| 52.4. Connecting Abilis and a SIP phone | ||
|---|---|---|
 | Chapter 52. SIP - Session Initiation Protocol |  |
| 52.4. Connecting Abilis and a SIP phone | ||
|---|---|---|
| | Chapter 52. SIP - Session Initiation Protocol | |
This section contains instructions for a correct set-up of Abilis CPX and SIP phone interconnection.
In the figure there are the following elements:
Site 1
Site 2
SIP phone;
Abilis that manages all the systems and a VPN with “Site 1”.
The purpose is to enable the communication between:
SIP users in “Site 1” (SIP phone and PC with Zoiper);
The POTS phone and the SIP users of “Site 1”;
SIP users in “Site 1” and the SIP users in “Site 2”.
Assumptions: the POTS cards, used to manage the analog phones and configuration for Cluster and CtiLink for compressed phone calls, are already active on Abilis.
Activate the CTISIP resource in the Abilis of “Site 1” and enable up to 10 SIP connections.
[10:27:42] ABILIS_CPX_1:s p ctisip act:yes sesnum:10COMMAND EXECUTED [10:27:53] ABILIS_CPX_1:d p ctisipRES:CtiSip - Not Saved (SAVE CONF), Not Refreshed (INIT) ---------------------- Run DESCR:Session_Initiation_Protocol LOG:NO 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:OUT-IP DISPLAY-NAME:NO IPSRC:127.000.000.001 IPSRCLIST:PrivateIpAdd SUB-LIFETIME:180 max-sub:100 CTIP-TYPE:USER AUTH:DIGEST KEEPALIVE:90 NPOO-CT:SYS LIFETIME:120 DISC-AUDIO:NO ROUTING:EN-BLOC REM-AUTH:DIGEST T1:500 DIALT:5 REM-LIFETIME:120 T2:4 T302:15 AUTH-TOUT:4 T4:5 ROUTE-BY-SD:NO 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:DISABLE 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: UA:AUTO (Abilis CPX - Ver. 8.3.0/STD - Build 4031.2 - Branch 8.3) wdir:C:\APP\SIP\
Keep all the default parameters.
Repeat the same operation for the Abilis of “Site 2”.
Activate two SIP user in the Abilis of “Site 1” (PC with Zoiper and SIP phone).
Add the user in the Abilis of “Site 1” and enable it to SIP.
[10:50:19] ABILIS_CPX_1:a user:zoiper sip:yesCOMMAND EXECUTED [10:50:28] ABILIS_CPX_1:d user- Not Saved (SAVE CONF) ------------------------------------------------------- ------------------------+-------------+---------------------------------------- USER PWD ACT|CTIP CLUS |CHAT LDAP PPP FTP HTTP MAIL IAX SIP VO ------------------------+-------------+---------------------------------------- admin *** YES # # YES YES YES NO YES NO NO NO NO guest NO # # NO NO NO NO NO NO NO NO NO zoiper YES # # NO NO NO NO NO NO NO YES NO
Give the new user the following characteristics:
Password: zoiper
Host: dynamic (the PC receives the IP address from a DHCP server)
Simultaneous half-duplex call: 2
SIP number: 610
[09:45:52] ABILIS_CPX_1:s user:zoiper pwd:zoiper sip-host:dynamic sip-maxses-bid:2 sip-number:610COMMAND EXECUTED [09:46:23] ABILIS_CPX_1:d user:zoiper- Not Saved (SAVE CONF) ------------------------------------------------------- Parameter: | Value: ------------------+------------------------------------------------------------ USER: zoiper ALIAS: zoiper ID: 9 <Read Only> PWD: *** ACT: YES GROUP: CTIP: # CLUS: # OPC-ROLE: USER OPC-VIEW: * OPC-MONITOR: NONE SIP: YES SIP-TYPE: PHONE SIP-DOMAIN: SYS SIP-HOST: DYNAMIC SIP-TCP-PORT: (DYNAMIC) SIP-UDP-PORT: (DYNAMIC) SIP-SRCADD: SYS SIP-PROT-IN: TCP,UDP SIP-PROT-OUT: UDP SIP-IP-PERMIT: * SIP-MAXSES-BID: 2 SIP-MAXSES-IN: 0 SIP-MAXSES-OUT: 0 SIP-BUSY-INUSE: NO SIP-NUMBER: 610 SIP-CG-NUM: AUTO SIP-FWD-CG-NUM: CALLER SIP-DISPLAY-NAME: SYS 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-SS: NO SIP-SS-PICKUP: NO SIP-SS-PRES-CG: YES SIP-SS-CF-DND: 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-AUTH-USER: AUTO () SIP-REM-REG: NO -------------------------------------------------------------------------------
Add the user in the Abilis of “Site 1” and enable it to SIP.
