TBwiki - User contributions [en]

Parameter: Proxy Port Type

2021-03-26T02:44:43Z

William Wong:

The Proxy Port Type parameter is used to identify the type of IP port used by Proxy. Values for this parameter are selected from a drop-down list. The Proxy Port Type parameter can take on the following values:

* UDP
* TCP

[[Category:Parameters]]

Parameter: Proxy Port Type

2021-03-26T02:42:09Z

William Wong:

The Proxy Port Type parameter is used to identify the type of IP port used by a SIP transport server. Values for this parameter are selected from a drop-down list. The Proxy Port Type parameter can take on the following values:

* UDP
* TCP

[[Category:Parameters]]

Toolpack Installation:Change hostname

2021-03-04T03:52:40Z

William Wong: /* Change hostname */

== Change hostname ==

This procedure applies to TMG800, TMG3200 and TMG7800 with static IP for management interfaces.
Require Version rel2.9 and above

*ssh to the Centos host
*Do shell command "hostname" to show the current hostname
<pre>
hostname
</pre>
For Centos 7, the hostname command is
<pre>
hostnamectl
</pre>

*Change to the new name:
<pre>
hostname NewHostName
</pre>
For CentOS 7,
<pre>
hostnamectl set-hostname NewHostName
</pre>

*Restart toolpack:
<pre>tbtoolpack stop
tbtoolpack start
</pre>
For TMG7800 CentOS 7, do reboot host, and after reboot host, if you notice
/etc/hosts
is still with old name, perform manual change of NewHostName for this file /etc/hosts, then reboot host again

Note: DO NOT change the hostname via webportal.

Toolpack Installation:Change hostname

2021-03-04T03:51:58Z

William Wong: /* Change hostname */

== Change hostname ==

This procedure applies to TMG800, TMG3200 and TMG7800 with static IP for management interfaces.
Require Version rel2.9 and above

*ssh to the Centos host
*Do shell command "hostname" to show the current hostname
<pre>
hostname
</pre>
*For Centos 7, the hostname command is
<pre>
hostnamectl
</pre>

*Change to the new name:
<pre>
hostname NewHostName
</pre>
For CentOS 7,
<pre>
hostnamectl set-hostname NewHostName
</pre>

*Restart toolpack:
<pre>tbtoolpack stop
tbtoolpack start
</pre>
For TMG7800 CentOS 7, do reboot host, and after reboot host, if you notice
/etc/hosts
is still with old name, perform manual change of NewHostName for this file /etc/hosts, then reboot host again

Note: DO NOT change the hostname via webportal.

Toolpack Installation:Change hostname

2021-02-26T08:08:55Z

William Wong: /* Change hostname */

== Change hostname ==

This procedure applies to TMG800, TMG3200 and TMG7800 with static IP for management interfaces.
Require Version rel2.9 and above

*ssh to the Centos host
*Do shell command "hostname" to show the current hostname
<pre>
hostname
</pre>
*For Centos 7, the hostname command is
<pre>
hostnamectl
</pre>

*Change to the new name:
<pre>
hostname NewHostName
</pre>
*For CentOS 7,
<pre>
hostnamectl set-hostname NewHostName
</pre>

*Restart toolpack:
<pre>tbtoolpack stop
tbtoolpack start
</pre>
For TMG7800 CentOS 7, do reboot host, and after reboot host, if you notice
/etc/hosts
is still with old name, perform manual change of NewHostName for this file /etc/hosts, then reboot host again

Note: DO NOT change the hostname via webportal.

Toolpack Installation:Change hostname

2021-02-26T08:07:52Z

William Wong: /* Change hostname */

== Change hostname ==

This procedure applies to TMG800, TMG3200 and TMG7800 with static IP for management interfaces.
Require Version rel2.9 and above

*ssh to the Centos host
*Do shell command "hostname" to show the current hostname
<pre>
hostname
</pre>
*For Centos 7, the hostname command is
</pre>
hostnamectl
</pre>

*Change to the new name:
<pre>
hostname NewHostName
</pre>
*For CentOS 7,
</pre>
hostnamectl set-hostname NewHostName
</pre>

*Restart toolpack:
<pre>tbtoolpack stop
tbtoolpack start
</pre>
For TMG7800 CentOS 7, do reboot host, and after reboot host, if you notice
/etc/hosts
is still with old name, perform manual change of NewHostName for this file /etc/hosts, then reboot host again

Note: DO NOT change the hostname via webportal.

