Enables/disables SIP service for the user [NO, YES].

Domain of the called SIP UA server in outgoing calls.From 0 up to 64 characters in the range ['0'..'9', 'a'..'z', '-', '.' ] or SYS. Case is not preserved. SYS means to use DOMAIN value in CtiSip configuration and it is allowed only for SIP-TYPE equal to PHONE or LOCAL-PEER.

	Tip
	Interesting chapter: Section 84.4.15, “How to solve SIP call transfer issues?”.

IP address of the SIP host [DYNAMIC, 1-126.x.x.x, 127.0.0.1, 128-223.x.x.x] or FQDN host name of max. 64 characters in the range ['0'..'9', 'a'..'z', '-', '.' ]. FQDN name is forced to lower case. Domain and Host may differ, because SIP registrar server may be different from SIP proxy; normally proxies and SIP registrar server are co-located [DYNAMIC: IP is not known in advance, it is known after the user executes the registration; 1.0.0.0-126.255.255.255, 128.0.0.0-223.255.255.255: remote IP is known in advance; calls and registrations are performed and accepted only with this IP].

UDP port on which the remote user is listening; Abilis outgoing UDP calls for this user will be sent to this port [DYNAMIC: the port is learned from incoming registration; 1..65535: calls and registrations are performed and accepted only with this port]. Only for SIP-HOST not equal to DYNAMIC.

UDP port on which the Abilis is listening for this user [SYS, AUTO, 1..65535] . The default value is SYS and refers to the port parameter udp-locport. AUTO and a port different from the one configured in SIP port parameter "udp-locport" may be assigned only to SIP-TYPE REMOTE-PEER or SERVER. Note that this is a lower cased parameter, it means that an Abilis CPX reboot must be performed to apply changes, in detail you need to save the configuration ( command save conf ) and restart the Abilis ( via the command warm start ).

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; SYS: uses the value in SRCADD parameter in CTISIP resource].

Allowed IP address of the SIP user. One or two IP addresses in the range [1-126.x.x.x, 127.0.0.1, 128-223.x.x.x] separated by ':' (colon) or the name of an IP/IR list or "*"..

Maximum number of simultaneous bidirectional sessions [0..255].

Maximum number of simultaneous reserved input sessions [0..255].

Maximum number of simultaneous reserved output sessions [0..255].

Return BUSY if one or more sessions are in use [NO, YES]. It allows SIP with SIP-TYPE:PHONE to refuse calls if the user already involved in a conversation.

User number that identifies the resource for call routings. From 0 up to 20 characters in the range [0..9, *] optionally preceded by TON [u, i, n, o, s, h, c] and/or NP [x, e, d, t, l, p] attributes.; if this number is not null, it is used to route calls to the user.

Address book SIP phone number assigned to this user. "#" or "SIP-NUMBER" or from 1 up to 20 digits ['0'..'9'], optionally preceded by TON [u, i, n, o, s, h, c] and/or NP [x, e, d, t, l, p] attributes or 'macro'. (E.g.: 0'SIP-NUMBER' or 123'SIP-NUMBER.s2' or 'SIP-NUMBER'99)

Calling number to use for calls coming from the user. The parameter accepts from 1 up to 20 characters in the following range: [AUTO: enforces caller id information element equal to SIP-NUMBER; [0..9]: enforces the content with these exact digits; [0..9]*: replaces first specified digits and passes the remaining transparently; *: passes calling address information element transparently; #: removes calling number information element; ##: enforces the presentation restricted: the calling number is sent empty; ##[0..9]: enforces the presentation restricted: the calling number is sent with these exact digits; ##[0..9]*: enforces the presentation restricted: the first specified digits are replaced and the remaining are passed transparently; ##*: enforces the presentation restricted: the calling number is sent transparently].

Indicates how the calling number is managed in unconditional call transfers and call forwarding [CALLER: the calling number of the original call is passed to the new recipient; USER: the calling number of the SIP user performing the action is passed to the new recipient].

Selects how to fill Display Name in From, P-Asserted-Identity, Remote-Party-ID fields [SYS: Use the value specified in CTISIP resource; 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 66, Address Book; Section 84.4.6, “How can I activate to view the caller's name on the display of SIP phones?”.

CTIP type [SYS, USER, NET-PRIVATE, NET-PUBLIC].

