49.5. Connecting Abilis and a SIP server

49.5. Connecting Abilis and a SIP server
	Chapter 49. SIP - Session Initiation Protocol

This section contains instructions for a correct set-up of Abilis CPX and SIP server interconnection.

49.5.1. Activation of the CTISIP resource

For the activation of the CTISIP resource refer to Section 49.1.1, “Activating the CTISIP resource”.

The basic parameters to configure are:

ACT: to activate the resource.
sesnum: to define the amount of simultaneous connections.

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].

	Tip
	If Abilis has only one IP resource (and only one IP address), you can use the default value; otherwise if Abilis has more IP resources and more IP addresses the suggested configuration is `OUT-IP`.

DOMAIN: if Abilis has clients in the public side you can also specify a FQDN.

49.5.2. Example: Abilis and a SIP server registered in Abilis domain (SIP-TYPE:LOCAL-PEER)

In this case the server is a “normal” user like a soft phone, but the SIP-TYPE is LOCAL-PEER.

	Note
	`LOCAL-PEER` handling is similar to `PHONE` but it allows calling/called number to pass unchanged.

In the figure there are the following elements:

SIP server has dynamic IP;
Abilis has only one SIP user;
Abilis user is “test” with password “secret”;
Tip
Interesting chapters:
Section 49.2.2, “CTISIP INVITE messages”.
Section 72.5, “How to configure a multicast group of SIP phones”.

Now you have to create a SIP user representing the user that is a client of the Abilis.

[14:49:07] ABILIS_CPX:a user:test pwd:secret sip:yes

COMMAND EXECUTED

[14:49:07] ABILIS_CPX:s user:test sip-type:local-peer sip-number:* sip-host:dynamic

COMMAND EXECUTED

	Note
	The SIP server isn't required to be only on the local network.

You can show the result in this way:

[14:49:07] ABILIS_CPX:d user:test

- Not Saved (SAVE CONF) -------------------------------------------------------
Parameter:          | Value:
--------------------+----------------------------------------------------------
USER:                 test
REAL-NAME:            test
ID:                   13            <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:          NONE
OPC-PRIVACY:          NO
CHAT:                 NO    
CHAT-USER:            SYS
CHAT-PWD:             SYS
SIP:                  YES   
SIP-TYPE:             LOCAL-PEER    
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            
-------------------------------------------------------------------------------

The CTISIP table, used to route calls toward SIP users, gets automatically populated with a unique route because you set SIP-NUMBER:*.

[14:49:07] ABILIS_CPX:d ctisip numbers

Total:1        Sip-Number:1         Static:0 
        
NUMx: [SIP-NUMBER:]       USER:                             Provenience:
------------------------------------------------------------------------
[*]                       test                                SIP-NUMBER

49.5.3. Example: Server and Abilis registered in Server remote domain (SIP-TYPE:SERVER)

In the figure there are the following elements:

Abilis is a user of SIP proxy domain “voip.it”;
“voip.it” has s static IP address: 88.88.88.88;
Abilis has a static IP address;
Abilis has only one SIP user;
Abilis user is “voipclient” with password “swordfish”.
SIP server provides advanced services like IVR and voice mail, let's say that 10 sessions are needed.
Tip
Interesting chapters:
Section 49.2.2, “CTISIP INVITE messages”.
Section 72.5, “How to configure a multicast group of SIP phones”.

Now you have to create a SIP user representing the user that is a client of voip.it SIP server:

[14:49:07] ABILIS_CPX:a user:voipclient pwd:swordfish sip:yes

COMMAND EXECUTED

[14:49:07] ABILIS_CPX:s user:voipclient sip-type:server sip-domain:voip.it

COMMAND EXECUTED

[14:49:07] ABILIS_CPX:s user:voipclient sip-host:88.88.88.88

COMMAND EXECUTED

[14:49:07] ABILIS_CPX:s user:voipclient sip-maxses-bid:10 sip-number:*

COMMAND EXECUTED

[14:49:07] ABILIS_CPX:s user:voipclient sip-rem-reg:yes sip-rem-user:voipclient sip-rem-pass:swordfish

COMMAND EXECUTED

You can show the result in this way:

[14:49:07] ABILIS_CPX:d user:voipclient

- Not Saved (SAVE CONF) -------------------------------------------------------
Parameter:          | Value:
--------------------+----------------------------------------------------------
USER:                 voipclient
REAL-NAME:            voipclient
ID:                   8             <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:             SERVER        
SIP-DOMAIN:           voip.it
SIP-HOST:             088.088.088.088
SIP-TCP-REMPORT:      5060
SIP-UDP-REMPORT:      5060
sip-udp-locport:      SYS
SIP-SRCADD:           SYS
SIP-PROT:             UDP
SIP-IP-PERMIT:        *
SIP-MAXSES-BID:       10
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:         voipclient
SIP-REM-PASS:         ********
SIP-REM-AUTH:         SYS
SIP-REM-REG:          YES           
-------------------------------------------------------------------------------

The CTISIP table, used to route calls toward SIP users, gets automatically populated with a unique route because you set SIP-NUMBER:*.

