49.2. CTISIP tables
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49.2. CTISIP tables

49.2.1. Users table

SIP users must be registrated in the Users table. All the parameters, mandatory for the registration, the authentication and the SIP identification are defined in each Abilis user's profile.

Use the below command to display the parameters of the users; the d user: ? command shows the meaning of all parameters.

[11:15:19] ABILIS_CPX:d user

- Not Saved (SAVE CONF) -------------------------------------------------------
------------------------+-------------+----------------------------------------
USER             PWD ACT|CTIP CLUS    |CHAT LDAP PPP FTP HTTP MAIL IAX SIP VO
------------------------+-------------+----------------------------------------
admin            *** YES #    #        YES  YES  YES YES YES  NO   NO  NO  NO
guest                YES #    #        NO   YES  NO  NO  NO   NO   NO  NO  NO
test                 YES #    #        NO   NO   NO  NO  NO   NO   NO  YES NO

Type the following command to view user's details:

[11:15:19] ABILIS_CPX:d user:test

Parameter:          | Value:
--------------------+----------------------------------------------------------
USER:                 test
REAL-NAME:            test
ID:                   5             <Read Only>
PWD:                     
ACT:                  YES
GROUP:                
CTIP:                 #
CLUS:                 #
ADDRBOOK-SYNC:        SYS           
ADDRBOOK-NUMBER:      AUTO          
ADDRBOOK-OUTDIAL:     NONE          
ADDRBOOK-PUB-ENABLED: SYS           
OPC-ROLE:             USER
OPC-VIEW:             *
OPC-HIDE-NUMBERS:     NO
OPC-MONITOR:          NONE
OPC-PRIVACY:          NO
CHAT:                 NO    
CHAT-USER:            SYS
CHAT-PWD:             SYS
SIP:                  YES   
SIP-TYPE:             PHONE         
SIP-DOMAIN:           SYS
SIP-HOST:             DYNAMIC
SIP-TCP-REMPORT:      (DYNAMIC)
SIP-UDP-REMPORT:      (DYNAMIC)
sip-udp-locport:      SYS
SIP-SRCADD:           SYS
SIP-PROT:             UDP
SIP-IP-PERMIT:        *
SIP-MAXSES-BID:       2
SIP-MAXSES-IN:        0
SIP-MAXSES-OUT:       0
SIP-NUMBER:           
SIP-ADDRBOOK-NUM:     SIP-NUMBER
SIP-CG-NUM:           AUTO
SIP-FWD-CG-NUM:       CALLER
SIP-CTIP-TYPE:        SYS
SIP-RG-IN:            SYS
SIP-ROUTE-BY-SD:      NO            
SIP-PROVIDE-SG:       NO
SIP-CLIP-RULE:        SYS
SIP-BUSY-NOCHAN:      NO
SIP-LCS-GROUP:        NONE
SIP-CPO-RTP:          SYS
SIP-CPO-SIGNALLING:   SYS
SIP-RCC-DISABLE:      SYS
SIP-SS:               NO
SIP-SS-PICKUP:        GROUPS
SIP-SS-PRES-CG:       YES
SIP-SS-CF-DND:        YES
SIP-SS-VM:            YES
SIP-AUTH:             SYS
SIP-CHAN-FREQ:        SYS
SIP-REMOTE-NAT:       NO
SIP-LOCAL-NAT:        NO
SIP-EXTERNAL-IP:      SYS
SIP-KEEPALIVE:        ENABLED
SIP-DTMF-MODE:        SYS
SIP-DISC-AUDIO:       SYS
SIP-BC-TRANSP:        UDI
SIP-T38:              SYS
SIP-T38-G711:         SYS
SIP-T38-PACKING:      SYS
SIP-T38-REDUND:       SYS
SIP-T38-REDUND-PCK:   SYS
SIP-UA:               SYS
SIP-UA-PERMIT:        *
SIP-REM-USER:         
SIP-REM-PASS:                 
SIP-REM-AUTH:         SYS
SIP-REM-REG:          NO            
-------------------------------------------------------------------------------
[Note]Note

This command displays only the parameters related to enabled driver; if you want to see all the user parameters type the d usere:sip_user command.

Meaning of the most parameters:

SIP

Enables/disables SIP service for the user.

SIP-TYPE

  • PHONE: The user is a SIP client of Abilis, in example it is a phone or a softphone and SIP-DOMAIN specifies the local domain of Abilis. Usually the user registers on Abilis or has a static SIP-HOST and SIP-UDP/TCP-PORT.

  • LOCAL-PEER: The user is a Peer as Abilis and SIP-DOMAIN specifies the local domain of Abilis. Calling and Called numbers are both passed to the user. Usually the user registers on Abilis or has a static SIP-HOST and SIP-UDP/TCP-PORT.

  • SERVER: The user is a SIP server for Abilis and SIP-DOMAIN specifies the remote domain. Usually the Abilis registers on this user.

  • REMOTE-PEER: The user is a Peer as Abilis and SIP-DOMAIN specifies the remote domain.Calling and Called numbers are both passed to the user. Usually the Abilis registers on this user.

Before 7.3.4 version a different naming was used. Here is the matching table:

Table 49.1. SIP-TYPE matching table

Current namePrevious Name
PHONEPHONE
LOCAL-PEERLOCAL-PROXY
SERVERREMOTE-PROXY
REMOTE-PEER<not available>

SIP-DOMAIN

Domain of the called SIP UA server in outgoing calls.

SIP-HOST

IP address of the SIP UAC/UAS. 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].

