49.4. Example of configuration

Figure 49.1. Configuration scheme

Configuration scheme

In the figure there are the following elements:

The purpose is to enable the communication between:

Assumptions: the POTS cards, used to manage the analog phones and configured for Cluster and CtiLink for compressed phone calls, are already active on Abilis.

[Tip]Tip

Interesting chapter: Section 69.7, “How to register a SIP telephone onto Abilis”.

49.4.1. Activation and configuration of the CtiSip resource

Activate the CTISIP resource in the Abilis of “Site 1” and enable up to 10 SIP connections.

[10:27:42] ABILIS_CPX_1:s p ctisip act:yes sesnum:10

COMMAND EXECUTED

[10:27:53] ABILIS_CPX_1:d p ctisip

RES:CtiSip - Not Saved (SAVE CONF), Not Refreshed (INIT) ----------------------
Run    DESCR:Session_Initiation_Protocol
       LOG:NO                  ACT:YES                 mxps:2172
       sesnum:10               non-invite-sesnum:50    tcp-sesnum:0
       tcp-locport:5060        UDP-PORT-BASE:6000      SIP-TOS:0-N
       udp-locport:5060        UDP-PORT-RANGE:200      RTP-TOS:0-D
       SRCADD:OUT-IP                 
       EXTERNAL-IP:OUT-IP                 
       IPSRC:127.000.000.001   IPSRCLIST:PrivateIpAdd
       SUB-LIFETIME:180        max-sub:100             CTIP-TYPE:USER
       AUTH:DIGEST             KEEPALIVE:90            NPOO-CT:SYS
       LIFETIME:120            DISC-AUDIO:NO           ROUTING:EN-BLOC
       REM-AUTH:DIGEST         T1:500                  DIALT:5
       REM-LIFETIME:120        T2:4                    T302:15
       AUTH-TOUT:4             T4:5                    ROUTE-BY-SD:NO
       AUTH-TOUT-INVITE:4      CHAN-FREQ:20            PROVIDE-SG:NO
       DTMF-MODE:RFC2833       T38:YES                 CLIP-RULE:PRIVATE
       PLAY-DTMF:100           T38-G711:NO             RG-IN:DISABLE
       PLAY-SILENCE:100        T38-PACKING:1           CPO-RTP:NO
       DETECT-DTMF:40          T38-REDUND:REDUNDANCY   CPO-SIGNALLING:NO
       DETECT-SILENCE:40       T38-REDUND-PCK:1        RCC-DISABLE:NO
       DOMAIN:
       UA:AUTO (Abilis CPX - Ver. 8.0.3/STD - Build 3961.10 - Branch 8.0)
       wdir:C:\APP\SIP\

Keep all the default parameters .

Repeat the same operation for the Abilis of “Site 2”.

49.4.2. Users configuration

Activate two SIP user in the Abilis of “Site 1” (PC with Zoiper and SIP phone).

49.4.2.1. “Zoiper” user configuration.

Add the user in the Abilis of “Site 1” and enable it to SIP.

[10:50:19] ABILIS_CPX_1:a user:zoiper sip:yes

COMMAND EXECUTED

[10:50:28] ABILIS_CPX_1: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 NO   YES  NO  NO  NO  NO
guest                NO   #     #       NO  NO   NO  NO   NO   NO  NO  NO  NO
zoiper               YES  #     #       NO  NO   NO  NO   NO   NO  NO  YES NO

Give the new user the following characteristics:

  • password: zoiper

  • host: dynamic (the PC receives the IP address from a DHCP server)

  • simultaneous half-duplex call: 2

  • SIP number: 610

[09:45:52] ABILIS_CPX_1:s user:zoiper pwd:zoiper sip-host:dynamic sip-maxses-bid:2 sip-number:610

