| 49.5. Connecting Abilis and a SIP server | ||
|---|---|---|
 | Chapter 49. SIP - Session Initiation Protocol |  |
| 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.
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]](../images/tip.png) | 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 server 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 server;
SIP server has static IP: 192.168.1.100;
Abilis has only one SIP user;
Abilis user is “test” with password “secret”;
| Tip |
|---|---|
Interesting chapter: Section 49.2.1.1, “CTISIP INVITE messages” |
Now you have to create a SIP user representing the user that is SIP PEER client of the Abilis
[14:49:07] ABILIS_CPX:a user:test pwd:secret sip:yesCOMMAND EXECUTED [14:49:07] ABILIS_CPX:s user:test sip-type:local-peer sip-number:* sip-host:192.168.1.100 sip-udp-remport:5060COMMAND EXECUTED
![[Note]](../images/note.png) | Note |
|---|---|
The SIP server is not required to be only on 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: 192.168.1.100
SIP-TCP-REMPORT: 5060
SIP-UDP-REMPORT: 5060
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
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.
| Tip |
|---|---|
interesting chapter: Section 49.2.1.1, “CTISIP INVITE messages” |
Now you have to create a SIP user representing the user that is client of voip.it SIP server:
[14:49:07] ABILIS_CPX:a user:voipclient pwd:swordfish sip:yesCOMMAND EXECUTED [14:49:07] ABILIS_CPX:s user:voipclient sip-type:server sip-domain:voip.itCOMMAND EXECUTED [14:49:07] ABILIS_CPX:s user:voipclient sip-host:88.88.88.88COMMAND 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:swordfishCOMMAND 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 Server interconnection is now correctly configured.
| Note |
|---|---|
|
In the figure there are the following elements:
Abilis is an user of SIP server witch 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”.
| Tip |
|---|---|
Interesting chapter: Section 49.2.1.1, “CTISIP INVITE messages” |
Now you have to create a SIP user representing the user that is client of voip.it SIP server
[14:49:07] ABILIS_CPX:a user:voipclient pwd:swordfish sip:yesCOMMAND EXECUTED [14:49:07] ABILIS_CPX:s user:voipclient sip-type:remote-peer sip-host:88.88.88.88COMMAND EXECUTED [14:49:07] ABILIS_CPX:s user:voipclient sip-number:* sip-rem-reg:yes sip-rem-user:voipclient sip-rem-pass:swordfishCOMMAND 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: REMOTE-PEER
SIP-DOMAIN: SYS
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-NUMBERAbilis and the Sip Server interconnection is now correctly configured.
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 poi:* out:sip cdi:*COMMAND EXECUTED [18:05:18] ABILIS_CPX:a ctir pr:1 sr:* out:sip cdi:*COMMAND EXECUTED [18:05:30] ABILIS_CPX:a ctir pr:2 poi:sip out:test cdi:* next:limitedCOMMAND EXECUTED [18:05:38] ABILIS_CPX:a ctir pr:3 poi:sip out:g1 cdi:*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 VOICE * # # Sip * * -------------------------------------------------------------------------------- 1 VOICE # * # Sip * * -------------------------------------------------------------------------------- 2 VOICE Sip # # test * * LIMITED ANY NO Dft * * -------------------------------------------------------------------------------- 3 VOICE Sip # # 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 poi:* out:sip cdi:* sp:64000COMMAND EXECUTED [18:12:37] ABILIS_CPX:a ctir pr:1 sr:* out:sip cdi:* spout:64000 codersout:G.711COMMAND 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:*,sysCOMMAND EXECUTED [18:12:53] ABILIS_CPX:a ctir pr:3 poi:sip out:g1 cdi:* sp:64000COMMAND 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 VOICE * # # Sip * * NO ANY NO Dft * * 64000 Sys Sys Sys Sys Sys * * -------------------------------------------------------------------------------- 1 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 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 VOICE Sip # # 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:80COMMAND EXECUTED [18:15:29] ABILIS_CPX:s ctir pr:2 dj:0 mj:80 djout:sys mjout:sysCOMMAND 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:noCOMMAND EXECUTED [18:18:35] ABILIS_CPX:s ctir pr:1 fmrelay:noCOMMAND EXECUTED [18:18:29] ABILIS_CPX:s ctir pr:2 fmrelay:noCOMMAND EXECUTED [18:15:29] ABILIS_CPX:s ctir pr:3 fmrelay:noCOMMAND EXECUTED
| |  | |
| 49.4. Connecting Abilis and a SIP phone |  | 49.6. Busy Lamp Field (BLF) |