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management-port

Section: TServer
Default Value: 0
Valid Values: 0 or any valid TCP/IP port
Changes Take Effect: After SIP Server restart


Specifies the TCP/IP port that management agents use to communicate with SIP Server. If set to 0 (zero), this port is not used.

http-port

Section: TServer
Default Value: 0
Valid Values: 0, 1024-65535
Changes Take Effect: After SIP Server restart


Specifies the HTTP interface port number. When set to 0, the HTTP server is disabled. The port numbers in the range of 1 through 1023 are the system ports and must not be used.

sip-port

Section: TServer
Default Value: 5060
Valid Values: Any valid TCP/IP port
Changes Take Effect: After SIP Server restart


Specifies the port on which SIP Server listens for incoming SIP requests. The same port number is used for both TCP and UDP transports.

sip-enable-x-genesys-route

Section: TServer
Default Value: false
Valid Values: true, false
Changes Take Effect: Immediately
Related Feature: SRV Address Support in Contact and Record-Route Headers

Specifies if SIP Server adds the private X-Genesys-Route header to SIP messages when deployed with SIP Proxy. This is for backward compatibility to disable new functionality in old deployments.

enable-strict-location-match

Section: TServer
Default Value: No default value (empty string)
Valid Values: msml (or true), softswitch, trunk, all
Changes Take Effect: For the next call
Related Feature: Geo-Location for MSML-Based Services: Strict Matching

Controls the SIP Server behavior in cases where an MSML service that matches a call by geo-location or overflow-location is not available, or, if during an attempt to apply a treatment, the matching service responds to the INVITE message with a SIP error, as follows:

  • If this option is not present or not configured, SIP Server tries other available services for a call.
  • If this option is set to msml (or true), SIP Server tries other available services that match a call by geo-location or overflow-location. If there is no match, SIP Server does not apply a service to the call with a different geo-location. (A value of true is supported for compatibility with previous releases of this feature.)
  • If this option is set to trunk or softswitch, SIP Server tries other available trunks or softswitches that match a call by geo-location. This applies to calls directed to an external destination or DNs located behind the softswitch. If there is no match, SIP Server does not send a call to a device with a different geo-location.
  • If this option is set to all, SIP Server applies the msml setting for calls to GVP and the trunk/softswitch setting to other cases.

If the enable-strict-location-match option is set to msml or true, it is possible to specify an alternative geo-location using the Application-level option overflow-location-map, or using the overflow-location key in AttributeExtensions of TRouteCall and TApplyTreatment client requests.

find-trunk-by-location

Section: TServer
Default Value: false
Valid Values: true, false
Changes Take Effect: For the next call


Determines SIP Server behavior for choosing a gateway or trunk for the outbound call.

If set to true, SIP Server considers the geo-location option setting when prioritizing the selection of an outbound gateway or trunk. When making a selection, SIP Server first narrows the pool of available in-service gateways or trunks based on the prefix match, then further narrows the pool by matching the value of the geo-location option of the DN to the value of thegeo-location option for the Trunk device. If more than one matching trunk is found, SIP Server can further narrow the selection by considering the value of the priority option (if it is configured). If no matching trunk is found, SIP Server selects a trunk as if find-trunk-by-location is set to false.

If set to false, SIP Server does not include the geo-location option setting when prioritizing the selection of an outbound gateway or trunk. SIP Server selects a gateway or trunk from the pool of all configured trunks that are in service based on the prefix match, disregarding the geo-location option (if the priority option is configured, it will still be considered). If there is more than one trunk in the pool, SIP Server chooses the trunk in a round-robin algorithm that provides equal gateway load. However, if an external party is transferred to an outbound destination, the same gateway that connected the external party to the call is used for the outbound transfer.

dial-plan

Section: TServer
Default Value: No default value
Valid Values: Any dial-plan Voice over IP Service DN
Changes Take Effect: For the next call
Related Feature: Dial Plan

Specifies which dial-plan DN will be applied to calls. You can define the option on any of the following locations listed in order of highest to lowest priority:

  1. Agent Login—Applies to calls made by a caller logged in under this Agent Login ID.
  2. DN-level—Applies to calls made from a DN (where an Agent Login dial-plan is undefined) or for inbound calls if the dial-plan is assigned to the Trunk DN.
  3. Application-level—Applies to all calls (where no Agent Login or DN dialplan is defined).

sip-enable-rfc3263

Section: TServer
Default Value: false
Valid Values: true, false
Changes Take Effect: After SIP Server restart
Related Feature: DNS Name Resolution

Specifies the DNS resolution mode.

