69.4. CTI system

69.4. CTI system
	Chapter 69. FAQ - Frequently Asked Questions

69.4.1. How can I manage alternative CTI routings?

If the called phone is busy or not responding, the call is forwarded to the next routing if in the current one the NEXT parameter is set to TRANSFERT.

When the parameter LAST is set to BUSY or NOANS or OTHER the routings can match only if the last failure reason matches the one specified in LAST.

Note that:

routing with LAST <> ANY can match only if a previous routing failed; they will never match as first routing;
when a routing with LAST <> ANY fails, the original failure reason is not updated.

For example:

[18:19:55] ABILIS_CPX:d ctir

--+-----+-----------------+---------+--------------------+--------------------
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   CtoV  #                 PBX       ??                   *
    TRANSFERT   ANY         NO   Dft  *                    *
-------------------------------------------------------------------------------
1   CtoV  #                 PBX       ??                   22
    TRANSFERT   BUSY        NO   Dft  *                    *
-------------------------------------------------------------------------------
2   CtoV  #                 PBX       ??                   01
    TRANSFERT   NOANS       NO   Dft  *                    *
-------------------------------------------------------------------------------
3   CtoD  #                 CtiDisa   ??                   99
    NO          OTHER       NO   Dft  *                    *
-------------------------------------------------------------------------------

PR:0 is main call.
If PR:0 fails with BUSY reason PR:1 is executed, e.g. call sent to a colleague.
If PR:0 fails with NOANS reason PR:2 is executed, e.g. call sent to PBX main operator.
If PR:0 fails with OTHER reasons (other than BUSY and NOANS) PR:3 is executed, e.g. call sent to a DISA group that plays a message telling that call could not be delivered.

69.4.2. How does Round Robin scanning method work in CTI Port Group?

If in a CTI group the S parameter is set to R, the incoming calls are directed towards the CTI ports in a circular manner; e.g. the first call is forwarded to the port set in P1 parameter, the second call is forwarded to the port set in P2 parameter, etc...

If a port is busy or not responding, the call is forwarded to next port if the R parameter is set to UN (unconditional). In case of “no answer” the call is forwarded to the next port after the time interval set in the T301 parameter; in case of “busy” the call is immediately forwarded to the next port.

For example:

[18:19:55] ABILIS_CPX:d ctir pr:38

---+-----+-----------------+---------+--------------------+--------------------
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
-------------------------------------------------------------------------------
28  CtoV  *                 G1        'technicians'        *
    TRANSFERT   ANY         NO   15   *                    *
    6400  Sys   Sys   Sys   Sys  Sys  *                    *
-------------------------------------------------------------------------------


[18:15:21] ABILIS_CPX:d ctig

--------------------------------------+--------------------+-------------------
 [DESCR:]
 ID: PICKUP: SCAN: RULE:              |P1  P2  P3  P4  P5  |P6  P7  P8  P9  ...
                                      |... P62 P63 P64
--------------------------------------+--------------------+-------------------
 1   YES     M     UN                 |104 110 108 .   .   |.   .   .   .   .  
--------------------------------------+--------------------+-------------------

	Note
	If the `T301` parameter is set to the default value (180 sec.), the call is not forwarded to the next port.

69.4.3. How can I change the POTS number length?

By default, the POTS port numbers are 2 digits long; it is possible to change their length by modifying the POTS-NUM-LENGTH parameter in the CTISYS resource (available values are: 1,2,3 or 4). For example:

s p ctisys pots-num-length:3	Change the `NUM` parameter length to 3.
save conf	Save the configuration.
init ctisys	Initialize the CTISYS resource.

69.4.4. How can I activate the Caller Identification Presentation (CLIP) on POTS ports?

Set the CLIP parameter to YES in the CTISYS resource.

s p ctisys clip:yes	Activate the Caller Identification Presentation.
save conf	Save the configuration.
init ctisys	Initialize the CTISYS resource.

	Tip
	Refer to this paragraph to know more about CLIP parameter.

69.4.5. How can I manage the Calling Line Identification (CLI) for outcoming calls?

Calling Line Identification can be statically managed in the CGO parameter in CTI Routings:

CGO:# : Set an empty information element. An empty information element in most cases is removed.