SIP-TCP-REMPORT

TCP port on which the remote user is listening; Abilis outgoing TCP 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].

SIP-UDP-REMPORT

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].

sip-udp-locport

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 ).

SIP-SRCADD

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].

SIP-PROT-IN

Transport protocol used to receive calls from this user.

SIP-PROT-OUT

Transport protocol used to place outgoing calls to this user.

SIP-IP-PERMIT

Range of allowed IP addresses of the SIP user.

SIP-MAXSES-BID

Maximum number of simultaneous bidirectional sessions.

SIP-MAXSES-IN

Maximum number of simultaneous reserved input sessions.

SIP-MAXSES-OUT

Maximum number of simultaneous reserved output sessions.

SIP-NUMBER

Number that identifies the user; if this number is not null, it is used to route calls to the user.

SIP-CG-NUM

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].

SIP-FWD-CG-NUM

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].

SIP-ROUTE-BY-SD

Allows routing using subaddress called field. 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.

SIP-PROVIDE-SG

Allows insertion of SIP USER NAME in subaddress calling field.

SIP-LCS-GROUP

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

SIP-AUTH

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-CHAN-FREQ

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].

SIP-CPO-RTP

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

SIP-SS

Enable/disable SIP supplementary services [NO, YES]

SIP-SS-PICKUP

SIP supplementary service. Pickup permissions [NO, ANY]

SIP-SS-PRES-CG

SIP supplementary service. Calling present [NO, YES

SIP-SS-CF-DND

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

SIP-REMOTE-NAT

Position of the client in Internet [NO: requests and responses to the customer's phone occurs on the Contact header field specified. RTP checks that remote ip/address matches with which one specified in SDP (symmetric RTP is not allowed); STRICT: requests and responses to the customer's phone occurs on the same address/port from which the remote requests/responses came from. RTP checks that remote address matches with which one of signaling (symmetric RTP is allowed); LOOSE: requests and responses to the customer's phone occurs on the same address/port from which remote requests/responses came from (symmetric RTP is fully allowed with no address checking)].

SIP-LOCAL-NAT

NAT traversal method [NO, EXTERNAL-IP].

SIP-EXTERNAL-IP

IP address of the SIP UA.

SIP-KEEPALIVE

Enables/disables the SIP KEEPALIVE.

SIP-DTMF-MODE

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].

SIP-REM-USER

The name used in the remote SIP UA server to identify the Abilis; this name is used for both registration and authentication purposes.

SIP-REM-PASS

The password used in the remote SIP UA server to identify the Abilis; this password is used for both registration and authentication purposes.

SIP-REM-AUTH

Authentication method when Abilis is authenticating/registering to a peer (outgoing calls/registrations) [SYS: uses the value in REM-AUTH parameter in CTISIP resource; PLAIN: basic authentication via user/password; DIGEST: DIGEST authentication type].

SIP-REM-REG

Enables/disables the registration of the Abilis to the remote SIP UA server [NO; YES: Abilis periodically register to the remote UA to inform remote peer about its IP address and TCP/UDP port].

49.2.2. CTISIP translation table

This table contains relations between a SIP-number (or a prefix, when * is included in the number) and a SIP-user. Calls which CTIR forwards to CTISIP finds the destination user by matching the called number (matching between the CDO field of the CTI routing and the CDI field of this table).

When SIP-CG-NUM:AUTO in the Users table, calls from CTISIP to CTIR will have:

  • the callerid provided by SIP user validated in the CTISIP translation table;

  • the SIP-number set in user service.

In case of validation failure the callerid will be overwritten with the value configured in the SIP-number of the user table (*, as wildcard, is not included).

Type the following command to view the details of the CTISIP translation table:

[17:22:35] ABILIS_CPX:d ctisip numbers  

Total:4        Sip-Number:3         Static:1 

NUMx:                     USER:                  Provenience:
------------------------------------------------------------
[500]                     test4                    SIP-NUMBER
[12]                      test3                    SIP-NUMBER
[11]                      test2                    SIP-NUMBER
10                        test                         STATIC

There are two types of entries:

  • SIP-NUMBER: when SIP-NUMBER is set in the SIP users chart, the CDI parameter in the chart will be the same.

    [Tip]Tip

    The connected entries are automatically added.

  • STATIC: when a SIP-NUMBER is not specified in the SIP users chart and it's associated by hand in the chart. This system is used to add several numbers to the same user (for instance in case of static routings).

Use the following commands to manage the SIP translation table:

  • a ctisip numx:<SIP-NUMBER> username:<name> : adds a new SIP-NUMBER;

  • s ctisip numx:<SIP-NUMBER> username:<name> : modifies the username of an existing SIP-NUMBER;

  • c ctisip numx:<SIP-NUMBER> : clears a SIP-NUMBER;

  • d ctisip numx:<SIP-NUMBER> : displays the list of SIP-NUMBER or a specific one.

[Tip]Tip

To a single user can be associated more SIP-numbers.

The SIP users creation generates automatically the NumSip list in which are located all the SIP-NUMBERS associated to the users (it is very useful for the CTIR configuration).

Type the following command to view the list :

[15:28:03] ABILIS_CPX:d list:numsip

LIST:NumSip               - IN                - Ref-Numb:1     Items-Numb:4    
     Automatically_generated_CTI_SIP_Numbers_list_(ReadOnly)
     --------------------------------------------------------------------------
     10                      11                      12
     500
[Note]Note

It is a “read only” list, you cannot modify it, as it is automatically created by the system.

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