If you set this option to true, SIP Server includes priority and weight information from the Returned Record Set (resolved from the contact option using the internal DNS) when it applies the destination selection procedure. This is in accordance with RFC 3263 and RFC 2782. If set to true, SIP Server ignores the values of the contacts-backup option as redundant.

If you set this option to false, SIP Server does not factor in priority or weight from the RR Set when applying the destination selection procedure (multiple active destinations are given equal ranking). The destinations in this case are taken from the URIs configured in the following DN options:

  • contact
  • contacts-backup (can be several URIs in a comma-separated list)

The Active Out-of-Service Detection procedure uses DNS SRV/A Resource Records resolution for composing a list of transports (protocol, IP address, port) for each contact’s URI. To use priority and weight, set this option to true. To treat all destinations as equal, set this option to false.

sip-enable-gdns

Section: TServer
Default Value: true
Valid Values: true, false
Changes Take Effect: After SIP Server restart


Specifies the DNS resolution mode. If you set this option to true, SIP Server uses its internal DNS client to connect to a DNS server available on the network to use its conversion services. If no DNS server is available, set this option to false. In this case, SIP Server resolves the domain names using local operating system utilities.

If set to false, SIP Server is unable to perform DNS resolution for SRV records with contacts that are missing port information (indicating the need to use SRV). Instead, ‘A’ record resolution and default ports will be used. The default port for UDP/TCP is 5060, while the default for TLS is 5061.

sip-outbound-proxy

Section: TServer
Default Value: false
Valid Values: true, false
Changes Take Effect: After SIP Server restart


If set to true, all SIP messages are sent through SIP Proxy. SIP Server looks for a VoIP Service DN with service-type=sip-outbound-proxy. For each initial out-of-dialog outgoing SIP request, SIP Server inserts a Route header with the value of the DN contact.

Note: SIP Server sends the REGISTER message directly to the Trunk DN configured with the force-register option, instead of SIP Proxy.

sip-address-srv

Section: TServer
Default Value: No default value
Valid Values: Valid Fully Qualified Domain Name (FQDN)
Changes Take Effect: After SIP Server restart


When specified, SIP Server can use this FQDN as its own contact for the DNS name resolution procedure.

sip-address

Section: TServer
Default Value: No default value
Valid Values: Any valid IP address or host name
Changes Take Effect: After SIP Server restart


Specifies an IP address of the SIP Server interface. This option must be set when deploying SIP Server on a host with multiple network interfaces. SIP Server uses this value to build the Via and the Contact headers in SIP messages. When this option is not set, SIP Server attempts to detect the IP address automatically.

geo-location

Section: TServer
Default Value: No default value
Valid Values: Any string
Changes Take Effect: For the next call


Specifies the data center to which this SIP Server instance belongs.

sip-link-type

Section: TServer
Default Value: 0
Valid Values: 0, 3, 4
Changes Take Effect: After SIP Server restart


Specifies whether SIP Server will run in multi-threaded mode, or in single-threaded mode for backward compatibility. Configure the valid values for this option as follows:

