52.2. CTISIP tables

52.2.1. Users table

SIP users must be registered 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:                   8             <Read Only>
PWD:                  ***
ACT:                  YES
GROUP:                
CTIP:                 #
CLUS:                 #
ADDRBOOK-SYNC:        SYS           
ADDRBOOK-NUMBER:      AUTO          
ADDRBOOK-OUTDIAL:     NONE          
ADDRBOOK-PRIV-MAX:    SYS
ADDRBOOK-PUB-ENABLED: SYS           
OPC-ROLE:             USER
OPC-VIEW:             *
OPC-HIDE-NUMBERS:     NO
OPC-MONITOR:          SIP
OPC-PRIVACY:          NO
CHAT:                 NO    
CHAT-USER:            SYS
CHAT-PWD:             SYS
HTTP:                 YES
HTTP-HOME-URL:        
HTTP-PROT:            PLAIN,SSL
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-BUSY-INUSE:       NO
SIP-NUMBER:           10
SIP-ADDRBOOK-NUM:     SIP-NUMBER
SIP-CG-NUM:           AUTO
SIP-FWD-CG-NUM:       CALLER
SIP-DISPLAY-NAME:     SYS
SIP-CTIP-TYPE:        NET-PUBLIC
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-AUTH-USER:    AUTO ()
SIP-REM-REG:          NO            
-------------------------------------------------------------------------------

Meaning of the most important parameters:

SIP

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

SIP-TYPE
  • PHONE: The user is a SIP client of Abilis, typically a phone or a softphone and the SIP-DOMAIN specifies the local domain of Abilis. If the SIP-HOST and/or SIP-UDP-REMPORT (SIP-TCP-REMPORT) are dynamic then the client has to register on Abilis.

  • LOCAL-PEER: The user is a SIP PEER as Abilis and SIP-DOMAIN specifies the local domain of Abilis. Calling and Called numbers are both passed to the user. If the SIP-HOST and/or SIP-UDP-REMPORT (SIP-TCP-REMPORT) are dynamic then the client has to register on Abilis.

  • 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.

    [Note]Note

    The user may also be a PEER, it means a device that has the same SIP role of the Abilis and the calling number has to be passed unchanged.

SIP-DOMAIN

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.

SIP-HOST

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].

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]. Only for SIP-HOST not equal to DYNAMIC.

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]. Only for SIP-HOST not equal to DYNAMIC.

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

Transport protocol used to receive/send calls for this user [TCP, UDP].

SIP-IP-PERMIT

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 "*"..

SIP-MAXSES-BID

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

SIP-MAXSES-IN

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

SIP-MAXSES-OUT

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

SIP-BUSY-INUSE

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.

SIP-NUMBER

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.

SIP-ADDRBOOK-NUM

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)

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-DISPLAY-NAME

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]Note

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

SIP-CTIP-TYPE

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

SIP-RG-IN

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

SIP-ROUTE-BY-SD

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.

SIP-PROVIDE-SG

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

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-CPO-SIGNALLING

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

SIP-RCC-DISABLE

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.

SIP-SS

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

SIP-SS-PICKUP

SIP supplementary service. Pickup permissions [NO, GROUPS, 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-SS-VM

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

SIP-REMOTE-NAT

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;].

SIP-LOCAL-NAT

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].

SIP-EXTERNAL-IP

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].

SIP-KEEPALIVE

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

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-DISC-AUDIO

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.

SIP-BC-TRANSP

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

SIP-T38

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

SIP-T38-G711

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

SIP-T38-PACKING

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

SIP-T38-REDUND

Error recovery method [SYS, NONE, REDUNDANCY].

SIP-T38-REDUND-PCK

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

SIP-UA

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").

SIP-UA-PERMIT

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

SIP-REM-USER

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-REM-PASS

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-REM-AUTH

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].

SIP-REM-AUTH-USER

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.

SIP-REM-REG

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

52.2.2. 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, isn't 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 isn't 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's 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's a “read only” list, you can't modify it, as it's automatically created by the system.