[14:49:07] ABILIS_CPX:d ctisip numbers

Total:1        Sip-Number:1         Static:0 

NUMx: [SIP-NUMBER:]       USER:                             Provenience:
------------------------------------------------------------------------
[*]                       voipclient                          SIP-NUMBER

Abilis and the Sip Server interconnection is now correctly configured.

	Note
	`REMOTE-PEER` handling is similar to `SERVER` but it allows calling/called number to pass unchanged.

49.5.4. Example: Server and Abilis registered in Server remote domain (SIP-TYPE:REMOTE-PEER)

In the figure there are the following elements:

Abilis is a user of SIP server with static IP address:88.88.88.88.
Abilis has a static IP address.
Abilis has only one SIP user.
Abilis user is “voipclient” with password “swordfish”.
The user “voipclient” is associated at the SIP number 5678.
Tip
Interesting chapter:
Section 49.2.2, “CTISIP INVITE messages”.
Section 72.5, “How to configure a multicast group of SIP phones”.

Now you have to create a SIP user representing the user that is a client of SIP server

[14:49:07] ABILIS_CPX:a user:voipclient pwd:swordfish sip:yes

COMMAND EXECUTED

[14:49:07] ABILIS_CPX:s user:voipclient sip-type:remote-peer sip-host:88.88.88.88

COMMAND EXECUTED

[14:49:07] ABILIS_CPX:s user:voipclient sip-number:5678 sip-rem-reg:yes sip-rem-user:voipclient sip-rem-pass:swordfish 

COMMAND EXECUTED

[14:49:07] ABILIS_CPX:s user:voipclient sip-cg-num:*

COMMAND EXECUTED

	Note
	The `SIP-CG-NUM` parameter is set to "`*`" to pass calling number transparently, because by default, this parameter is `AUTO` (enforces caller id information element equal to SIP-NUMBER).

You can show the result in this way:

[14:49:07] ABILIS_CPX:d user:voipclient

- Not Saved (SAVE CONF) -------------------------------------------------------
Parameter:          | Value:
--------------------+----------------------------------------------------------
USER:                 voipclient
REAL-NAME:            voipclient
ID:                   8             <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:             REMOTE-PEER       
SIP-DOMAIN:           
SIP-HOST:             088.088.088.088
SIP-TCP-REMPORT:      5060
SIP-UDP-REMPORT:      5060
sip-udp-locport:      SYS
SIP-SRCADD:           SYS
SIP-PROT:             UDP
SIP-IP-PERMIT:        *
SIP-MAXSES-BID:       10
SIP-MAXSES-IN:        0
SIP-MAXSES-OUT:       0
SIP-NUMBER:           5678
SIP-ADDRBOOK-NUM:     SIP-NUMBER
SIP-CG-NUM:           *
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:         voipclient
SIP-REM-PASS:         ********
SIP-REM-AUTH:         SYS
SIP-REM-REG:          YES           
-------------------------------------------------------------------------------

The CTISIP table, used to route calls toward SIP users, gets automatically populated with a unique route because you set SIP-NUMBER:5678.

[14:49:07] ABILIS_CPX:d ctisip numbers

Total:1        Sip-Number:1         Static:0 

NUMx: [SIP-NUMBER:]       USER:                             Provenience:
------------------------------------------------------------------------
[5678]                    voipclient                          SIP-NUMBER

It's need to add a static SIP translation route to add several numbers to this user.

Use the following command to add a new SIP-NUMBER:

[14:49:07] ABILIS_CPX:a ctisip numbers numx:* user:voipclient

COMMAND EXECUTED

[18:05:18] ABILIS_CPX:d ctisip numbers

Total:2         Sip-Number:1         Static:1         

NUMx: [SIP-NUMBER:]       USER:                             Provenience:
------------------------------------------------------------------------
[5678]                    voipclient                          SIP-NUMBER
 *                        voipclient                              STATIC

Abilis and the SIP Server interconnection is now correctly configured.