Toolpack Installation:Change hostname

2021-02-26T08:06:17Z

William Wong: /* Change hostname */

== Change hostname ==

This procedure applies to TMG800, TMG3200 and TMG7800 with static IP for management interfaces.
Require Version rel2.9 and above

*ssh to the Centos host
*Do shell command "hostname" to show the current hostname
<pre>
hostname
</pre>
For Centos 7, the hostname command is
</pre>
hostnamectl
</pre>

*Change to the new name:
<pre>
hostname NewHostName
</pre>
For CentOS 7,
</pre>
hostnamectl set-hostname NewHostName
</pre>

*Restart toolpack:
<pre>tbtoolpack stop
tbtoolpack start
</pre>
For TMG7800 CentOS 7, do reboot host, and after reboot host, if you notice
/etc/hosts
is still with old name, perform manual change of NewHostName for this file /etc/hosts, then reboot host again

Note: DO NOT change the hostname via webportal.

Toolpack Installation:Change hostname

2021-02-26T08:04:32Z

William Wong: /* Change hostname */

== Change hostname ==

This procedure applies to TMG800, TMG3200 and TMG7800 with static IP for management interfaces.
Require Version rel2.9 and above

*ssh to the Centos host
*Do shell command "hostname" to show the current hostname
<pre>
hostname
</pre>
For Centos 7, the hostname command is,
</pre>
hostnamectl
</pre>

*Change to the new name:
<pre>
hostname NewHostName
</pre>
For CentOS 7,
</pre>
hostnamectl set-hostname NewHostName
</pre>

*Restart toolpack:
<pre>tbtoolpack stop
tbtoolpack start
</pre>
For TMG7800 CentOS 7, do reboot host, and after reboot host, if you notice
/etc/hosts
is still with old name, perform manual change of NewHostName for this file /etc/hosts, then reboot host again

Note: DO NOT change the hostname via webportal.

Toolpack Installation:Change hostname

2021-02-26T08:03:38Z

William Wong: /* Change hostname */

== Change hostname ==

This procedure applies to TMG800, TMG3200 and TMG7800 with static IP for management interfaces.
Require Version rel2.9 and above

*ssh to the Centos host
*Do shell command "hostname" to show the current hostname
<pre>
hostname
</pre>
For Centos 7, the hostname command is,
</pre>
hostnamectl
</pre>

*Change to the new name:
<pre>hostname NewHostName
</pre>
For CentOS 7,
hostnamectl set-hostname NewHostName

*Restart toolpack:
<pre>tbtoolpack stop
tbtoolpack start
</pre>
For TMG7800 CentOS 7, do reboot host, and after reboot host, if you notice
/etc/hosts
is still with old name, perform manual change of NewHostName for this file /etc/hosts, then reboot host again

Note: DO NOT change the hostname via webportal.

Toolpack:Creating a SIP Domain SBC A

2021-01-18T10:39:26Z

William Wong: /* Applies to version(s): v3.0 */

=== '''''Applies to version(s): v3.0''''' ===
{{DISPLAYTITLE:Creating a SIP Domain}}
A SIP domain represents a grouping of devices (or users) that can communicate with one another.
You must configure SIP Registration Domain for your system. The first step in doing so is to create a SIP Domain:

1- Click '''SIP Domain''' in the navigation panel

[[Image:SIP_Domain_NavigationMenu.png]]

2- Click '''Create New Domain'''

[[Image:Create_SIP_Domain.png]]

3- Create the new Domain:

* Enter a configuration '''Name''' for this domain.
* Enter a '''Domain Name''' for the SIP Registration Domain (domain can be a FQDN or an IP address)
* Set the number '''Maximum Registered Users''' for this domain
* Set the Expires value used by SBC when the remote device doesn't supply one ('''Default Contact Expire''')
* Select '''Routing Method''' the system will use to route calls to registered users (if enabled in routing scripts).
** '''Register source''': Sends SIP Invite to the registering source IP address.
** '''Contact''': Sends SIP Invite to the 'contact' from the Register message.
* Set the '''Default Contact Expiration''', this value will be used when no ''Expires'' value is supplied by the user agent.
* Set the '''Minimum Contact Expiration''', this is the minimum ''Expires'' value that can be supplied by a user agent. Lower values will be rejected with a 423 'Interval too brief' response.
* Set the '''Maximum Contact Expiration''', this is the maximum ''Expires'' value that can be supplied by a user agent. Higher value are replaced by this parameter.
* Forwarding Parameters:
** Select the Registration '''Forwarding Mode''' to the registrar:
*** '''Contact Remapping''': Changes the user and the IP address.
*** '''Contact Passthrough''': Doesn't change anything. Enables devices to be contacted directly without going through the SBC.
** Set '''Minimum Registrar Expiration''', this is the minimum ''Expires'' value sent by the SBC to the registrar. If a user agent 'Expires' value is less than this parameter, the SBC will do rate adaptation between the user agent and the Registrar.
** Set '''Maximum Pending Register Forward''', the maximum number of simultaneous pending register requests allowed for this domain. New REGISTER request are being refused passed this threshold.
** Set '''Maximum Simultaneous Register Forward''', the maximum number of simultaneous active register requests allowed for this domain. New REGISTER request are being refused passed this threshold.
** Select the Registration '''Registrar Selection Mode''' to the registrar:
*** '''Active/Standby''': The active registrar is the one with status up and the lowest priority (automatic fallback).
** Enable the '''Forward After Switchover''': After a switchover, the first register refresh request from each user will be forwarded to the active registrar.
* Click '''Create'''

