanafont, show up incorrectly.
Available only if Group Type is h.323 or sip.
Valid Entry |
Usage |
|---|---|
5000 to 9999 |
An unused port number for the near-end listen port. |
1719 |
For LRQ-enabled ports |
1720 |
For h.323-enabled ports |
Available only if Group Type is h.323 or sip. This field can be set to procr or procr6, for IPv4 and IPv6 signaling.
Enables or disables Network Call Transfer so that D-channels support Explicit Network Call Transfer (ENCT). Available only if Group Type is atm.
The passphrase used to generate a shared secret
for symmetric encryption of the media session key. The same passphrase must be assigned to the corresponding signaling groups at both ends of an IP trunk. The passphrase:
Consists of 8 to 30 alphanumeric characters
Is case sensitive
Must contain at least one alphabetic and at least one numeric
Valid characters also include letters, numbers, and these symbols: !&*?;’^(),.:-
The passphrase is used for both Media Encryption and authentication. This field cannot be left blank.
Available only if Media Encryption is enabled or the H.235 Annex H Required is enabled.
The values for this field are n(o) and y(es). The default value is y(es). This field is only available if Group Type is sip. When set to y(es) then Communication Manager will automatically set the value in the Peer Server field when the sip signaling group comes into service.
The values for this field are Others, SM, and AMS. The default value is Others. This field is only available if Group Type is sip. If this sip signaling group terminates to an Avaya Session Manager, then Communication Manager will automatically set value in this field to SM if the Peer Detection Enabled field is set to y(es).
If Clustered field is set to y, then the value in this field is automatically changed to SM, and this field will be in read-only mode.
If you set the Prepend '+' to Outgoing Calling/Alerting/Diverting/Connected Public Numbers field on the SIP Signaling Group screen to y and if Communication Manager uses the public-unknown-numbering table, Communication Manager inserts a leading plus (+) digit in calling, alerting, diverting, or connected numbers. The default value is n.
If the Peer Server field on the SIP Signaling Group screen is SM, the system sets the Prepend '+' to Outgoing Calling/Alerting/Diverting/Connected Public Numbers field to display-only, with the default value of y. If the Peer Server field on the SIP Signaling Group screen is Others, the system sets the Prepend '+' to Outgoing Calling/Alerting/Diverting/Connected Public Numbers field to n and as editable.
Available only if Group Type is isdn-pri.
Valid Entry |
Usage |
|---|---|
1 to 999 |
First to third are the media gateway number. |
V |
Fourth character is the board. |
1 to 8 |
Fifth character is the board number. |
01 to 04 (Analog TIE trunks) 09 to 32 |
Six and seventh characters are the circuit number. |
xxxVmpp |
Specifies the Branch Gateway.
|
Enables or disables the priority video function that specifies that incoming video calls have an increased likelihood of receiving bandwidth, and are also allocated a larger maximum bandwidth per call.
Available only if:
Group Type is h.323 or sip
Multimedia SIP trunking is enabled for the system
Available only if Group Type is atm.
Valid Entry |
Usage |
|---|---|
a b c d |
In countries whose public networks allow multiple layer-3 signaling protocols for ISDN-PRI service. Selects the protocol that matches the network service provider’s protocol. |
The system displays this only when the Group Type field is h.323 or sip.
Valid Entry |
Usage |
|---|---|
y |
Enables the QSIG over SIP (Q-SIP) feature for the signaling group. |
n |
The QSIG over SIP feature is disabled. By default, the value is n. |
The system displays this only when the Group Type field is sip and the Q-SIP field is set to y.
Valid Entry |
Usage |
|---|---|
1 to 999 |
Assigns a number for the QSIG signaling group. |
blank |
No QSIG signaling group is assigned. By default, the value is blank. |
If you set the Remove '+' from Incoming Called/Calling/Alerting/Diverting/Connected Numbers field on the SIP Signaling Group screen to y, Communication Manager removes the leading plus (+) digit, if present, from both calling and called numbers. The default value is y.