COMMAND EXECUTED

[09:46:23] ABILIS_CPX_1:d user:zoiper

- Not Saved (SAVE CONF) -------------------------------------------------------
Parameter:        | Value:
------------------+------------------------------------------------------------
USER:               zoiper
ALIAS:              zoiper
ID:                 9    <Read Only>
PWD:                ***
ACT:                YES
GROUP:
CTIP:               #
CLUS:               #
OPC-ROLE:           USER
OPC-VIEW:           *
OPC-MONITOR:        NONE
SIP:                YES
SIP-TYPE:           PHONE
SIP-DOMAIN:         SYS
SIP-HOST:           DYNAMIC
SIP-TCP-PORT:       (DYNAMIC)
SIP-UDP-PORT:       (DYNAMIC)
SIP-SRCADD:         SYS
SIP-PROT-IN:        TCP,UDP
SIP-PROT-OUT:       UDP
SIP-IP-PERMIT:      *
SIP-MAXSES-BID:     2
SIP-MAXSES-IN:      0
SIP-MAXSES-OUT:     0
SIP-NUMBER:         610
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-SS:             NO
SIP-SS-PICKUP:      NO
SIP-SS-PRES-CG:     YES
SIP-SS-CF-DND:      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
-------------------------------------------------------------------------------

49.4.2.2. “Sip phone” user configuration

Add the user in the Abilis of “Site 1” and enable it to SIP.

[09:51:19] ABILIS_CPX_1:a user:sip_phone sip:yes

COMMAND EXECUTED

[09:51:31] ABILIS_CPX_1: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  NO  YES  NO  NO  NO  NO
guest                  NO   #     #     NO   NO   NO  NO   NO  NO  NO  NO  NO
sip_phone              YES  #     #     NO   NO   NO  NO   NO  NO  NO  YES NO
zoiper            ***  YES  #     #     NO   NO   NO  NO   NO  NO  NO  YES NO

Give the new user the following characteristics:

  • password:sip_phone

  • host: dynamic (the PC receives the IP address from a DHCP server)

  • 2 simultaneous half-duplex calls

  • SIP number: 630

[09:51:51] ABILIS_CPX_1:s user:sip_phone pwd:sip_phone sip-host:dynamic sip-maxses-bid:2 sip-number:630

COMMAND EXECUTED

[09:52:38] ABILIS_CPX_1:d user:sip_phone

- Not Saved (SAVE CONF) -------------------------------------------------------
Parameter:          | Value:
--------------------+----------------------------------------------------------
USER:                 sip_phone
REAL-NAME:            sip_phone
ID:                   7             <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:           630
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            
-------------------------------------------------------------------------------

Add another “sip_phone” user in the Abilis of “Site 2” and assign it the SIP-NUMBER:910 (the configuration is similar to that one of “Site 1”).

49.4.3. CTISIP translation table

49.4.3.1. CTISIP translation table of “Site 1

Since the SIP-NUMBER parameter for the SIP users of “Site 1” is specified, the routings will be automatically connected in the table.

[11:25:38] ABILIS_CPX_1:d ctisip numbers 

Total:2        Sip-Number:2         Static:0 

NUMx:                     USER:                  Provenience:
------------------------------------------------------------
610                      zoiper                    SIP-NUMBER
630                      sip_phone                 SIP-NUMBER

Further modifications are not necessary.

49.4.3.2. CTISIP transaltion table of “Site 2

Since the SIP-NUMBER parameter for the SIP user of “Site 2” is specified, the routing will be automatically connected in the table.

[12:03:55] ABILIS_CPX_2:d ctisip numbers

Total:1        Sip-Number:1         Static:0 

NUMx:                     USER:                  Provenience:
------------------------------------------------------------
910                   sip_phone                    SIP-NUMBER

Further modifications are not necessary.

49.4.4. CTI Routings configuration

49.4.4.1. Abilis CTI Routing of “Site 1

The purpose is to enable the communication between:

  • SIP users in “Site 1” (SIP phone and PC with Zoiper);

  • the POTS phone and the SIP users of “Site 1”;

  • SIP users in “Site 1” and the SIP users in “Site 2”.

49.4.4.1.1. Communication between the SIP users

To connect up the PC with Zoiper and the SIP phone in “Site 1”, type:

[10:27:19] ABILIS_CPX_1:a ctir pr:0 type:stos cdi:'numsip' sp:64000

COMMAND EXECUTED

[16:42:17] ABILIS_CPX_1:d ctir

- Not Saved (SAVE CONF), Not Refreshed (INIT) ---------------------------------
Last change: 28/05/2015 10:25:10 CET

