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 the main
CTIR.
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.
Tip | |
---|---|
Interesting chapter: Section 47.5, “CTI Router Overview”. |
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:28
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 ------------------------------------------------------------------------------- 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/VPOTS port numbers are 2 digits long; it's
possible to change their length by modifying the
NUM-LENGTH
parameter in the CTISYS resource
(available values are: [1..20]). For example:
s p ctisys 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 Calling Line 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 |
Set the DISPLAY-NAME
parameter to
ADDRBOOK
in the CTISIP resource.
s p ctisip display-name:addrbook | Activate to view the caller's name on the display of SIP phones. It searches the calling number in the user address book. |
save conf | Save the configuration. |
init res:ctisip | Initialize the CTISIP resource. |
Tip | |
---|---|
Refer to this paragraph to know more about Abilis Address Book. |
"Echo" consists in outgoing voice which is returned back to the talker with a delay of more than (say) 50 milliseconds. Indeed, echo is another annoying aspect of VoIP. With ISDN (TDM) networks the problem is not there because the round-trip delay stays within a few milliseconds.
About "Echo" there are a few basic concepts to keep in mind:
Echo originates from an unwished feedback loudspeaker-microphone and it starts to be perceivable when the roundtrip of the vocal path exceeds 20/30 msec, as it is the case of all VoIP systems.
When Echo occurs, the problem is not on the side of the listener, it is on the side of the talker (the feedback takes place there).
Such feedback -to some extent- is always there. Some telephone systems suppress it well, others make it worse. Besides the mechanical/electrical reduction of the feedback there is an algorithmic way to cancel it, when feedback is within certain limits.
when the signal is distorted or the feedback is too strong or too delayed (as with smartphones in handsfree operation) there is no way to cancel the echo.
Keeping these concepts in mind, I think that you can find where the problem resides. If solutions can be found within our range of control, we will surely help.
Note | |
---|---|
Abilis-units are already suppressing the echo in a flawless way. |
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 47.13.3.5, “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 needed to create heterogeneous groups of SIP phones.
Tip | |
---|---|
Refer to chapter Section 73.5, “How to configure a multicast group of SIP phones”. |
The transfer of calls is allowed only for users of the same SIP-DOMAIN.
Please configure the same SIP-DOMAIN
for all
SIP users. For example if the SIP-DOMAIN
is
"sipserver.com":
[12:30:48] ABILIS_CPX:s user:test sip-domain:sipserver.com
COMMAND EXECUTED [12:30:51] ABILIS_CPX:s user:test2 sip-domain:sipserver.com
COMMAND EXECUTED
Note | |
---|---|
If the |
Note | |
---|---|
Remember to save the configuration (save conf). |
Because SIP users can be registered LOCAL (a remote peer is registered to Abilis) well as REMOTE (Abilis is registered to a remote peer), we will analyze both cases separately.
[11:04:10] ABILIS_CPX:d ctisip registry
User Host Prot Port REG LIFETIME AGE
-------------------------------------------------------------------------------
voipclient 080.080.080.080 UDP 5060 REMOTE 240 9
sip_peer 192.168.020.104 UDP 46022 LOCAL 120 112
sip_phone 192.168.020.107 UDP 56125 LOCAL 120 112
Tip | |
---|---|
Interesting chapter: Section 50.1.2, “CTISIP Registration”. |
Verify the configuration of SIP user. The
SIP-HOST
and/or
SIP-UDP-REMPORT
(SIP-TCP-REMPORT
) must be dynamic.
[11:50:48] ABILIS_CPX:d user:test | dynamic
SIP-HOST: DYNAMIC
SIP-TCP-REMPORT: (DYNAMIC)
SIP-UDP-REMPORT: (DYNAMIC)
Another parameter to be checked is
SIP-IP-PERMIT
.
[14:43:03] ABILIS_CPX:d user:test | sip-ip
SIP-IP-PERMIT: *
If the value is "*
", then any IP is
allowed. The problem can be when is configured a specific IP and
the registration is trying from another.