[18:19:55] ABILIS_CPX:d ctir pr:11

---+-----+-----------------+---------+--------------------+--------------------
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
-------------------------------------------------------------------------------
1   TR    G1                G2        *                    *
    NO          ANY         NO   Dft  *                    #
-------------------------------------------------------------------------------

In alternative, if the call is incoming from a POTS port which is configured with SS:YES and SS-PRES-CG:YES, it is possible to force CLIR/CLIP typing the following codes:

*60#<number-to-dial>: make a call to <number-to-dial> hiding the calling number;
*61#<number-to-dial>: make a call to <number-to-dial> presenting the calling number.

[22:47:50] ABILIS_CPX:d p ctipe:106

CTIP:106 DESCR:
Act     card:POTS-1<6>
        OPSTATE:UP        LOG:NO                               TYPE:USER
        signalling:POTS                 HOLD:YES     CT:ICT    SS:YES
        mode:Pots                                    NUM:06
        LOOP:NO         TEST:NO
        CLIP:SYS        CLIP-STD:SYS      CLIP-TAS:SYS    MIN-FLASH:SYS
        SENSING:SYS     ABF:SYS  HPF:SYS  DEBOUNCE:SYS    MAX-FLASH:SYS
        COUNTRY:SYS                  IN-GAIN-ADJ:SYS      OUT-GAIN-ADJ:SYS
        AT:SYS          AT-CODE:SYS  AT-DURATION:SYS      BC:SYS
        DT:SYS          DT-CODE:SYS  DT-DURATION:SYS      OUT-DIAL-TONE:SYS
        SS-PICKUP:GROUP SS-PRES-CG:YES                    NP-CT:SYS
        SS-CF-DND:YES

	Caution
	In this case the `CGO` parameter in the CTIR routing must not be set to `#`.

69.4.6. Why during a call do I hear a fax handshaking?

During the compressed call, the calling or called phone intercepts the handshaking of a fax located near the phone. The DSP used by Abilis identifies this signal as a request of fax transmission and starts to simulate the fax modulation.

The possible solutions are:

mute the fax volume;
disable the fax relay feature (FM-RELAY:NO in the setting of CTISYS resource, or FMRLY:NO in the specific CTIR routings).

	Tip
	Refer to chapter Section 43.12.3.5, “Fm-relay parameter”.

69.4.7. How can I view the maximum number of simultaneous calls supported by Abilis?

To view the maximum number of simultaneous calls supported by Abilis type the command: d d ctiac or d de ctiac for the extended mode.

For example the following Abilis allows up to 8 simultaneous calls.

[16:16:35] ABILIS_CPX:d d ctiac

-------------------------------------------------------------------------------
AC   Card    DSP/C Bus/TS  DSPState ACState  ModeIn  ModeOut Coder      Ctip/BC
-------------------------------------------------------------------------------
  0  BRI4-2    0/0   8/00  RUN      IDLE     -       -       -                -
  1  BRI4-2    0/1   8/01  RUN      IDLE     -       -       -                -
  2  BRI4-2    0/2   8/02  RUN      IDLE     -       -       -                -
  3  BRI4-2    0/3   8/03  RUN      IDLE     -       -       -                -
  4  BRI4-2    1/0   9/05  RUN      IN-USE   VOICE   VOICE   Spirit/6.4k 108/01
  5  BRI4-2    1/1   9/06  RUN      IDLE     -       -       -                -
  6  BRI4-2    1/2   9/07  RUN      IDLE     -       -       -                -
  7  BRI4-2    1/3   9/08  RUN      IDLE     -       -       -                -

69.4.8. How can I view the available coders supported by the cards installed in Abilis?

To view the available coders supported by CTI cards installed in Abilis type the command: d d ctisys.