[[Image:Create_SIP_Domain1.png]]

4- Verify that the '''Sip registration domain was successfully created''' message appears

[[Image:Create_SIP_Domain2.png]]
 
[[Image:Create_SIP_Domain3.png]]

5- Add the domain to one or multiple NAPs from which users are allowed to register for this domain
* Click '''NAPs''' in the navigation panel
* Click on a NAP to bind the SIP Domain to
* Add the SIP Domain to the NAP (bottom of the page):
[[Image:bind_domain_to_nap.png]]

6- Multiple registrars (registration redundancy)
Multiple registrars can be created per domain, for redundancy.
* Registrars are ordered by priority
* Users' registration requests are forwarded to the available registrar with the highest priority in the list
* SIP 'options' polling is used to determine which registrars are available or not

[[Image:multiple_registrars.png]]

== Create routes for registered users ==
In most cases, TBSC will be required to route SIP calls (SIP Invite) to registered users. That's done by [[Sip_registration_forwarding#SIP_Calls_routing|creating appropriate routes]].

VerifyIua B

2021-01-04T08:42:10Z

William Wong: /* Status menu */

__FORCETOC__
<div style="font-size: 132%;">'''''Applies to version(s): v2.9, v2.10'''''</div>

{{DISPLAYTITLE:IUA Status}}

This article illustrates how to view IUA status and to set a periodic refresh of the IUA protocol stack. This is done from the Status menu and the Navigation panel.

===Status menu===

1- Click '''Status''' in the navigation panel.

[[Image:Status_0.png]] 

2- Click the '''IUA''' tab.

[[Image:TabIUA_0.png]] 

* Note that SCTP is used as the Transport Layer for this protocol, if any firewall is sitting between Telcobridges device and Sigtran network, this firewall needs to be set to allow SCTP traffic to pass through, and enabling the basic connectivity for sigtran link to be up

The status of the IUA protocol stack is displayed.

[[Image:StatusIUA_0.png]]

===Navigation Panel===
1- Click '''IUA''' from the navigation panel.

[[Image:StatusIUA_1.png]] 

2- Click the '''Status''' tab.

[[Image:StatusIUA_2.png]] 

3- To configure a periodic refresh of the IUA status, select a value from '''Refresh Every'''.

[[Image:StatusIUA_3a.png]] 

4- Select an IUA SAP to view its status.

[[Image:StatusIUA_3.png]] 

The SAP status is displayed.

[[Image:StatusIUA_4.png]]

5-To view the cluster, select the '''Clusters''' tab.

[[Image:StatusIUA_5.png]] 

The Clusters link(s) are displayed.

[[Image:StatusIUA_6.png]] 

<div class="mw-collapsible mw-collapsed" data-collapsetext="Northbound Interface" data-expandtext="Northbound Interface" style="width: 400px;">

'''Path'''
<pre>
/configurations/@[configuration_name]/hardware_units/@[hardware_name]/iua_stacks/@[iua_stack]/status
</pre>
</div>

VerifyIua A

2021-01-04T08:41:42Z

William Wong: /* Status menu */

__FORCETOC__
<div style="font-size: 132%;">'''''Applies to version(s): v2.8.'''''</div>

{{DISPLAYTITLE:IUA Status}}

This article illustrates how to view IUA status and to set a periodic refresh of the IUA protocol stack. This is done from the Status menu and the Navigation panel.

===Status menu===

1- Click '''Status''' in the navigation panel.

[[Image:Status_0.png]] 

2- Click the '''IUA''' tab.