	Note
	`REMOTE-PEER` handling is similar to `SERVER` but it allows calling/called number to pass unchanged.

49.5.5. CTI Routings configuration

Some routings in the CTIR table must be added in order to route the calls to and from the CTISIP resource.

49.5.5.1. Any coder, transcoding disallowed

Purpose of configuration: calls arriving from ISDN/POTS/GSM/CLUSTER are routed to SIP users, and calls arriving from SIP users are first sent to cluster test; in case of failure (NEXT:LIMITED) it's attempted on ISDN/POTS/GSM group G1.

In this situation any coder with maximal speed 6400 (the default for SP parameter) is allowed, but transcoding is disallowed. This means that the same coder must be used by the SIP proxy and Abilis.

[18:05:14] ABILIS_CPX:a ctir pr:0 poi:* out:sip cdi:* descr:From_ISDN/POTS/GSM_to_SIP 

COMMAND EXECUTED

[18:05:18] ABILIS_CPX:a ctir pr:1 sr:* out:sip cdi:* descr:From_Cluster_to_SIP

COMMAND EXECUTED

[18:05:30] ABILIS_CPX:a ctir pr:2 poi:sip out:test cdi:* next:limited descr:From_SIP_to_Cluster

COMMAND EXECUTED

[18:05:38] ABILIS_CPX:a ctir pr:3 poi:sip out:g1 cdi:* descr:From_SIP_to_ISDN/POTS/GSM

COMMAND EXECUTED

[18:05:51] ABILIS_CPX:d ctir

- Not Saved (SAVE CONF), Not Refreshed (INIT) ---------------------------------
Last change: 17/06/2015 10:01:34 CET

---+------+-----------------+---------+--------------------+--------------------
PR |[DESCR]
   |BCI   |POI |SR      |GI |OUT      |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
   |                        |BCO      |RGI                 |RGO
   |FMRLY |FAXSP|MODSP|FMLVL|ECM |UDT |IG  |OG  |SG        |DL  |DH
   |CODERS
   |CODERSOUT
   |TI1 .. TI5
--------------------------------------------------------------------------------
0   [From_ISDN/POTS/GSM_to_SIP]
    VOICE  *    #        #   Sip       *                    *                   
--------------------------------------------------------------------------------
1   [From_Cluster_to_SIP]
    VOICE  #    *        #   Sip       *                    *                   
--------------------------------------------------------------------------------
2   [From_SIP_to_Cluster]
    VOICE  Sip  #        #   test      *                    *                   
    LIMITED      ANY         NO   Dft  *                    *                   
--------------------------------------------------------------------------------
3   [From_SIP_to_ISDN/POTS/GSM]
    VOICE  Sip  #        #   G1        *                    *                   
--------------------------------------------------------------------------------

	Note
	The routes `pr`:`0` and `pr`:`1` may be unified into one: *a ctir pr:0 poi: sr:* out:sip cdi:* descr:From_ISDN/POTS/GSM/Cluster_to_SIP**

	Note
	The `POI`:`` in CTIR routings identifies all CTIP ports: ISDN, POTS, VPOTS, CELL; and SR: identifies all clusters. Refer to Table 46.2, “Special characters and values available in CTI routing table” to know more about special characters and values available in CTI routing.

	Tip
	To allow G.729A you have to set `SP`:`8000` in every routing.

49.5.5.2. Only G.711 on SIP proxy, any coder on Cluster, transcoding allowed

Purpose of example: calls arriving from ISDN/POTS/GSM/CLUSTER are routed to SIP users, and calls arriving from SIP users are first sent to cluster test; in case of failure (NEXT:LIMITED) it's attempted on ISDN/POTS/GSM group G1.

In this situation only G.711 A-law or u-law can be used by SIP proxy and Abilis. Since transcoding is enabled by CODERSOUT <> * the CTI routings will negotiate for the “C” side any coder with maximum speed up 6400 bps.