Enable/disable management of incoming redirecting [SYS, DISABLE, ENABLE]. Set such parameter to allow the redirecting number coming from SIP to be passed to the CTI rouer

Allows routing using subaddress called field [NO, YES]. Calls from CTIR and directed to SIP users are first directed to the user with a USERNAME equal to what is specified in Subaddress Called; if such user does not exists, or the user disallows SIP-ROUTE-BY-SD, the call is routed using standard CTISIP table matches.

	Tip
	Interesting chapter: Section 58.4.4.2, “Abilis CTI Routing of “Site 1 ” using subaddress called field”.

Allows insertion of SIP USER NAME in subaddress calling field [NO, YES].

Last Calling number Service group identifier [NONE, 1..32].

	Tip
	Interesting chapters: Section 58.5.6, “Last Calling Service for a SIP user”; Section 81.18, “How to configure Last Calling Service”; Section 17.3, “Last Calling Service for UMTSBOX”.

Authentication types offered to autenticating/registering users (incoming calls/registrations) [SYS: uses the value in AUTH parameter in CTISIP resource; PLAIN: basic authentication via user/password; DIGEST: DIGEST authentication type].

SIP desired channel frequency for bandwidth optimisation, to be rounded down to a codec frame length multiple [SYS: uses the value in CHAN-FREQ parameter in CTISIP resource; 30..90: frequency for banwidth optimisation].

Enables/disables Call Path Optimization (CPO) [SYS: uses the value in CPO parameter in CTISIP resource; NO: doesn't allow CPO; YES: allows CPO].

Call Path Optimization signalling [SYS, NO, TRANSFER, ALWAYS].

Enable/disable Runtime Codec Change (RCC) [SYS, 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.

Enable/disable SIP supplementary services [NO, YES].

SIP supplementary service. Pickup permissions [NO, GROUPS, ANY].

	Tip
	Interesting chapter: Section 81.16, “How to enable pickup service for a SIP account”.

SIP supplementary service. Calling present [NO, YES].

supplementary service. Call forwarding and Do-Not-Disturb [NO, YES].

SIP supplementary service. Voice Mail [NO, YES].

NAT Traversal method when remote user is behind NAT [NO: send audio to the udp port specified in the SIP protocol (SDP); STRICT: Signaling and RTP must come from the same IP address, may be different from the payload of SIP signaling and SDP. It must be equal to the IP of the registration. Requires symmetric RTP (Cisco symmetric RTP), one in which the first audio part from the client behind NAT and then the server responds using the same reversed ports. In the case of transfers with optimized RTP, it uses private IP and private ports contained in SIP and SDP signaling; LOOSE: SIP signaling must indicate the IP equal to the real ones from which the packet, while the RTP (Cisco symmetric RTP) is symmetrical and the IP may be different from that used for the signaling. In the case of transfers with optimized RTP, it uses private IP and private ports contained in SIP and SDP signaling, as STRICT;].

NAT traversal method [NO In the signalling specify the real IP address of the Abilis; EXTERNAL-IP: In the signalling specify the IP address in SIP-EXTERNAL-IP parameter].

Numeric IPv4 address of the SIP UA [R-ID, OUT-IP, SYS, 1-126.x.x.x, 127.0.0.1, 128-223.x.x.x].

Enable/disable Keep-alive feature [ENABLED, DISABLED]. It's very important to have the SIP-KEEPALIVE enabled to avoid pending calls.

DTMF mode sent to the remote UA [SYS: uses DTMF-MODE value in CTISIP resource; 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 [SYS, 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.

Sets the ISDN Bearer Capability (BC) for incoming calls with codec CLEARMODE (TRANSPARENT coder for CPX) [UDI, SPEECH].

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

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

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

Error recovery method [SYS, NONE, REDUNDANCY].

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

Local user agent. "SYS" 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").

Allowed peer User Agent. "*" or the name of a TXT/RU/MR list.

SIP user name. From 0 up to 32 ASCII printable characters. Spaces are not allowed. Case is preserved.; this name is used for both registration and authentication purposes.

SIP user password. From 0 up to 32 ASCII printable characters. Spaces are not allowed. Case is preserved.; this password is used for both registration and authentication purposes.

SIP authentication methods offered to users [SYS: uses the value in REM-AUTH parameter in CTISIP resource; PLAIN: basic authentication via user/password; DIGEST: DIGEST authentication type].

Authentication user name. "AUTO" (value equal to SIP-REM-USER) or from 1 up to 32 ASCII printable characters. Spaces are not allowed. Case is preserved.

Enable/disable SIP auto-registration [NO; YES: Abilis periodically register to the remote UA to inform remote peer about its IP address and UDP port].