[[Image:TabIUA_0.png]] 

* Note that SCTP is used as the Transport Layer for this protocol, if any firewall is sitting between Telcobridges device and Sigtran network, this firewall needs to be set to allow SCTP traffic to pass through, and enabling the basic connectivity for sigtran link to be up

The status of the IUA protocol stack is displayed.

[[Image:StatusIUA_0.png]]

===Navigation Panel===
1- Click '''IUA''' from the navigation panel.

[[Image:StatusIUA_1.png]] 

2- Click the '''Status''' tab.

[[Image:StatusIUA_2.png]] 

3- To configure a periodic refresh of the IUA status, select a value from '''Refresh Every'''.

[[Image:StatusIUA_3a.png]] 

4- Select an IUA SAP to view its status.

[[Image:StatusIUA_3.png]] 

The SAP status is displayed.

[[Image:StatusIUA_4.png]]

5-To view the cluster, select the '''Clusters''' tab.

[[Image:StatusIUA_5.png]] 

The Clusters link(s) are displayed.

[[Image:StatusIUA_6.png]]

Toolpack:SIGTRAN M3uaASPStatus B

2021-01-04T08:39:44Z

William Wong: /* Status menu */

__FORCETOC__
<div style="font-size: 132%;">'''''Applies to version(s): v2.9, v2.10'''''</div>

{{DISPLAYTITLE:M3UA ASP Status}}

This article illustrates how to view M3UA ASP status. This can be done from either the Status menu or the Navigation panel.

===Status menu===

1- Click '''Status''' in the navigation panel.

[[Image:Status_0.png]] 

2- Click the '''SIGTRAN M3ua''' tab.

[[Image:TabM3UA_0.png]] 

* Note that SCTP is used as the Transport Layer for this protocol, if any firewall is sitting between Telcobridges device and Sigtran network, this firewall needs to be set to allow SCTP traffic to pass through, and enabling the basic connectivity for sigtran link to be up

The status of the M3UA protocol stack is displayed.

[[Image:StatusM3UA_SGP_0.png]]

===Navigation Panel===
1- Click '''M3UA''' from the navigation panel.

[[Image:StatusM3UA_1.png]] 

2- Click the '''Status''' tab.

[[Image:StatusM3UA_2.png]] 

3- To configure a periodic refresh of the M3UA status, select a value from '''Refresh Every'''.
* To view the M3UA SAPS, click the '''Saps''' tab.

[[Image:StatusM3UA_3.png]] 

4- To view the status of an M3UA SAP, select an M3UA SAP.

[[Image:StatusM3UA_4.png]] 

The M3UA SAP status is displayed.

[[Image:StatusM3UA_5.png]] 

5- To view the status of an M3UA Network, select an M3UA Network.

[[Image:StatusM3UA_6.png]] 

6- Select a userpart to view its detailed status.

[[Image:StatusM3UA_7.png]] 

Detailed status is displayed.

[[Image:StatusM3UA_8.png]] 

7- Select a PSP to view its detailed status.

[[Image:StatusM3UA_9.png]] 

Detailed PSP status is displayed.

[[Image:StatusM3UA_A.png]] 

8- Select a PRSV to view its detailed status.

[[Image:StatusM3UA_B.png]] 

Detailed PSRV status is displayed. (In this example local)

[[Image:StatusM3UA_C.png]] 

Detailed PSRV status is displayed. (In this example remote)

[[Image:StatusM3UA_D.png]] 

9- Select a Route to view its detailed status.

[[Image:StatusM3UA_E.png]] 

Detailed Route status is displayed. (In this example local)

[[Image:StatusM3UA_F.png]] 

Detailed Route status is displayed. (In this example remote)

[[Image:StatusM3UA_G.png]] 

<div class="mw-collapsible mw-collapsed" data-collapsetext="Northbound Interface" data-expandtext="Northbound Interface" style="width: 400px;">

'''Path'''
<pre>
/configurations/@[configuration_name]/m3ua_stacks/@[m3ua_name]/status
</pre>
</div>

Toolpack:SIGTRAN M3uaSgpStatus B

2021-01-04T08:39:20Z

William Wong: /* Status menu */

__FORCETOC__
<div style="font-size: 132%;">'''''Applies to version(s): v2.9, v2.10'''''</div>

{{DISPLAYTITLE:M3UA SGP Status}}

This article illustrates how to view M3UA SGP status. This can be done from either the Status menu or the Navigation panel.

===Status menu===

1- Click '''Status''' in the navigation panel.

[[Image:Status_0.png]] 

2- Click the '''SIGTRAN M3ua''' tab.