Verify the CTISIP log
[11:53:53] ABILIS_CPX:s ctisip log event:full
Debug log mask was set to:ERR REG INFO [11:54:15] ABILIS_CPX:d ctisip log
CURRENT EVENTS LOG MASK: ERR REG INFO 20/07 11:54:32 [ 10] Opened new SIP REG channel 20/07 11:54:32 [ 10] Declared user: test 20/07 11:54:32 [ 10] Password is not valid, registration failed for user: test
In this example the password entered is incorrect. Check your password and try again.
Verify the CTISIP statistics
DENIED-IP
[11:58:44] ABILIS_CPX:d s ctisip
RES:CtiSip --------------------------------------------------------------------
Session_Initiation_Protocol
--- Cleared 0 days 01:56:34 ago, on 20/07/2016 at 10:02:12 -------------
-----------|---INPUT---|--OUTPUT---|-----------|---INPUT---|--OUTPUT---|
SUCC-CALL | 1| 2|FAIL-CALL | 0| 0|
SUCC-REG | 121| 14|FAIL-REG | 2| 0|
SUCC-SUB | 0| |FAIL-SUB | 5| |
SUCC-NOT | | 0|FAIL-NOT | | 0|
SUCC-TRAN | 0| 0|FAIL-TRAN | 0| 0|
NOCHAN-CALL| 0| 0|NOCHAN-REG | 0| 0|
NOCHAN-SUB | 0| |NOCHAN-NOT | | 0|
EXP-RETRY | 0| 0|DTMF | 0| 0|
HOLD | 0| 0|UN-HOLD | 0| 0|
UNKNOWN | 0| 4|REINVITES | 0| 0|
LOST-NOBUF | 0| 0|OUTSEQ | 0| |
DENIED-IP | 20| |BANNED-IP | 0| |
------------------------------------------------------------------------
If
value of the parameter DENIED-IP
increases,
then it means that registration requests arrive at ABILIS, but
they are blocked by ABILIS, because IP addresses from that
tries to register aren't allowed. In this case verify the
configuration of parameters IPSRC
and
IPSRCLIST
of the resource CTISIP.
[12:19:41] ABILIS_CPX:d p ctisip
RES:CtiSip --------------------------------------------------------------------
Run DESCR:Session_Initiation_Protocol
LOG:ALL 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:092.115.190.246 DISPLAY-NAME:ADDRBOOK
IPSRC:127.000.000.001 IPSRCLIST:PrivateIpAdd
SUB-LIFETIME:180 max-sub:100 CTIP-TYPE:NET-PUBLIC
AUTH:DIGEST KEEPALIVE:90 NPOO-CT:SYS
LIFETIME:120 DISC-AUDIO:NO ROUTING:EN-BLOC
REM-AUTH:DIGEST,PLAIN T1:500 DIALT:5
REM-LIFETIME:240 T2:4 T302:15
AUTH-TOUT:4 T4:5 ROUTE-BY-SD:YES
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:ENABLE
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:abilis.voip.net
UA:AUTO (Abilis CPX - Ver. 8.3.7/STD - Build 4031.65 - Branch 8.3)
wdir:C:\APP\SIP\
The default configured list of source IP address
acceptance (IPSRCLIST
) is the list of
private IPs.
[12:58:32] ABILIS_CPX:d list:PrivateIpAdd
LIST:PrivateIpAdd - IR - Ref-Numb:6 Items-Numb:4
Automatically_generated_Private_Networks_list_(ReadOnly)
--------------------------------------------------------------------------
010.000.000.000:010.255.255.255 127.000.000.000:127.255.255.255
172.016.000.000:172.031.255.255 192.168.000.000:192.168.255.255
This situation happens often when the SIP phone is
located outside the local network. This problem is solved by
modifying the parameters mentioned above
(IPSRC
and
IPSRCLIST
).
If the IP from which trying to register the SIP phone is
static, then you can create another list of IP that will be
needed later to set at the parameter
IPSRCLIST
. But if the IP is dynamic will be
necessary to set the parameter IPSRC
with
value "*
". The second solution is valid for
both cases, but it has a greater security risk.