  • 0 (default)—SIP Server runs in single-threaded mode, as in pre-8.0.3 releases, with the Main thread processing T-Library requests, distributing Events, managing SIP calls, and processing SIP signaling
  • 1,2—Reserved for debugging
  • 3—Enables multi-threaded mode with the following threads:
    • T-Server thread—processes T-Library requests and distributes Events
    • Call Manager thread—manages SIP calls and processes SIP signaling (except OPTIONS messages)
    • Service Checker thread— performs Active Out-of-Service Detection (OPTIONS messages)
  • 4—Enables multi-threaded mode designed for IMS double-dip deployments, with the following threads:
    • T-Server thread
    • 16 Call Manager threads
    • Service Checker thread
    • Presence Manager thread

In both single-threaded and multi-threaded modes, SIP Server runs the following threads:

  • SIP transport layer thread to dispatch SIP messages
  • Operational Information thread to collect and report statistics; to perform NIC monitoring
  • A number of auxiliary threads

Important: For an HA configuration, this option must be set to the same value in both primary and backup SIP Servers.

server-role

Section: TServer
Default Value: 0
Valid Values: 0, 1, 5
Changes Take Effect: After SIP Server restart


Specifies the role that SIP Server plays in the deployment scenario:

  • 0—SIP Server runs in a standalone deployment, where the server is not integrated into an IMS environment (DNs are registered or provisioned on SIP Server).
  • 1—SIP Server runs as a SIP Application Server (SIP-AS) in an IMS deployment.
  • 5—SIP Server runs in cluster mode.

Configuring SIP Servers

You must configure SIP Server applications for the following purposes:

Configuring SIP Servers for SIP Cluster

  1. Deploy SIP Servers as an HA pair, Hot Standby redundancy mode, by following the standard procedure.
    • Suggested application names: SIPS_<datacenter>_1, SIPS_<datacenter>_1_B.
  2. On the Switches tab, add the SIP Cluster Switch object to each SIP Server application.
  3. On the Connections tab, add the following connections:
    • confserv_proxy_<datacenter>—Set to the following parameters:
      • Connection Protocol: addp
      • Trace Mode: Trace On Both Sides
      • Local Timeout: 60
      • Remote Timeout: 90
    • MessageServer_<datacenter>—Set to the following parameters:
      • Connection Protocol: addp
      • Trace Mode: Trace On Both Sides
      • Local Timeout: 7
      • Remote Timeout: 11
  4. On the Server Info tab, configure the following ports for each SIP Server application:
ID Listening Port Connection Protocol
default Any available port number
TCport Any available port number TController
IPport Any available port number IProxy
SmartProxy Any available port number SmartProxy
Note: Changes in port numbers take effect after SIP Server restart.

5. On the Options tab in the [TServer] section, configure the following mandatory options for each SIP Server application to be run in cluster mode:

Name Cluster Value Description
server-role 5 For SIP Server to run in cluster mode.
sip-link-type 3 For SIP Server to run in multi-threaded mode.
geo-location <string> A string identifying the data center to which this SIP Server instance belongs. It is used by SIP Proxy to select SIP Server in same data center as SIP Proxy. SIP Server uses it to select geo-location for 3PCC calls. All applications deployed in the same data center use the same value for this geo-location parameter.
sip-address <SIP Server A-Record FQDN> For SIP Server to build the Via and Contact headers in SIP messages.
sip-address-srv <blank> For SIP Server to build the Via and Contact headers in SIP messages.
sip-outbound-proxy true
sip-enable-gdns true To resolve SRV contacts.
sip-enable-rfc3263 true To resolve priority and weight in SRV tables.
dial-plan <dial plan DN name>_<short data center name> The name of the Dial Plan DN (the VoIP Service DN with service-type set to feature-server).
find-trunk-by-location true To enable selection of the trunk and softswitch by geo-location. This is required to keep SIP signalling on the correct data center.
enable-strict-location-match all To enable strict matching of MSML resources, which is required for the SIP Cluster. SIP Server in a particular geo-location must only use MCP resources in the same geo-location.
sip-enable-x-genesys-route true To enable a private X-Genesys-Route header in SIP messages towards SIP Proxy. It's exclusively used by (and not propagated beyond) the SIP Proxy.
sip-port <SIP port>
http-port <HTTP port>
management-port <management port>

The sample configuration:

