This section contains instructions for a correct set-up of Abilis CPX and SIP proxy interconnection.
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 resoures and more
IP addresses the suggested configuration is
|
DOMAIN
: if Abilis has clients in the public
side you can also specify a FQDN.
In this case the proxy is a “normal” user like a soft
phone but the SIP-TYPE
is
LOCAL-PEER
.
In the figure there are the following elements:
Abilis is an user of SIP proxy domain “voip.it”;
“voip.it” is 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 proxy provides advanced services like IVR and voice mail, let's say that 10 sessions are needed.
Now you have to create a SIP user representing the user that is client of voip.it SIP proxy:
[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: USER: Provenience:
------------------------------------------------------------
* voipclient SIP-NUMBER
Abilis and the Sip Proxy interconnection is now correctly configured.
Note | |
---|---|
REMOTE-PEER handling is similar to SERVER but it allows calling/called number to pass unchanged. |
Some routings in the CTIR table must be added in order to route the calls to and from the CTISIP resource.
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 is
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 type:vtos poi:* cdi:*
COMMAND EXECUTED [18:05:18] ABILIS_CPX:a ctir pr:1 type:ctos sr:* cdi:*
COMMAND EXECUTED [18:05:30] ABILIS_CPX:a ctir pr:2 type:stoc ds:test cdi:* next:limited
COMMAND EXECUTED [18:05:38] ABILIS_CPX:a ctir pr:3 type:stov poo:g1 cdi:*
COMMAND EXECUTED [18:05:51] ABILIS_CPX:d ctir
- Not Saved (SAVE CONF), Not Refreshed (INIT) --------------------------------- Last change: 28/05/2015 10:36:34 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 VtoS * CtiSip * * ------------------------------------------------------------------------------- 1 CtoS * CtiSip * * ------------------------------------------------------------------------------- 2 StoC CtiSip test * * LIMITED ANY NO Dft * * ------------------------------------------------------------------------------- 3 StoV CtiSip G1 * * -------------------------------------------------------------------------------
Tip | |
---|---|
To allow G.729A you have to set
|
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 is 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 StoC
and CtoS routings will negotiate for the “C” side any
coder with maximum speed up 6400 bps.
[18:12:28] ABILIS_CPX:a ctir pr:0 type:vtos poi:* cdi:* sp:64000
COMMAND EXECUTED [18:12:37] ABILIS_CPX:a ctir pr:1 type:ctos sr:* cdi:* spout:64000 codersout:G.711
COMMAND EXECUTED [18:12:45] ABILIS_CPX:a ctir pr:2 type:stoc ds:test cdi:* next:limited sp:64000 coders:g.711 spout:6400 codersout:*,sys
COMMAND EXECUTED [18:12:53] ABILIS_CPX:a ctir pr:3 type:stov poo:g1 cdi:* sp:64000
COMMAND EXECUTED [18:13:00] ABILIS_CPX:d ctir
- Not Saved (SAVE CONF), Not Refreshed (INIT) --------------------------------- Last change: 28/05/2015 10:38:02 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 VtoS * CtiSip * * NO ANY NO Dft * * 64000 Sys Sys Sys Sys Sys * * ------------------------------------------------------------------------------- 1 CtoS * CtiSip * * NO ANY NO Dft * * 6400 Sys * * Sys Sys * * 64000 * * * NO Sys * * * * Sys AUTO AUTO Sys Sys NO * G.711 ------------------------------------------------------------------------------- 2 StoC CtiSip Clus1 * * LIMITED ANY NO Dft * * 64000 Sys Sys Sys Sys Sys * * 6400 * * * NO Sys * * * * Sys AUTO AUTO Sys Sys NO G.711 *,Sys ------------------------------------------------------------------------------- 3 StoV CtiSip G1 * * NO ANY NO Dft * * 64000 Sys Sys Sys Sys Sys * * -------------------------------------------------------------------------------
Tip | |
---|---|
To allow G.729A you have to set
|
When the Abilis-SIP proxy interconnection occurs via local LAN, (i.e. with high speed, minimal delays, minimal jitter), optimising 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 | |
---|---|
|
When transcoding takes place in StoC and CtoS, with G.711 toward the SIP proxy, something interesting happens: on the WAN FAX relay can be used! Abilis can exchange FAX with following characteristics:
use 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