[11:22:02] ABILIS_CPX:s p ctisip ipsrc:*
COMMAND EXECUTED
Warning | |
---|---|
To activate the changes made, execute the initialization command init ctisys. Remember to save the configuration (save conf). |
BANNED-IP
[13:14:15] ABILIS_CPX:d s ctisip
RES:CtiSip --------------------------------------------------------------------
Session_Initiation_Protocol
--- Cleared 1 days 03:12:48 ago, on 20/07/2016 at 10:02:14 -------------
-----------|---INPUT---|--OUTPUT---|-----------|---INPUT---|--OUTPUT---|
SUCC-CALL | 1| 2|FAIL-CALL | 0| 0|
SUCC-REG | 835| 21|FAIL-REG | 5| 0|
SUCC-SUB | 0| |FAIL-SUB | 73| |
SUCC-NOT | | 0|FAIL-NOT | | 0|
SUCC-TRAN | 0| 0|FAIL-TRAN | 0| 0|
NOCHAN-CALL| 0| 0|NOCHAN-REG | 0| 0|
NOCHAN-SUB | 0| |NOCHAN-NOT | | 0|
EXP-RETRY | 0| 0|DTMF | 0| 0|
HOLD | 0| 0|UN-HOLD | 0| 0|
UNKNOWN | 0| 63|REINVITES | 0| 0|
LOST-NOBUF | 0| 0|OUTSEQ | 0| |
DENIED-IP | 0| |BANNED-IP | 38| |
------------------------------------------------------------------------
If value of the parameter BANNED-IP
increases, then it means that registration requests arrive at
ABILIS, but they are blocked by IPBAN of ABILIS, because it was
entered wrong password several times.
Type the following command to view the banned IP of resource CTISIP.
[13:15:02] ABILIS_CPX:d ipban banned res:ctisip
Banned IP addresses:1
RES | IP | Banned Time (mm:ss) | Remaining Time (mm:ss)
---------+-----------------+---------------------+-------------------------
CtiSip 192.168.020.107 60:0 58:26
If among banned IP is present your IP, then delete it from the list.
[13:28:49] ABILIS_CPX:c ipban banned res:ctisip ip:192.168.20.107
COMMAND EXECUTED
Change the correct password and try again.
Check if there is any access list (IPACL) on ABILIS that can block ports used by SIP resource. Type the following command to view the configured IPACL:
[13:34:00] ABILIS_CPX:d ipacl
IPRTR resource parameters: ACL:YES ACLBYPASS:#
COS:ENABLED COSDFT:NORMAL
Tot-IPACL-Number:6
-------------------------------------------------------------------------------
PR: [DESCR:]
TYPE: SA: PROT: ICMP-TYPE:
IPCOS: DA: SPO:/PO: DPO:
TOS-O: TOS-I: SRES: DRES:
TI:
-------------------------------------------------------------------------------
0 DENY * udp
DFT * 5060
-------------------------------------------------------------------------------
1 PERMIT * udp
HIGH * *
* *-D INT *
-------------------------------------------------------------------------------
2 PERMIT 192.168.001.070 udp
HIGH * * *
* * Ip-2 Ip-5
-------------------------------------------------------------------------------
In this example, the
pr
:0
, block the UDP port
5060, that is used by CTISIP resource. Remove this rule with
following command:
[13:41:55] ABILIS_CPX:c ipacl pr:0
COMMAND EXECUTED
Warning | |
---|---|
Remember to save the configuration (save conf). |
Verify if SIP packets arrive at ABILIS using the IPFLOW tracer
Add a IPFLOW filter to verify the IP packets with source/destination port 5060.
[13:56:33] ABILIS_CPX:ipflow filter add id:1 po:5060
COMMAND EXECUTED
Remember to save the configuration (save conf).
Activate trace.
[14:04:09] ABILIS_CPX:ipflow act
COMMAND EXECUTED
Once IPFLOW tracer is activated, it must be started to make packet trace.
[14:07:12] ABILIS_CPX:ipflow start
COMMAND EXECUTED
To display what IPFLOW has traced, the ipflow display command is used.