[16:32:16] ABILIS_CPX:d d ctisys

RES:CtiSys --------------------------------------------------------------------
       CTI_System_general_properties                                           
       CTIR-STATE:ENABLED     CALLS-CURRENT:0      CALLS-PEAK:0
       AC-STATE:ENABLED       AC-CURRENT:0         AC-PEAK:0
       ------------------------------------------------------------------------
       -- Number of simultaneous calls ----------------------------------------
                      | State: Alerting/Connected |        State: Any         |
       ---------------|---Current---|----Peak-----|---Current---|----Peak-----|
       TR             |           0 |           0 |           0 |           0 |
       DATA           |           0 |           0 |           0 |           0 |
       VtoCHISD       |           0 |           0 |           0 |           0 |
       CHISDtoCHISD   |           0 |           0 |           0 |           0 |
       CHISDtoV       |           0 |           0 |           0 |           0 |
       ALL            |           0 |           0 |           0 |           0 |
       ------------------------------------------------------------------------
       - Available coders -----------------------------------------------------
       -- Coder ---|-- Bit rates (kbps) --|-- Coder ---|-- Bit rates (kbps) --|
       G.711A      |64                    |G.711u      |64                    |
       G.726       |16, 24, 32, 40        |G.729A      |8                     |
       TRANSPARENT |64                    |Spirit      |6.4, 7.2, 8, 8.8, 9.6 |
       G.727       |16/16, 24/16, 24/24,  |
                   |32/16, 32/24, 32/32,  |
                   |40/16, 40/24, 40/32   |
       ------------------------------------------------------------------------
       - Clock Sources for H100 cards -----------------------------------------
       CLK:34 
       - Clock Sources for NOT-H100 cards -------------------------------------
                          *** No NOT-H100 cards running ***

69.4.9. How can I change the volume of a compressed call?

It is possible to modify the OUT-GAIN parameter in the CTISYS resource. For example:

s p ctisys out-gain:+3	Change the output gain in the range [MUTE, -31..+31 dB].
save conf	Save the configuration.
init ctisys	Initialize the CTISYS resource.

For the phones connected to POTS cards it is possible to modify the OG parameter in the specific CTI Routing. For example:

s ctir pr:5 og:+5	Change the output gain in the range [SYS, MUTE, -31..+31 dB].
save conf	Save the configuration.
init ctir	Initialize the CTI Routings.

69.4.10. How many VoIP channels are used, depending on the call's source and destination?

Codec used by the call coming from VoIP telephones in the LAN	Abilis routes the call toward	VoIP channels occupied	Bandwidth occupied by each channel (Kbit/s)
SIP Codec G.711	Telecom ISDN network	1	64
SIP Codec G.729	Telecom ISDN network	1	64
SIP Codec G.711	Abilis over ISDN network	2	9
SIP Codec G.729	Abilis over ISDN network	2	9
SIP Codec G.711	VoIP provider with G.729	2	32
SIP Codec G.729	VoIP provider with Abilis Codec	2	30 the first call, 9 the others calls
SIP Codec G.711	VoIP provider with G.729	0	32
SIP Codec G.729	VoIP provider with Abilis Codec	2	30 the first call, 9 the others calls
SIP Codec G.711	Abilis GSM Box	0	1 GSM channel
SIP Codec G.729	Abilis GSM Box	1	1 GSM channel
IAX2 Codec G.711	Telecom ISDN network	1	64
IAX2 Codec G.729	Telecom ISDN network	1	64
IAX2 Codec G.711	Abilis over ISDN network	2	9
IAX2 Codec G.729	Abilis over ISDN network	2	9
IAX2 Codec G.711	VoIP provider with G.729	2	32
IAX2 Codec G.729	VoIP provider with Abilis Codec	2	30 the first call, 9 the others calls
IAX2 Codec G.711	VoIP provider with G.729	0	32
IAX2 Codec G.729	VoIP provider with Abilis Codec	2	30 the first call, 9 the others calls
IAX2 Codec G.711	Abilis GSM Box	0	1 GSM channel
IAX2 Codec G.729	Abilis GSM Box	1	1 GSM channel

	Note
	1 DSP manages 4 channels.

69.4.11. What happens if a call fails with the “CAUSE:FF B4 (CPX,Loop)”?

When a call ends with the disconnection code CAUSE:FF B4 (CPX,Loop), Abilis blocks the calls which enter and exit from the same ISDN port, when in the CTI Routing the POI parameter is set to *.

To enable the loop, you must add a CTI Routing with POI:<port_number> and POO:<port_number> (e.g. POI:32, POO:32) before the CTI Routing with POI:*.

[18:27:15] ABILIS_CPX:d ctir

---+-----+-----------------+---------+--------------------+--------------------
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   DATA  *                 CtiSLink  *                    *
-------------------------------------------------------------------------------
1   TR    32                32        *                    *
-------------------------------------------------------------------------------
2   TR    *                 32        *                    *
-------------------------------------------------------------------------------

69.4.12. How can i configure sip multialerting?

Sip multialerting is not yet implemented. The only way to configure sip multialerting is to use forking which let to create heterogeneous groups of phones.


69.3. ADSL/HDSL lines		69.5. IP router