[18:12:28] ABILIS_CPX:a ctir pr:0 poi:* out:sip cdi:* sp:64000 descr:From_ISDN/POTS/GSM_to_SIP

COMMAND EXECUTED

[18:12:37] ABILIS_CPX:a ctir pr:1 sr:* out:sip cdi:* spout:64000 codersout:G.711 descr:From_Cluster_to_SIP

COMMAND EXECUTED

[18:12:45] ABILIS_CPX:a ctir pr:2 poi:sip out:test cdi:* next:limited sp:64000 coders:g.711 spout:6400 codersout:*,sys descr:From_SIP_to_Cluster

COMMAND EXECUTED

[18:12:53] ABILIS_CPX:a ctir pr:3 poi:sip out:g1 cdi:* sp:64000 descr:From_SIP_to_ISDN/POTS/GSM

COMMAND EXECUTED

[18:13:00] ABILIS_CPX:d ctir

- Not Saved (SAVE CONF), Not Refreshed (INIT) ---------------------------------
Last change: 17/06/2015 10:01:34 CET

---+------+-----------------+---------+--------------------+--------------------
PR |[DESCR]
   |BCI   |POI |SR      |GI |OUT      |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
   |                        |BCO      |RGI                 |RGO
   |FMRLY |FAXSP|MODSP|FMLVL|ECM |UDT |IG  |OG  |SG        |DL  |DH
   |CODERS
   |CODERSOUT
   |TI1 .. TI5
--------------------------------------------------------------------------------
0   [From_ISDN/POTS/GSM_to_SIP]
    VOICE  *    #        #   Sip       *                    *                   
    NO           ANY         NO   Dft  *                    *                   
    64000  Sys   Sys   Sys   Sys  Sys  *                    *                   
--------------------------------------------------------------------------------
1   [From_Cluster_to_SIP]
    VOICE  #    *        #   Sip       *                    *                   
    NO           ANY         NO   Dft  *                    *                   
    6400   Sys   Sys   Sys   Sys  Sys  *                    *                   
    64000  *     *     *     NO   Sys  *                    *                   
                             *         *                    *                   
    Sys    AUTO  AUTO  Sys   SYS  NO   Sys  Sys  Sys        Sys  Sys
    Sys
    G.711
--------------------------------------------------------------------------------
2   [From_SIP_to_Cluster]
    VOICE  Sip  #        #   test      *                    *                   
    LIMITED      ANY         NO   Dft  *                    *                   
    64000  Sys   Sys   Sys   Sys  Sys  *                    *                   
    6400   *     *     *     NO   Sys  *                    *                   
                             *         *                    *                   
    Sys    AUTO  AUTO  Sys   SYS  NO   Sys  Sys  Sys        Sys  Sys
    G.711
    *,Sys
--------------------------------------------------------------------------------
3   [From_SIP_to_ISDN/POTS/GSM]
    VOICE  Sip  #        #   G1        *                    *                   
    NO           ANY         NO   Dft  *                    *                   
    64000  Sys   Sys   Sys   Sys  Sys  *                    *                   
--------------------------------------------------------------------------------

	Tip
	To allow G.729A you have to set `SP`:`8000` in `PR`:`1` and `SPOUT`:`8000` in `PR`:`2`.

49.5.5.2.1. Transcoding optimization

When the Abilis-SIP proxy interconnection occurs via local LAN, (i.e. With high speed, minimal delays, minimal jitter), optimizing the transcoding can be done so that the SIP proxy side uses minimal jitter, minimal delays.

This is obtained by properly setting DJ, MJ, DJOUT, MJOUT.

[18:15:17] ABILIS_CPX:s ctir pr:1 djout:0 mjout:80

COMMAND EXECUTED

[18:15:29] ABILIS_CPX:s ctir pr:2 dj:0 mj:80 djout:sys mjout:sys

COMMAND EXECUTED

	Tip
	`DJ` and `MJ` in `pr`:`1` as well as `DJOUT` and `MJOUT` in `pr`:`2` may assume other values appropriate for the WAN link or specifically required by calls matching the routing.

49.5.5.2.2. Fax

When transcoding takes place in CTI routing table, with G.711 toward the SIP proxy, something interesting happens: on the WAN FAX relay can be used! The Abilis can exchange FAX with following characteristics:

UIse G.711, 64 kbps plus IP overhead on the Abilis-SIP proxy interconnection;
Use G3 Fax relay, 2400/4800/9600/14400 kbps plus IP overhead on the WAN link.

Set FMRELAY:NO in the desired routing to disable fax relay:

[18:18:2] ABILIS_CPX:s ctir pr:0 fmrelay:no

COMMAND EXECUTED

[18:18:35] ABILIS_CPX:s ctir pr:1 fmrelay:no

COMMAND EXECUTED

[18:18:29] ABILIS_CPX:s ctir pr:2 fmrelay:no

COMMAND EXECUTED

[18:15:29] ABILIS_CPX:s ctir pr:3 fmrelay:no

COMMAND EXECUTED


49.4. Connecting Abilis and a SIP phone		49.6. Busy Lamp Field (BLF)