[14:08:10] ABILIS_CPX:ipflow display
--------------------------------------------------------------------------------
REC: 1 21/07/2016 - 14:07:18
Packet IN:
SA:192.168.020.107 DA:192.168.029.254 PROT:udp SPO:64703 DPO:5060 LEN:32
SIP:8 SRC-MAC:00-00-00-00-00-00 DST-MAC[?]:00-00-00-00-00-00 VLAN-PRIO:0
NAT:
IncomingSide:INSIDE
DstRC:NONE
SrcRC:NONE
Internal destination:
Packet forwarded to internal udp upper layer
Main return Code: GOOD (UDP)
--------------------------------------------------------------------------------
REC: 2 21/07/2016 - 14:07:18
Packet IN:
SA:212.000.211.152 DA:192.168.010.254 PROT:udp SPO:5060 DPO:5060 LEN:33
SIP:2 SRC-MAC:6C-19-8F-F9-73-D0 DST-MAC[U]:00-E0-4C-20-07-17 VLAN-PRIO:0
IPSEC decode:
RetCode:BYPASSED Tunnel:NO NatSide:NP IntDst:YES
NAT:
IncomingSide:OUTSIDE
DstRC:NONE
SrcRC:NONE
Internal destination:
Packet forwarded to internal udp upper layer
Main return Code: GOOD (UDP)
--------------------------------------------------------------------------------
In this example the packets arrive at ABILIS.
If packets do not arrive, we will see the following error:
[14:06:36] ABILIS_CPX:ipflow display
IPFLOW TRACER EMPTY
Using the IPFLOW traces can also see if the packets are blocked by IPACL. If a packet is blocked by IPACL, we have the following output:
[14:22:08] ABILIS_CPX:ipflow display
--------------------------------------------------------------------------------
REC: 1 21/07/2016 - 14:22:11
Packet IN:
SA:192.168.020.107 DA:192.168.029.254 PROT:udp SPO:64703 DPO:5060 LEN:32
SIP:8 SRC-MAC:00-00-00-00-00-00 DST-MAC[?]:00-00-00-00-00-00 VLAN-PRIO:0
IPACL check:
DIP:1
IPACL PR:0 IPCOS:DEFAULT
Main return Code: DATAGRAM DISCARDED DUE TO IPACL
In this example the packet is blocked by IPACL
pr
:0
. Please refer to the
previous
point.
Verify the SIP REGISTER messages using the Trace functionalities
Please refer to: Section 71.29.1, “How to trace only the SIP signaling traffic”.
Verify the configuration of SIP user. Pay attention to
SIP-REM-REG
parameter must be enabled and the
SIP-REM-USER
,
SIP-REM-AUTH-USER
and
SIP-REM-PASS
parameters must be
configured.
[15:13:19] ABILIS_CPX:d user:voipclient | rem-
SIP-REM-USER: voipclient
SIP-REM-PASS: ********
SIP-REM-AUTH: SYS
SIP-REM-AUTH-USER: AUTO (voipclient)
SIP-REM-REG: YES
Verify the CTISIP log
[08:52:32] ABILIS_CPX:s ctisip log event:full
Debug log mask was set to:ERR REG INFO [08:53:32] ABILIS_CPX:d ctisip log
CURRENT EVENTS LOG MASK: ERR REG INFO 22/07 08:52:24 [ 10] Opened new SIP REG channel with 192.168.20.254:5060 22/07 08:52:24 [ 10] Registering with peer test3 22/07 08:52:28 [ 10] Reg Info in OUT channel with 192.168.20.254:5060 22/07 08:52:28 [ 10] AUTH timer timeout 22/07 08:52:28 [ 10] Freeing the channel 22/07 08:52:28 [ 10] Info in IN channel with 192.168.20.254:5060 22/07 08:52:28 [ 10] Info in IN channel with 192.168.20.254:5060 22/07 08:52:28 [ 10] Freeing the channel 22/07 08:52:52 [ 10] Opened new SIP REG channel with 192.168.20.254:5060 22/07 08:52:52 [ 10] Registering with peer test3 22/07 08:52:56 [ 10] Error in OUT channel with 192.168.20.254:5060 22/07 08:52:56 [ 10] Error in OUT channel with 192.168.20.254:5060 22/07 08:52:56 [ 10] Freeing the channel
In this example, the possible issue is the incorrect
SIP-REM-USER
or
SIP-REM-AUTH-USER
or
SIP-REM-PASS
SIP-REM-USER
.