---+-----+-----------------+---------+--------------------+--------------------
PR |[DESCR]
   |TYPE |POI/SR   [SR]    |POO/DS   |CDI                 |CDO
ACT|NEXT       |LAST       |EEC |T301|CGI                 |CGO
EDT|SP   |SC   |DJ   |MJ   |FMDJ|FMMJ|SDI                 |SDO
   |SPOUT|SCOUT|DJOUT|MJOUT|LCS |LCST|SGI                 |SGO
   |                                 |RGI                 |RGO
   |FMRLY|FAXSP|MODSP|FMLVL|ECM |UDT |IG  |OG  |SG        |DL  |DH
   |CODERS
   |CODERSOUT
   |TI1 .. TI5
-------------------------------------------------------------------------------
0   StoS  CtiSip            CtiSip    'NumSip'             *
    NO          ANY         NO   Dft  *                    *
    64000 Sys   Sys   Sys   Sys  Sys  *                    *
---------------------------------------------------------------------------

[13:49:17] ABILIS_CPX_1:d list:numsip

LIST:NumSip               - IN
     Automatically_generated_CTI_SIP_Numbers_list_(ReadOnly)
     610                     630

We set up the SP parameter to 64000 to keep the available coder pool.

49.4.4.1.2. Communication between the analog phone and the SIP users

Add the following routings:

[17:22:52] ABILIS_CPX_1:a ctir pr:1 type:stov poo:101 cdi:01 sp:64000

COMMAND EXECUTED

[17:23:10] ABILIS_CPX_1:a ctir pr:2 type:vtos poi:pbx cdi:'numsip' sp:64000

COMMAND EXECUTED

[16:42:17] ABILIS_CPX_1:d ctir

- Not Saved (SAVE CONF), Not Refreshed (INIT) ---------------------------------
Last change: 28/05/2015 10:26:01 CET

---+-----+-----------------+---------+--------------------+--------------------
PR |[DESCR]
   |TYPE |POI/SR   [SR]    |POO/DS   |CDI                 |CDO
ACT|NEXT       |LAST       |EEC |T301|CGI                 |CGO
EDT|SP   |SC   |DJ   |MJ   |FMDJ|FMMJ|SDI                 |SDO
   |SPOUT|SCOUT|DJOUT|MJOUT|LCS |LCST|SGI                 |SGO
   |                                 |RGI                 |RGO
   |FMRLY|FAXSP|MODSP|FMLVL|ECM |UDT |IG  |OG  |SG        |DL  |DH
   |CODERS
   |CODERSOUT
   |TI1 .. TI5
-------------------------------------------------------------------------------
0   StoS  CtiSip            CtiSip    'NumSip'             *
    NO          ANY         NO   Dft  *                    *
    64000 Sys   Sys   Sys   Sys  Sys  *                    *
-------------------------------------------------------------------------------
1   StoV  CtiSip            101       01                   *
    NO          ANY         NO   Dft  *                    *
    64000 Sys   Sys   Sys   Sys  Sys  *                    *
-------------------------------------------------------------------------------
2   VtoS  PBX               CtiSip    'NumSip'             *
    NO          ANY         NO   Dft  *                    *
    64000 Sys   Sys   Sys   Sys  Sys  *                    *
-------------------------------------------------------------------------------

The PR:1 routing routes toward port 101 (where the analog phone is connected) the calls coming from one of the SIP users and directed to the number 01.

The PR:2 routing routes the calls coming from the Abilis POTS port towards the CTISIP and directed to one number of the NumSip list; the CTISIP translation table will forward the call to the user.

49.4.4.1.3. Communication between the SIP users in “Site 1” and the SIP users in “Site 2

Add a StoC and a CtoS routing with the following commands to connect up the PC with Zoiper in “Site 1” and the sip phone in “Site 2”:

[10:27:19] ABILIS_CPX_1:a ctir pr:3 type:stoc ds:site2 cdi:910 sp:64000

COMMAND EXECUTED

[10:29:50] ABILIS_CPX_1:a ctir pr:4 type:ctos sr:* cdi:'numsip' sp:64000

COMMAND EXECUTED

[16:33:00] ABILIS_CPX_1:d ctir

- Not Saved (SAVE CONF), Not Refreshed (INIT) ---------------------------------
Last change: 28/05/2015 10:30:04 CET