[09:51:19] ABILIS_CPX_1:a user:sip_phone sip:yesCOMMAND EXECUTED [09:51:31] ABILIS_CPX_1:d user- Not Saved (SAVE CONF) ------------------------------------------------------- ------------------------+-------------+---------------------------------------- USER PWD ACT|CTIP CLUS |CHAT LDAP PPP FTP HTTP MAIL IAX SIP VO ------------------------+-------------+---------------------------------------- admin *** YES # # YES YES YES NO YES NO NO NO NO guest NO # # NO NO NO NO NO NO NO NO NO sip_phone YES # # NO NO NO NO NO NO NO YES NO zoiper *** YES # # NO NO NO NO NO NO NO YES NO
Give the new user the following characteristics:
Password:sip_phone
Host: dynamic (the PC receives the IP address from a DHCP server)
2 simultaneous half-duplex calls
SIP number: 630
[09:51:51] ABILIS_CPX_1:s user:sip_phone pwd:sip_phone sip-host:dynamic sip-maxses-bid:2 sip-number:630COMMAND EXECUTED [09:52:38] ABILIS_CPX_1:d user:sip_phone- Not Saved (SAVE CONF) ------------------------------------------------------- Parameter: | Value: --------------------+---------------------------------------------------------- USER: sip_phone REAL-NAME: sip_phone ID: 7 <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: PHONE SIP-DOMAIN: SYS SIP-HOST: DYNAMIC SIP-TCP-REMPORT: (DYNAMIC) SIP-UDP-REMPORT: (DYNAMIC) 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-BUSY-INUSE: NO SIP-NUMBER: 630 SIP-ADDRBOOK-NUM: SIP-NUMBER SIP-CG-NUM: AUTO SIP-FWD-CG-NUM: CALLER SIP-DISPLAY-NAME: SYS 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-AUTH-USER: AUTO () SIP-REM-REG: NO -------------------------------------------------------------------------------
Add another “sip_phone” user in the Abilis of
“Site 2” and assign it the
SIP-NUMBER:910 (the
configuration is similar to that one of “Site 1”).
Since the SIP-NUMBER parameter for the SIP
users of “Site 1” is specified, the routings will be
automatically connected in the table.
![[Tip]](../images/tip.png)
[11:25:38] ABILIS_CPX_1:d ctisip numbers
Total:2 Sip-Number:2 Static:0
NUMx: [SIP-NUMBER:] USER: Provenience:
------------------------------------------------------------------------
[610] zoiper SIP-NUMBER
[630] sip_phone SIP-NUMBER
Further modifications aren't necessary.
Since the SIP-NUMBER parameter for the SIP
user of “Site 2” is specified, the routing will be
automatically connected in the table.
[12:03:55] ABILIS_CPX_2:d ctisip numbers
Total:1 Sip-Number:1 Static:0
NUMx: [SIP-NUMBER:] USER: Provenience:
------------------------------------------------------------------------
[910] sip_phone SIP-NUMBER
Further modifications aren't necessary.
The purpose is to enable the communication between:
SIP users in “Site 1” (SIP phone and PC with Zoiper);
The POTS phone and the SIP users of “Site 1”;
SIP users in “Site 1” and the SIP users in “Site 2”.
To connect up the PC with Zoiper and the SIP phone in “Site 1”, type:
[10:27:19] ABILIS_CPX_1:a ctir pr:0 poi:sip out:sip cdi:'numsip' sp:64000COMMAND EXECUTED [16:42:17] ABILIS_CPX_1:d ctirLast 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 # # Sip 'NumSip' * NO ANY NO Dft * * 64000 Sys Sys Sys Sys Sys * * -------------------------------------------------------------------------------- [13:49:17] ABILIS_CPX_1:d list:numsipLIST:NumSip - IN Automatically_generated_CTI_SIP_Numbers_list_(ReadOnly) -------------------------------------------------------------------------- 610 630
We set up the SP parameter to
64000 to keep the available coder pool.
Add the following routings:
[17:22:52] ABILIS_CPX_1:a ctir pr:1 poi:sip out:101 cdi:01 sp:64000COMMAND EXECUTED [17:23:10] ABILIS_CPX_1:a ctir pr:2 poi:pbx out:sip cdi:'numsip' sp:64000COMMAND EXECUTED [16:42:17] ABILIS_CPX_1: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 # # Sip 'NumSip' * NO ANY NO Dft * * 64000 Sys Sys Sys Sys Sys * * -------------------------------------------------------------------------------- 1 VOICE Sip # # 101 01 * NO ANY NO Dft * * 64000 Sys Sys Sys Sys Sys * * -------------------------------------------------------------------------------- 2 VOICE PBX # # Sip 'NumSip' * NO ANY NO Dft * * 64000 Sys Sys Sys Sys Sys * * --------------------------------------------------------------------------------
The PR:1 routing routes
toward port 101 (where the analog phone is
connected) the calls coming from one of the SIP users and directed
to the number 01.