Verify these parameters and try again.
Verify the CTISIP statistics
[11:58:44] ABILIS_CPX:d s ctisip
RES:CtiSip --------------------------------------------------------------------
Session_Initiation_Protocol
--- Cleared 0 days 01:56:34 ago, on 20/07/2016 at 10:02:12 -------------
-----------|---INPUT---|--OUTPUT---|-----------|---INPUT---|--OUTPUT---|
SUCC-CALL | 1| 2|FAIL-CALL | 0| 0|
SUCC-REG | 121| 14|FAIL-REG | 2| 0|
SUCC-SUB | 0| |FAIL-SUB | 5| |
SUCC-NOT | | 0|FAIL-NOT | | 0|
SUCC-TRAN | 0| 0|FAIL-TRAN | 0| 0|
NOCHAN-CALL| 0| 0|NOCHAN-REG | 0| 0|
NOCHAN-SUB | 0| |NOCHAN-NOT | | 0|
EXP-RETRY | 0| 0|DTMF | 0| 0|
HOLD | 0| 0|UN-HOLD | 0| 0|
UNKNOWN | 0| 4|REINVITES | 0| 0|
LOST-NOBUF | 0| 0|OUTSEQ | 0| |
DENIED-IP | 20| |BANNED-IP | 0| |
------------------------------------------------------------------------
If
value of the parameter DENIED-IP
increases,
then it means that registration answers arrive at ABILIS, but they
are blocked by ABILIS. In this case verify the configuration of
parameters IPSRC
and
IPSRCLIST
of the resource CTISIP.
[12:19:41] ABILIS_CPX:d p ctisip
RES:CtiSip --------------------------------------------------------------------
Run DESCR:Session_Initiation_Protocol
LOG:ALL 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:092.115.190.246 DISPLAY-NAME:ADDRBOOK
IPSRC:127.000.000.001 IPSRCLIST:PrivateIpAdd
SUB-LIFETIME:180 max-sub:100 CTIP-TYPE:NET-PUBLIC
AUTH:DIGEST KEEPALIVE:90 NPOO-CT:SYS
LIFETIME:120 DISC-AUDIO:NO ROUTING:EN-BLOC
REM-AUTH:DIGEST,PLAIN T1:500 DIALT:5
REM-LIFETIME:240 T2:4 T302:15
AUTH-TOUT:4 T4:5 ROUTE-BY-SD:YES
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:ENABLE
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:abilis.voip.net
UA:AUTO (Abilis CPX - Ver. 8.3.7/STD - Build 4031.65 - Branch 8.3)
wdir:C:\APP\SIP\
The default configured list of source IP address acceptance
(IPSRCLIST
) is the list of private IPs.
[12:58:32] ABILIS_CPX:d list:PrivateIpAdd
LIST:PrivateIpAdd - IR - Ref-Numb:6 Items-Numb:4
Automatically_generated_Private_Networks_list_(ReadOnly)
--------------------------------------------------------------------------
010.000.000.000:010.255.255.255 127.000.000.000:127.255.255.255
172.016.000.000:172.031.255.255 192.168.000.000:192.168.255.255
This situation happens often when the SIP server is located
outside the local network. This problem is solved by modifying the
parameters mentioned above (IPSRC
and
IPSRCLIST
).
If the IP of the SIP server (the parameter
SIP-HOST
) is static, then you can create
another list of IP, where
insert the IP of SIP server and set the new list at the parameter
IPSRCLIST
. But if the IP is dynamic will be
necessary to set the parameter IPSRC
with value
"*
". The second solution is valid for both
cases, but it has a greater security risk.