---+-----+-----------------+---------+--------------------+--------------------
PR |[DESCR]
   |TYPE |POI/SR   [SR]    |POO/DS   |CDI                 |CDO
ACT|NEXT       |LAST       |EEC |T301|CGI                 |CGO
EDT|SP   |SC   |DJ   |MJ   |FMDJ|FMMJ|SDI                 |SDO
   |SPOUT|SCOUT|DJOUT|MJOUT|LCS |LCST|SGI                 |SGO
   |                                 |RGI                 |RGO
   |FMRLY|FAXSP|MODSP|FMLVL|ECM |UDT |IG  |OG  |SG        |DL  |DH
   |CODERS
   |CODERSOUT
   |TI1 .. TI5
-------------------------------------------------------------------------------
0   StoS  CtiSip            CtiSip    'NumSip'             *
    NO          ANY         NO   Dft  *                    *
    64000 Sys   Sys   Sys   Sys  Sys  *                    *
-------------------------------------------------------------------------------
1   StoV  CtiSip            101       01                   *
    NO          ANY         NO   Dft  *                    *
    64000 Sys   Sys   Sys   Sys  Sys  *                    *
-------------------------------------------------------------------------------
2   VtoS  PBX               CtiSip    'NumSip'             *
    NO          ANY         NO   Dft  *                    *
    64000 Sys   Sys   Sys   Sys  Sys  *                    *
-------------------------------------------------------------------------------
3   StoC  CtiSip            site2     910                  *
    NO          ANY         NO   Dft  *                    *
    64000 Sys   Sys   Sys   Sys  Sys  *                    *
-------------------------------------------------------------------------------
4   CtoS  *                 CtiSip    'NumSip'             *
    NO          ANY         NO   Dft  *                    *
    64000 Sys   *     *     Sys  Sys  *                    *
-------------------------------------------------------------------------------

We set up the SP parameter to 64000 to keep the available coder pool.

The PR:3 routing routes towards the Site2 cluster the calls coming from SIP users and directed to the number 910.

The PR:4 routing routes towards the CTISIP the calls coming from any cluster and directed to one of the numbers of the NumSip list; the CtiSip translation table will forward the calls to the user.

49.4.4.2. Abilis CTI Routing of “Site 2

49.4.4.2.1. Communication between the SIP users in “Site 2” and the SIP users in “Site 1

The example is similar to the configuration of the Abilis of the “Site 1”; type the following commands:

[10:39:13] ABILIS_CPX_2:a ctir pr:0 type:stoc ds:site1 cdi:610 sp:64000

COMMAND EXECUTED

[10:40:10] ABILIS_CPX_2:a ctir pr:1 type:ctos sr:* cdi:'numsip' sp:64000

COMMAND EXECUTED

[10:40:17] ABILIS_CPX_2:d ctir

- Not Saved (SAVE CONF), Not Refreshed (INIT) ---------------------------------
Last change: 28/05/2015 10:31:56 CET

---+-----+-----------------+---------+--------------------+--------------------
PR |[DESCR]
   |TYPE |POI/SR   [SR]    |POO/DS   |CDI                 |CDO
ACT|NEXT       |LAST       |EEC |T301|CGI                 |CGO
EDT|SP   |SC   |DJ   |MJ   |FMDJ|FMMJ|SDI                 |SDO
   |SPOUT|SCOUT|DJOUT|MJOUT|LCS |LCST|SGI                 |SGO
   |                                 |RGI                 |RGO
   |FMRLY|FAXSP|MODSP|FMLVL|ECM |UDT |IG  |OG  |SG        |DL  |DH
   |CODERS
   |CODERSOUT
   |TI1 .. TI5
-------------------------------------------------------------------------------
0   StoC  CtiSip            site1     610                  *
    NO          ANY         NO   Dft  *                    *
    64000 Sys   Sys   Sys   Sys  Sys  *                    *
-------------------------------------------------------------------------------
1   CtoS  *                 CtiSip    'NumSip'             *
    NO          ANY         NO   Dft  *                    *
    64000 Sys   *     *     Sys  Sys  *                    *
-------------------------------------------------------------------------------


[13:49:17] ABILIS_CPX_2:d list:numsip

LIST:NumSip               - IN
     Automatically_generated_CTI_SIP_Numbers_list_(ReadOnly)
     910

Like the previous chart, the SP parameters is set at 64000 to keep the available coder pool.

The PR:0 routing routes toward the “Site1”cluster the calls coming from SIP users and directed to the number 610.

The PR:1 routing routes toward the CTISIP the calls coming from any cluster and directed to one of the numbers of the NumSip list; the CTISIP translation table will forward the calls to the user.