[agent-reservation]
request-collection-time=300 msec

[backup-sync]
addp-remote-timeout=11
addp-timeout=7
addp-trace=full
protocol=addp

[call-cleanup]
cleanup-idle-tout=60 min
notify-idle-tout=5 min
periodic-check-tout=10 min

[extrouter]
cast-type=route direct-notoken direct-callid reroute direct-uui direct-ani dnis-pool direct-digits pullback route-uui direct-network-callid

[Log]
all=/mnt/log/SIPS_<datacenter>_1/SIPS_<datacenter>_1
buffering=false
expire=15
segment=100 MB
spool=/mnt/log/SIPS_<datacenter>_1
standard=network
time-format=iso8601
verbose=all
x-gsipstack-trace-level=3
x-server-trace-level=3

[log-filter]
default-filter-type=hide

[TServer]
acw-persistent-reasons=false
after-routing-timeout=18
agent-emu-login-on-call=true
agent-logout-on-unreg=true
agent-no-answer-action=notready
agent-no-answer-timeout=12
call-observer-with-hold=true
consult-user-data=inherited
clamp-dtmf-allowed=true
default-dn=default_rp
default-route-point=reject=404
default-route-point-order=after-dial-plan
default-music=music/on_hold_saas
dial-plan=DialPlan
divert-on-ringing=false
emulated-login-state=not-ready
extn-no-answer-timeout=12
greeting-call-type-filter=
greeting-delay-events=false
greeting-notification=
http-port=9096
init-dnis-by-ruri=true
logout-on-out-of-service=true
management-port=5002
merged-user-data=merged-over-main
monitor-consult-calls=true
msml-record-metadata-support=true
msml-record-support=true
msml-support=true
music-in-conference-file=qtmf://music/silence
override-to-on-divert=true
posn-no-answer-timeout=12
record-consult-calls=true
record-moh=false
recording-failure-alarm-timeout=900
recording-filename=$UUID$_$DATE$_$TIME$
registrar-default-timeout=140
ring-tone=qtmf://music/ring_back
rq-expire-tmout=0
rq-expire-tout=0
server-id=
set-notready-on-busy=true
shutdown-sip-reject-code=503
sip-address=<A-record FQDN>
sip-address-srv=
sip-call-retain-timeout=1
sip-dtmf-send-rtp=true
sip-enable-100rel=false
sip-enable-call-info=true
sip-enable-ivr-metadata=true
sip-enable-moh=true
sip-enable-rfc3263=true
sip-invite-treatment-timeout=15
sip-port=5060
sip-preserve-contact=true
sip-treatments-continuous=true
timeguard-reduction=1000
unknown-gateway-reject-code=503
userdata-map-trans-prefix=X-Genesys-


Configuring SIP Servers for Virtual Queues

Virtual Queue (VQ) SIP Servers are used primarily to manage Virtual Queues. This eliminates the need to synchronize Virtual Queue states across SIP Cluster Nodes.

  1. Deploy SIP Servers as an HA pair (one HA pair per data center), Hot Standby redundancy mode, by following the standard procedure.
    • Suggested application names: SIPS_VQ_<datacenter>, SIPS_VQ_<datacenter>_B.
  2. On the Connections tab, add the following connections:
    • confserv_proxy_<datacenter>—Set to the following parameters:
      • Connection Protocol: addp
      • Trace Mode: Trace On Both Sides
      • Local Timeout: 60
      • Remote Timeout: 90
    • MessageServer_<datacenter>—Set to the following parameters:
      • Connection Protocol: addp
      • Trace Mode: Trace On Both Sides
      • Local Timeout: 7
      • Remote Timeout: 11
  3. On the Switches tab, add the VQ-switch object to each VQ SIP Server application. All VQ SIP Servers must be associated with the same VQ-switch.
  4. On the Server Info tab, must be only the default port.
  5. VQ SIP Server sample configuration:
[agent-reservation]
request-collection-time=300 msec

[backup-sync]
addp-remote-timeout=11
addp-timeout=7
addp-trace=full
protocol=addp

