Logging functionalities activation/deactivation.

Runtime CTISIP activation/deactivation [NO, YES].

Traced data [SIP, ALL], where:

Log events filter at system startup [ERR, REG, INFO] or FULL, where:

Log size [20..2000 Kibyte].

UDP payload [1472..65507 byte].

The maximum number of SIP sessions available for phone calls [1..255].

Number of SIP sessions/dialogs initiated by NON-INVITE methods: i.e. Register, Subscribe and Notify [30..100].

Local UDP port of SIP protocol [1..65535].

Base local UDP port usable for RTP/RTCP sessions [1024..65535]; this feature is required to assign high priority to RTP/RTCP packets.

Range of permitted local UDP ports usable for RTP/RTCP sessions [50..1000]; this value must not be lower than (sesnum * 2 + 50).

Sets the TOS value for SIP protocol.

Sets the TOS value for RTP protocol.

Source IP address for outgoing connections [R-ID: the source IP address of the outgoing datagrams will be set to the current RouterID value; OUT-IP: the source IP address of the outgoing datagrams will be set on the base of the output IP interface; 1-126.x.x.x, 128-223.x.x.x: the source IP address of the outgoing datagrams will be set to the selected value; Ip-nnn: use the current IPADD of the specified IP resource].

Abilis external IP address. [R-ID: the source IP address of the outgoing datagrams will be set to the current RouterID value; OUT-IP: the source IP address of the outgoing datagrams will be set on the base of the output IP interface; 1-126.x.x.x, 128-223.x.x.x: the source IP address of the outgoing datagrams will be set to the selected value]. When OUT-IP or R-ID are set, the address must be determined at the call start and must not change until call end.

Selects how to fill Display Name in From, P-Asserted-Identity, Remote-Party-ID fields [NO: Do not add display name; CG: the value present in the CG field (calling number) provided by CTIR; SG: the value present in the SG field (subaddress calling) provided by CTIR; SG-CG: the value present in the SG field or CG field if SG field is missing; ADDRBOOK: field with the name of calling number from address book; ADDRBOOK-SG: field with calling number if the name of calling number is missing in address book].

	Note
	When SIP-TYPE:SERVER the CG is replaced by SIP-REM-USER value.

	Tip
	Interesting chapters: Chapter 67, Address Book; Section 85.4.6, “How can I activate to view the caller's name on the display of SIP phones?”.

Incoming requests: accepted source IP address [*, 1-126.x.x.x, 127.0.0.1, 128-223.x.x.x]; a client is accepted if it also satisfies SIP-IP-PERMIT specified in User table.

Incoming requests: list of further accepted source IP addresses [#, IP/IR/RU/MR listname]; a client is accepted if it also satisfies SIP-IP-PERMIT specified in User table.

Expiration time of incoming subscriptions [60..3600 sec].

Maximum number of subscriptions that CTISIP may handle independently from the event type, the subscriber and the monitored resource [0..1000]. If we are handling two SIP phones with BLF and one phone monitors 150 phones and the other monitors 50 phones then max-sub is 150+50=200. If we have 30 phones and each monitors 20 phones, then we have 600 subscriptions.

Authentication types offered to autenticating/registering users (incoming calls/registrations) [PLAIN, DIGEST]. Values can be joined using ',' character.

Authentication types acceptable when authenticating/registering to a peer (outgoing calls/registrations).[PLAIN, DIGEST]. Values can be joined using ',' character.

Default duration of incoming registration; users must register themselves before this time interval expires.

Qualify time interval [4..300 sec].

Maximal number of redirections (forwards) between two NNI users [1..10].

SIP NOT-INVITE authentication timeout for incoming/outgoing calls initiated by REGISTER method [2..255 sec].

SIP INVITE authentication timeout for incoming/outgoing calls initiated by INVITE method [2..255 sec].

Session keepalive time [90..600 sec]; when this session inactivity timer expires a RE-INVITE or UPDATE message is sent. The keepalive during a call is used to monitor if the connection is stil alive.

CTIP type assigned to CTISIP driver [USER, NET-PRIVATE, NET-PUBLIC].

Enable/disable Net-Public to Net-Public call transfer when both calls are outgoing (outgoing to outgoing) [NO: NP to NP disabled; YES: NP to NP enabled, except when both calls are outgoing, which is not allowed to avoid the “hang trunk” problem; SYS: NP to NP depends on NP-CT in CtiSys].

Routing management [PREFIX, EN-BLOC].

Dialing timeout [1..15 sec]. When SIP resource is the destination and at least a number was already submitted the timeout between digits is 5 seconds (DIALT) but may be changed.

Timeout for “empty” setup [15..120 sec].

It allows routing using subaddress called field (SD) [NO: the SD is completely ignored; YES: the user corresponding to SD is searched: if found the call is forwarded to that user, otherwise the standard search in CTISIP table is performed; USER: the user corresponding to SD is searched: if found and the user has parameter SIP-ROUTE-BY-SD:YES, the call is forwarded to that user, otherwise the standard search in CTISIP table is performed].

	Tip
	Interesting chapter: Section 59.4.4.2, “Abilis CTI Routing of “Site 1 ” using subaddress called field”.

