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 isn't
updated.
For example:
[18:19:55] ABILIS_CPX:d ctir
Last change: 03/09/2015 15:02:30 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 VOICE # * # PBX ?? *
TRANSFERT ANY NO Dft * *
-------------------------------------------------------------------------------
1 VOICE # * # PBX ?? 22
TRANSFERT BUSY NO Dft * *
-------------------------------------------------------------------------------
2 VOICE # * # PBX ?? 01
TRANSFERT NOANS NO Dft * *
-------------------------------------------------------------------------------
3 VOICE # * # DISA ?? 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 the 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.
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 the
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:28Last change: 03/09/2015 15:02:30 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 ------------------------------------------------------------------------------- 28 VOICE # * # G1 'technicians' * TRANSFERT ANY NO 15 * * 6400 Sys Sys Sys Sys Sys * * ------------------------------------------------------------------------------- [18:15:21] ABILIS_CPX:d ctig----------------------------------+------------------------------------------------ ID: [DESCR:] S: R: MC: P: |P1 P2 P3 P4 P5 ... |... P62 P63 P64 [Px: CDO: CGO: SDO: SGO: RGO: SP: CODERS: DJ: MJ: T301: ] ----------------------------------+------------------------------------------------ 0 [Iax/Sip/Disa/Vo group (Read Only)] R ST MAX NO |Iax Sip Disa Vo . ----------------------------------+------------------------------------------------ 1 M UN MAX YES |104 110 108 . . ----------------------------------+------------------------------------------------
![]() | Note |
|---|---|
If the |
By default, the POTS port numbers are 2 digits long; it's possible
to change their length by modifying the
POTS-NUM-LENGTH parameter in the CTISYS resource
(available values are: [1..20]). 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. |
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. |
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
Last change: 03/09/2015 15:02:30 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
-------------------------------------------------------------------------------
11 VOICE G1 # # G2 * *
NO ANY NO Dft * #
64000 Sys Sys Sys Sys Sys * *
-------------------------------------------------------------------------------
In the alternative, if the call is incoming from a POTS port which
is configured with SS:YES and
SS-PRES-CG:YES, it's 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:BSE-1<9> USER:06
Run OPSTATE:UP LOG:NO TYPE:USER
signalling:POTS HOLD:YES CT:ICT SS:YES
DEVICE:PHONE AC:NO
NUM:06 ADDRBOOK-NUM:NUM
AC-CDO:# AC-DLY:60
LOOP:NO TEST:NO
CLIP:SYS CLIP-STD:SYS CLIP-TAS:SYS CLIP-LEVEL:SYS
SENSING:SYS ABF:SYS HPF:SYS DEBOUNCE:SYS MIN-FLASH:SYS
COUNTRY:SYS MAX-FLASH:SYS
DIALT:5 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:GROUPS SS-PRES-CG:YES NPOO-CT:SYS
SS-CF-DND:YES SS-VM:YES![]() | Caution |
|---|---|
In this case the |
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 46.12.3.5, “Fm-relay parameter”. |
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 - - - -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 |
VtoCISDA | 0 | 0 | 0 | 0 |
CISDAtoCISDA | 0 | 0 | 0 | 0 |
CISDAtoV | 0 | 0 | 0 | 0 |
ALL | 0 | 0 | 0 | 0 |
------------------------------------------------------------------------
- AC and SWAC common available coders ----------------------------------
-- Coder ---|-- Bit rates (kbps) --|-- Coder ---|-- Bit rates (kbps) --|
G.711A |64 |G.711u |64 |
G.723.1 |5.3, 6.3 |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 |
------------------------------------------------------------------------
- SWAC and MCD limits by CPU -------------------------------------------
MAX-SWAC-0ms:5 MAX-SWAC-8ms:4 MAX-SWAC-16ms:4 MAX-SWAC-32ms:4
MAX-MCD-SPIRIT:5 MAX-MCD-G729A:6
- SWAC and MCD diagnostics ---------------------------------------------
CUR-SWAC:0 PEAK-SWAC:0 REST-SWAC:0 MAX-SWAC:0 LIMIT-SWAC:CFG
CUR-MCD:0 PEAK-MCD:0 MAX-HDLC:8
------------------------------------------------------------------------
- Clock Sources for H100 cards -----------------------------------------
CLK:INT
- Clock Sources for NOT-H100 cards -------------------------------------
--- CARD ---|- CLK -|
BSE-1 | INT |
------------------------------------------------------------------------It's 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's 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. |
| 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 |
| 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 |
![]() | Note |
|---|---|
1 DSP manages 4 channels. |
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
OUT:<port_number> (e.g.
POI:32,
OUT:32) before the CTI Routing
with POI:*.
[18:27:15] ABILIS_CPX:d ctir
Last change: 03/09/2015 15:02:30 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 VOICE 32 # # 32 * *
NO ANY NO Dft * *
64000 Sys Sys Sys Sys Sys * *
--------------------------------------------------------------------------------
1 VOICE * # # 32 * *
NO ANY NO Dft * *
64000 Sys Sys Sys Sys Sys * *
-------------------------------------------------------------------------------
SIP multialerting is implemented. To configure SIP multialerting is need to create heterogeneous groups of SIP phones.
![]() | Tip |
|---|---|
Refer to chapter Section 72.5, “How to configure a multicast group of SIP phones”. |