[call-cleanup]
cleanup-idle-tout=60 min
notify-idle-tout=5 min
periodic-check-tout=10 min

[extrouter] 
cast-type=route direct-notoken direct-callid reroute direct-uui direct-ani dnis-pool direct-digits pullback route-uui direct-network-callid

[Log]
all=/mnt/log/SIPS_VQ_<datacenter>/SIPS_VQ_<datacenter>
buffering=false
expire=15
segment=100 MB
spool=/mnt/log/SIPS_VQ_<datacenter>
standard=network
time-format=iso8601
verbose=all
x-gsipstack-trace-level=3
x-server-trace-level=3

[TServer]
acw-persistent-reasons=false
after-routing-timeout=18
agent-emu-login-on-call=true
agent-logout-on-unreg=true
agent-no-answer-action=notready
agent-no-answer-timeout=12
call-observer-with-hold=true
consult-user-data=inherited
clamp-dtmf-allowed=true
default-dn=default_rp
default-route-point=reject=404
default-route-point-order=after-dial-plan
default-music=music/on_hold_saas
dial-plan=DialPlan
divert-on-ringing=false
emulated-login-state=not-ready
extn-no-answer-timeout=12
greeting-call-type-filter=
greeting-delay-events=false
greeting-notification=
http-port=9096
init-dnis-by-ruri=true
logout-on-out-of-service=true
management-port=5002
merged-user-data=merged-over-main
monitor-consult-calls=true
msml-record-metadata-support=true
msml-record-support=true
msml-support=true
music-in-conference-file=qtmf://music/silence
override-to-on-divert=true
posn-no-answer-timeout=12
record-consult-calls=true
record-moh=false
recording-failure-alarm-timeout=900
recording-filename=$UUID$_$DATE$_$TIME$
registrar-default-timeout=140
ring-tone=qtmf://music/ring_back
rq-expire-tmout=0
rq-expire-tout=0
server-id=
set-notready-on-busy=true
shutdown-sip-reject-code=503
sip-address=<A-record FQDN>
sip-address-srv=
sip-call-retain-timeout=1
sip-dtmf-send-rtp=true
sip-enable-100rel=false
sip-enable-call-info=true
sip-enable-moh=true
sip-enable-rfc3263=true
sip-invite-treatment-timeout=15
sip-port=5060
sip-preserve-contact=true
sip-treatments-continuous=true
timeguard-reduction=1000
userdata-map-trans-prefix=X-Genesys-

You will add VQ SIP Servers to the following applications:

  • Stat Servers in each data center
  • A dedicated HA pair of Interaction Concentrator instances to monitor an HA pair of VQ SIP Servers in the same data center
  • Each URS and ORS located in the same data center


Configuring SIP Servers for Historical Reporting

When operating in cluster mode, Interaction Concentrator server (ICON) must connect to two ports of SIP Server: T-Controller (TCport) and Interaction Proxy (IPport). For this purpose, a dummy SIP Server application must be created. When configuring ICON for Voice details, add a connection to the IPport of the actual SIP Server application, and add a connection to the TCport of the dummy SIP Server application. Each connection represents a session. Genesys Info Mart requires each session to be associated with a SIP Server application.

  1. Deploy a SIP Server application, by following the standard procedure.
  2. On the Switches tab, add the SIP Cluster Switch, the same Switch as in the actual SIP Server application.
  3. On the Server Info tab, add the same listening ports as in the actual SIP Server application. The Server Info tab must not contain HA configuration.
  4. On the Connections tab, don't add anything. It must be empty.
  5. On the Options tab in the [TServer] section, don't make any changes.
  6. On the Start Info tab, clear the Auto-Restart box to avoid SCS restarting the application.
  7. On the Annex tab in the [sml] section, set autostart=false to avoid SCS restarting the application.
Important
The dummy SIP Server application must not be added to the applications option of the SIP Cluster DN.

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This page was last modified on October 31, 2018, at 15:56.