[11:22:02] ABILIS_CPX:s p ctisip ipsrc:*
COMMAND EXECUTED
Warning | |
---|---|
To activate the changes made, execute the initialization command init ctisys. Remember to save the configuration (save conf). |
Check if there is any access list (IPACL) on ABILIS that can block ports used by SIP resource. Type the following command to view the configured IPACL:
[13:34:00] ABILIS_CPX:d ipacl
IPRTR resource parameters: ACL:YES ACLBYPASS:#
COS:ENABLED COSDFT:NORMAL
Tot-IPACL-Number:6
-------------------------------------------------------------------------------
PR: [DESCR:]
TYPE: SA: PROT: ICMP-TYPE:
IPCOS: DA: SPO:/PO: DPO:
TOS-O: TOS-I: SRES: DRES:
TI:
-------------------------------------------------------------------------------
0 DENY * udp
DFT * 5060
-------------------------------------------------------------------------------
1 PERMIT * udp
HIGH * *
* *-D INT *
-------------------------------------------------------------------------------
2 PERMIT 192.168.001.070 udp
HIGH * * *
* * Ip-2 Ip-5
-------------------------------------------------------------------------------
In this example, the
pr
:0
, block the UDP port
5060, that is used by CTISIP resource. Remove this rule with
following command:
[13:41:55] ABILIS_CPX:c ipacl pr:0
COMMAND EXECUTED
Warning | |
---|---|
Remember to save the configuration (save conf). |
Verify if SIP packets leave the Abilis and arrive at ABILIS using the IPFLOW tracer
Add a IPFLOW filter to verify the IP packets with source/destination port 5060.
[13:56:33] ABILIS_CPX:ipflow filter add id:1 po:5060
COMMAND EXECUTED
Remember to save the configuration (save conf).
Activate trace.
[14:04:09] ABILIS_CPX:ipflow act
COMMAND EXECUTED
Once IPFLOW tracer is activated, it must be started to make packet trace.
[14:07:12] ABILIS_CPX:ipflow start
COMMAND EXECUTED
To display what IPFLOW has traced, the ipflow display command is used.
[09:43:28] ABILIS_CPX:ipflow display
--------------------------------------------------------------------------------
REC: 1 22/07/2016 - 09:43:28
Packet IN:
SA:192.168.029.254 DA:192.168.020.254 PROT:udp SPO:5060 DPO:5060 LEN:468
SIP:INTERNAL
NAT:
IncomingSide:INSIDE
DstRC:NONE
SrcRC:NONE
External destination:
DIP:8 DST-GW:DIRECT
Packet OUT:
DIP:8 SRC-MAC:00-00-00-00-00-00 DST-MAC:00-00-00-00-00-00 VLAN-PRIO:0
DST-GW:192.168.020.254 IPCOS-PRIO:NORMAL
Main return Code: GOOD
--------------------------------------------------------------------------------
REC: 2 22/07/2016 - 09:43:28
Packet IN:
SA:192.168.020.254 DA:192.168.029.254 PROT:udp SPO:5060 DPO:5060 LEN:493
SIP:8 SRC-MAC:00-00-00-00-00-00 DST-MAC[?]:00-00-00-00-00-00 VLAN-PRIO:0
NAT:
IncomingSide:INSIDE
DstRC:NONE
SrcRC:NONE
Internal destination:
Packet forwarded to internal udp upper layer
Main return Code: GOOD (UDP)
--------------------------------------------------------------------------------
In this example the packets arrive at ABILIS.
If neither one packet is not recorded, we will see the following error:
[14:06:36] ABILIS_CPX:ipflow display
IPFLOW TRACER EMPTY
Using the IPFLOW traces can also see if the packets are blocked by IPACL. If a packet is blocked by IPACL, we have the following output:
[09:47:40] ABILIS_CPX:ipflow display
--------------------------------------------------------------------------------
REC: 1 22/07/2016 - 09:47:40
Packet IN:
SA:192.168.029.254 DA:192.168.020.254 PROT:udp SPO:5060 DPO:5060 LEN:469
SIP:INTERNAL
IPACL check:
DIP:8
IPACL PR:0 IPCOS:DEFAULT
Main return Code: DATAGRAM DISCARDED DUE TO IPACL
--------------------------------------------------------------------------------
In this example the packet is blocked by IPACL
pr
:0
. Please refer to the
previous
point.
Verify the SIP REGISTER messages using the Trace functionalities
Please refer to: Section 71.29.1, “How to trace only the SIP signaling traffic”.