| 49.2. CTISIP tables | ||
|---|---|---|
 | Chapter 49. SIP - Session Initiation Protocol |  |
| 49.2. CTISIP tables | ||
|---|---|---|
| | Chapter 49. SIP - Session Initiation Protocol | |
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 NOType 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]](../images/note.png) | 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:
SIPEnables/disables SIP service for the user
[NO, YES].
SIP-TYPEPHONE: 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 the 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.
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.
| 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. |
![[Tip]](../images/tip.png) | Tip |
|---|---|
Interesting chapter: Section 49.2.1.1, “CTISIP INVITE messages” |
SIP-DOMAINDomain 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-HOSTIP 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-REMPORTTCP 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-REMPORTUDP 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-locportUDP 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-SRCADDSource 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-PROTTransport protocol used to receive/send calls for this user
[TCP, UDP].
SIP-IP-PERMITAllowed 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-BIDMaximum number of simultaneous bidirectional sessions [0..255].
SIP-MAXSES-INMaximum number of simultaneous reserved input sessions [0..255].
SIP-MAXSES-OUTMaximum number of simultaneous reserved output sessions [0..255].
SIP-NUMBERUser 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-NUMAddress 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-NUMCalling 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-NUMIndicates 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-CTIP-TYPECTIP type [SYS, USER,
NET-PRIVATE,
NET-PUBLIC].
SIP-RG-INEnable/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-SDAllows 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-SGAllows insertion of SIP USER NAME in subaddress calling
field [NO, YES].
SIP-LCS-GROUPLast Calling number Service group identifier [NONE, 1..32].
SIP-AUTHAuthentication 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-FREQSIP 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].
SIP-CPO-SIGNALLINGCall Path Optimization signalling [SYS,
NO, TRANSFER,
ALWAYS].
SIP-RCC-DISABLEEnable/disable Runtime Codec Change (RCC)
[SYS, NO,
YES].
SIP-SSEnable/disable SIP supplementary services
[NO, YES].
SIP-SS-PICKUPSIP supplementary service. Pickup permissions
[NO, GROUPS,
ANY].
SIP-SS-PRES-CGSIP supplementary service. Calling present
[NO, YES].
SIP-SS-CF-DNDsupplementary service. Call forwarding and
Do-Not-Disturb [NO,
YES].
SIP-SS-VMSIP supplementary service. Voice Mail
[NO, YES].
SIP-REMOTE-NATNAT Traversal method when remote user is behind NAT
[NO: send audio to the udp port specified in
the SIP protocol (SDP); STRICT: send audio to
the same RTP port from which the media is received (Cisco
symmetric RTP) but the IP address must be the same of the SIP
signalling channel; LOOSE: send audio to the
same RTP port from which the media is received (Cisco symmetric
RTP) whatever is the IP address].
SIP-LOCAL-NATNAT 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-IPNumeric 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-KEEPALIVEEnable/disable Keep-alive feature
[ENABLED, DISABLED]
SIP-DTMF-MODEDTMF 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-AUDIOEnable/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-TRANSPSets the ISDN Bearer Capability (BC) for incoming calls with
codec CLEARMODE (TRANSPARENT coder for CPX)
[UDI, SPEECH].
SIP-T38Enable/disable T.38 support [SYS,
NO, YES].
SIP-T38-G711Enable/disable T.38 support with G.711 codec
[SYS, NO,
YES].
SIP-T38-PACKINGNumber of T.38 packets in UDP packet [SYS, 1..4].
SIP-T38-REDUNDError recovery method [SYS,
NONE, REDUNDANCY].
SIP-T38-REDUND-PCKNumber of T.38 packets used for error recovery [SYS, 1..4].
SIP-UALocal 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-PERMITAllowed peer User Agent. "*" or the name of a TXT/RU/MR list.
SIP-REM-USERSIP 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-PASSSIP 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-AUTHSIP 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-REGEnable/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].
A caller sends this message to request that another endpoint join a SIP session, such as a conference or a call. This message can also be sent during a call to change session parameters.
Meaning of the most important parameters:
Request-URI: The
Request-URI header field contains URI
of the next hop of the message.
From: The From header
field identify initiator of the invitation
To: The To header
field identify recipient of the invitation
In a SIP-TYPE:PHONE the
INVITE from 101 to the user sent by the abilis contains:
Request-Uri:
test@80.80.80.80 (where
test is the sip user and
80.80.80.80 is the IP address of the
device)
From:
101@abilis.voip.net (where
abilis.voip.net is the local
domain)
To:
test@abilis.voip.net
In a SIP-TYPE:LOCAL-PEER
the INVITE from 101 to 0123456 sent by the abilis contains:
Request-Uri:
0123456@80.80.80.80 (where
0123456 is the called number and
80.80.80.80 is the IP address of the
user)
From:
101@77.77.77.77 (where
77.77.77.77 is the IP adress of the
Abilis)
To:
0123456@80.80.80.80
In a SIP-TYPE:SERVER the
INVITE to 0123456 sent by the abilis contains:
Request-Uri:
0123456@sipserver.com (where
0123456 is the called number and
sipserver.com is the FQDN
of the server)
From:
abilis@sipserver.com (where
abilis is the SIP remote user and
sipserver.com is the remote
domain)
To:
0123456@sipserver.com
In a SIP-TYPE:REMOTE-PEER
the INVITE from 101 to 0123456 sent by the abilis contains:
Request-Uri:
0123456@sipserver.com (where
0123456 is the called number and
sipserver.com is the FQDN
of the remote peer)
From: 101@77.77.77.7
(where 77.77.77.77 is the
IP address of the Abilis)
To:
0123456@sipserver.com
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 STATICThere 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 |
|---|---|
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 |
|---|---|
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 |
|---|---|
It is a “read only” list, you cannot modify it, as it is automatically created by the system. |
| |  | |
| 49.1. CTISIP resource |  | 49.3. CTI Routings |