The PR:2 routing routes
the calls coming from the Abilis POTS port towards the CTISIP and
directed to one number of the NumSip list; the CTISIP translation table
will forward the call to the user.
Add routings with the following commands to connect up the PC with Zoiper in “Site 1” and the SIP phone in “Site 2”:
[10:27:19] ABILIS_CPX_1:a ctir pr:3 poi:sip out:site2 cdi:910 sp:64000COMMAND EXECUTED [10:29:50] ABILIS_CPX_1:a ctir pr:4 sr:* out:sip cdi:'numsip' sp:64000COMMAND EXECUTED [16:33:00] ABILIS_CPX_1: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 # # Sip 'NumSip' * NO ANY NO Dft * * 64000 Sys Sys Sys Sys Sys * * -------------------------------------------------------------------------------- 1 VOICE Sip # # 101 01 * NO ANY NO Dft * * 64000 Sys Sys Sys Sys Sys * * -------------------------------------------------------------------------------- 2 VOICE PBX # # Sip 'NumSip' * NO ANY NO Dft * * 64000 Sys Sys Sys Sys Sys * * -------------------------------------------------------------------------------- 3 VOICE Sip # # site2 910 * NO ANY NO Dft * * 64000 Sys Sys Sys Sys Sys * * -------------------------------------------------------------------------------- 4 VOICE # * # Sip 'NumSip' * NO ANY NO Dft * * 64000 Sys Sys Sys Sys Sys * * --------------------------------------------------------------------------------
We set up the SP parameter to
64000 to keep the available coder pool.
The PR:3 routing routes
towards the Site2 cluster the calls coming from
SIP users and directed to the number 910.
The PR:4 routing routes
towards the CTISIP the calls coming from any cluster and directed to
one of the numbers of the NumSip list; the CtiSip translation table
will forward the calls to the user.
Add the following command for enable routing using the subaddress called field:
[10:39:13] ABILIS_CPX_2:s p ctisip route-by-sd:yes
COMMAND EXECUTEDAdd the following command for change CTIR Routing:
[10:39:13] ABILIS_CPX_2:s ctir pr:0 sdo:zoiperCOMMAND EXECUTED [16:42:17] ABILIS_CPX_1:d ctirLast change: 29/07/2015 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 # # Sip 'NumSip' * NO ANY NO Dft * * 64000 Sys Sys Sys Sys Sys * ZOIPER --------------------------------------------------------------------------------
The match for CTIR Routing in this example is SDO (The SIP user "Zoiper") but not CDO (SIP number).
![[Note]](../images/note.png) | Note |
|---|---|
If SDO does not match any SIP user then it uses the matching with CDO. |
The example is similar to the configuration of the Abilis of the “Site 1”; type the following commands:
[10:39:13] ABILIS_CPX_2:a ctir pr:0 poi:sip out:site1 cdi:610 sp:64000COMMAND EXECUTED [10:40:10] ABILIS_CPX_2:a ctir pr:1 sr:* out:sip cdi:'numsip' sp:64000COMMAND EXECUTED [10:40:17] ABILIS_CPX_2: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 -------------------------------------------------------------------------------- 3 VOICE Sip # # site1 610 * NO ANY NO Dft * * 64000 Sys Sys Sys Sys Sys * * -------------------------------------------------------------------------------- 4 VOICE # * # Sip 'NumSip' * NO ANY NO Dft * * 64000 Sys Sys Sys Sys Sys * * -------------------------------------------------------------------------------- [13:49:17] ABILIS_CPX_2:d list:numsipLIST:NumSip - IN Automatically_generated_CTI_SIP_Numbers_list_(ReadOnly) -------------------------------------------------------------------------- 910
Like the previous chart, the SP parameter
is set at 64000 to keep the available coder
pool.
The PR:0 routing routes
toward the “Site1”cluster the calls coming from SIP
users and directed to the number 610.
The PR:1 routing routes
toward the CTISIP the calls coming from any cluster and directed to
one of the numbers of the NumSip list; the CTISIP
translation table will forward the calls to the user.
| |  | |
| 52.3. CTI Routings |  | 52.5. Connecting Abilis and a SIP server |