Allows insertion of SIP USER NAME in subaddress calling field [NO: disable insertion of USERNAME in calling subaddress; YES: enable insertion of USERNAME in calling subaddress; USER: insertion of USERNAME in calling subaddress depends on individual user configuration]. When it's enabled, either as result of value YES or because value is USER and the user corresponing to the subaddress content allows it, the call sent to CTIR will include in the subaddress calling field the SIP USER NAME of the caller.

DTMF mode used by CTISIP in outgoing DTMF [INBAND: the outband DTMF received from CTIR is not dropped, only the audio stream is passed; INFO: the outband DTMF received from CTIR is sent using INFO message; RFC2833: the outband DTMF received from CTIR is sent using RFC2833 payload].

Enable/Disable the reproduction of the audio message present in DISCONNECT with in-band-info received from CTIR [NO, YES]. If set ot YES the duration of the SIP session in active state is increased until CTIR times-out (typically up to 30 sec), or the SIP agent closes the call.

Duration of a DTMF played from a DTMF FULL FRAME [40..1000 msec]. When CTISIP has to convert a DTMF FULL FRAME command to CTIR DTMF frame, it produces as many frames as needed to cover the period specified by this parameter. The actual interval must be rounded to the value immediately higher than configured one.

Duration of a silence played after a PLAY-DTMF [50..1000 msec]. When CTISIP has to convert a DTMF FULL FRAME command to CTIR DTMF frame, it produces as many frames as needed to cover the perios specified by PLAY-DTMF parameter. Before a new DTMF can be played, a silence period must elapse, and this is controlled by this parameter.

Duration of a DTMF from CTIR in order to allow digit recognition [20..100 msec]. When CTISIP received DTMF frames from CTIR, and after it created the most accurate and monotonical timestamp, the DTMF length is measured and for a successful digit recognition it must be of at least the length specified here. If the configured value is not an exact multiple of the frame rate, it must be rounded to the next upper value multiple of frame rate.

Duration of a silence after a DTMF from CTIR in order to recognise a digit [20..100 msec].When receiving DTMFs from CTIR to SIP this is the minimum duration of the silence after a tone to actually recognize the following tone and digit to be sent to SIP.

SIP T1 time; Round Trip Time (RTT) estimate.

SIP T2 time; maximum retransmission interval for NON-INVITE requests and INVITE responses.

SIP T4 time; maximum duration that a message can remain in the network.

Desired channel frequency for bandwidth optimisation, to be rounded down to a codec frame length multiple [20, 30, 40, 50, 60, 70, 80, 90 msec]. It represents how many msec of voice are desired to be packed into a single UDP packet; it's a “desire” in the sense that the actual optimisation will depend on the frame length in msec of the coder in use, and the result will be an a approximation by defect.C

Enables/disables Call Path Optimization between two SIP users. It is used when SIP-CPO:SYS in user service, otherwise the value of SIP-CPO is used.

Call Path Optimization signalling [NO: never attempt a CPO of the signalling path; TRANSFER: attempt the CPO just in case of transferring; ALWAYS: always attempt the CPO between two SIP call legs]

Domain for local users. From 0 up to 64 characters in the range ['0'..'9', 'a'..'z', '-', '.']. DN name is forced to lower case. If Abilis has clients in the public side you can also specify a FQDN.

Specifies the threatment of the CG number for calls TO the SIP users (OUTGOING) [PRIVATE, PUBLIC]. If PRIVATE the CG is sent unmodified, if PUBLIC the CG is emptied when PI is set to RESTRICTED.

Enable/disable management of incoming redirecting [DISABLE, ENABLE].

Enable/disable T.38 support [NO, YES].

Number of T.38 packets in UDP packet [1..4].

Error recovery method [NONE, REDUNDANCY: if the redundancy method is used it's possible to increase the redundancy packets].

Number of T.38 packets used for error recovery [1..4].

Enable/disable T.38 support with G.711 codec [NO, YES].

Enable/disable Runtime Codec Change (RCC) [NO, YES]. This feature permits the change of the coder once the call is already established. The purpose of this feature, which is perfectly SIP compliant, is to avoid transcoding all the times that it is possible by choosing a coder which is supported by both sides although not currently in use. This feature is very effective when call transfers takes place. A user may have two calls with two different parties that use two different codec, e.g. G.711 and G.729, when a call transfer is ordered the two parties will be directly connected but since one party was using G.711 and the other G.729 we were forced to make a transcoding even if both supports G.729. With the RCC feature the party running G.711 will be changed on the fly to G.729. The run time codec change allows to save voice quality and sw and hw resource in case of transcoding. Disable the RCC only if the SIP devices have troubles in handling the codec change.

Local user agent. AUTO or from 1 up to 32 ASCII printable characters. Case is preserved. Spaces are allowed. Strings holding spaces must be written between quotation marks (E.g.: "my user agent"). Abilis may present itself as a different device, it's useful to avoid filters based on User Agent.

Working directory where persistent information is stored. It cannot be empty. Physical full path in DOS notation, i.e. starting with a drive letter in the range ['A'..'Z'] and ending with the '\' character. Max. 128 characters. Case is preserved. Spaces are allowed. Strings holding spaces must be written between quotation marks (E.g.: "C:\My dir\").