This field interacts with the Peer Server field in the following manner:
If the Peer Server field on the SIP Signaling Group screen is SM, the system sets the Remove '+' from Incoming Called/Calling/Alerting/Diverting/Connected Numbers field to n. The Remove '+' from Incoming Called/Calling/Alerting/Diverting/Connected Numbers field is editable.
If the Peer Server field on the SIP Signaling Group screen is Others, the system sets the Remove '+' from Incoming Called/Calling/Alerting/Diverting/Connected Numbers field to y. The Remove '+' from Incoming Called/Calling/Alerting/Diverting/Connected Numbers field is editable.
Enables or disables administering this signaling group for a remote office. Available only if Group Type is h.323.
The system displays this on the Signaling Group screen when the Group Type field is sip.
Valid entries |
Usage |
|---|---|
y |
This enables SIP signaling for comfort noise. If RFC 3389 Comfort Noise field is set to y, this field overrides the Silence Suppression field on the IP Codec Set screen. |
n |
This disables SIP signaling for comfort noise. Default is n. |
Enables or disables the requirement that a vendor registration be sent. Available only if Group Type is h.323.
Enables or disables the Separation of Bearer and Signaling (SBS) trunk groups. If this field is enabled, both the Trunk Group for NCA TSC and the Trunk Group for Channel Selection must be set to the signaling group number administered for the SBS trunk group. Available only if Group Type is set to h.323.
Use this field to specify the time in minutes that Communication Manager waits before tearing down a ring no answer call. You can enter a value from 3 to 240. The default value is 3.
The system displays this field only when the Group Type field is set to sip.
Displays the isdn-pri signaling mode. Available only if Group Type is atm.
The system displays this only when the Group Type field is h.323 and the Q-SIP field is set to y.
Valid Entry |
Usage |
|---|---|
1 to 999 |
Assigns a number for the SIP signaling group. |
blank |
No SIP signaling group is assigned. By default, the value is blank. |
Available only if Group Type is h.323.
Valid Entry |
Usage |
|---|---|
2 to 10 |
The number of seconds the system waits for a response from the far end before invoking Look Ahead Routing. Default is 10. |
Available only if Group Type is sip.
Valid Entry |
Usage |
|---|---|
tcp |
Transport is accomplished using Transmission Control Protocol (TCP). |
tls |
Transport is accomplished using Transport Layer Security (TLS). This is the default. For the SIP Resiliency feature, ensure that the TLS is selected as a transport method. |
Available only if Group Type is atm, h.323, or isdn-pri.
Valid Entry |
Usage |
|---|---|
1 to 2000 |
Trunk group number used for channel selection. |
Available only if Group Type is atm, h.323, or isdn-pri.
Valid Entry |
Usage |
|---|---|
1 to 2000 blank |
Trunk group number used for non-call-associated Temporary Signaling Connections. |
The supplementary service protocol used for temporary signaling connections.
Available only if Group Type is atm, h.323, or isdn-pri.
Valid Entry |
Usage |
|---|---|
a |
AT&T, Telcordia Technologies, Nortel. |
b |
ISO QSIG. Also, for SBS signaling groups. |
c |
ETSI. Available only if Group Type is isdn-pri. |
d |
ECMA QSIG |
e |
Allows DCS with rerouting. DCS with Rerouting must be enabled, and Used for DCS must be enabled for the trunk group. |
f |
Feature Plus |
g |
ANSI. Available only if ISDN-PRI, or ISDN-BRI, or Used for DCS are enabled for the system. |
Available only if Group Type is atm.
Valid Entry |
Usage |
|---|---|
32 to 1023 blank |
The number used as a Virtual Channel Identifier. |
Available only if Group Type is atm.
Valid Entry |
Usage |
|---|---|
32 to 1023 blank |
The number used as a Virtual Path Identifier. |
The type of X-Mobile endpoints allowed. Available only if Group Type is isdn-pri.
Valid Entry |
Usage |
|---|---|
DECT |
The remote end of the trunk group controlled by the signaling group is a DECT mobility controller. This allows X-Mobility to work over ISDN-PRI trunks between the server/switch and adjunct. |
none |
Not administered. |