[[Image:TabM3UA_0.png]] 

* Note that SCTP is used as the Transport Layer for this protocol, if any firewall is sitting between Telcobridges device and Sigtran network, this firewall needs to be set to allow SCTP traffic to pass through, and enabling the basic connectivity for sigtran link to be up

The status of the M3UA protocol stack is displayed.

[[Image:StatusM3UA_SGP_0.png]]

===Navigation Panel===
1- Click '''M3UA''' from the navigation panel.

[[Image:StatusM3UA_1.png]] 

2- Click the '''Status''' tab.

[[Image:StatusM3UA_2.png]] 

3- To configure a periodic refresh of the M3UA status, select a value from '''Refresh Every'''.
* To view the M3UA SAPS, click the '''Saps''' tab.

[[Image:StatusM3UA_3.png]] 

4- To view the status of an M3UA SAP, select an M3UA SAP.

[[Image:StatusM3UA_4.png]] 

The M3UA SAP status is displayed.

[[Image:StatusM3UA_5.png]] 

5- To view the status of an M3UA Network, select an M3UA Network.

[[Image:StatusM3UA_6.png]] 

6- Select a userpart to view its detailed status.

[[Image:StatusM3UA_7.png]] 

Detailed status is displayed.

[[Image:StatusM3UA_8.png]] 

7- Select a PSP to view its detailed status.

[[Image:StatusM3UA_9.png]] 

Detailed PSP status is displayed.

[[Image:StatusM3UA_A.png]] 

8- Select a PRSV to view its detailed status.

[[Image:StatusM3UA_B.png]] 

Detailed PSRV status is displayed. (In this example local)

[[Image:StatusM3UA_C.png]] 

Detailed PSRV status is displayed. (In this example remote)

[[Image:StatusM3UA_D.png]] 

9- Select a Route to view its detailed status.

[[Image:StatusM3UA_E.png]] 

Detailed Route status is displayed. (In this example local)

[[Image:StatusM3UA_F.png]] 

Detailed Route status is displayed. (In this example remote)

[[Image:StatusM3UA_G.png]] 

<div class="mw-collapsible mw-collapsed" data-collapsetext="Northbound Interface" data-expandtext="Northbound Interface" style="width: 400px;">

'''Path'''
<pre>
/configurations/@[configuration_name]/m3ua_stacks/@[m3ua_name]/status
</pre>
</div>

Toolpack:SIGTRAN M3uaIPSPStatus B

2021-01-04T08:39:01Z

William Wong: /* Status menu */

__FORCETOC__
<div style="font-size: 132%;">'''''Applies to version(s): v2.9, v2.10'''''</div>

{{DISPLAYTITLE:M3UA IPSP Status}}

This article illustrates how to view M3UA IPSP status. This can be done from either the Status menu or the Navigation panel.

===Status menu===

1- Click '''Status''' in the navigation panel.

[[Image:Status_0.png]] 

2- Click the '''SIGTRAN M3ua''' tab.

[[Image:TabM3UA_0.png]] 

* Note that SCTP is used as the Transport Layer for this protocol, if any firewall is sitting between Telcobridges device and Sigtran network, this firewall needs to be set to allow SCTP traffic to pass through, and enabling the basic connectivity for sigtran link to be up

The status of the M3UA protocol stack is displayed.

[[Image:StatusM3UA_0.png]]

===Navigation Panel===
1- Click '''M3UA''' from the navigation panel.

[[Image:StatusM3UA_1.png]] 

2- Click the '''Status''' tab.

[[Image:StatusM3UA_2.png]] 

3- To configure a periodic refresh of the M3UA status, select a value from '''Refresh Every'''.
* To view the M3UA SAPS, click the '''Saps''' tab.

[[Image:StatusM3UA_3.png]] 

4- To view the status of an M3UA SAP, select an M3UA SAP.

[[Image:StatusM3UA_4.png]] 

The M3UA SAP status is displayed.

[[Image:StatusM3UA_5.png]] 

5- To view the status of an M3UA Network, select an M3UA Network.

[[Image:StatusM3UA_6.png]] 

6- Select a userpart to view its detailed status.

[[Image:StatusM3UA_7.png]] 

Detailed status is displayed.

[[Image:StatusM3UA_8.png]] 

7- Select a PSP to view its detailed status.

[[Image:StatusM3UA_9.png]] 

Detailed PSP status is displayed.

[[Image:StatusM3UA_A.png]] 

8- Select a PRSV to view its detailed status.

[[Image:StatusM3UA_B.png]] 

Detailed PSRV status is displayed. (In this example local)

[[Image:StatusM3UA_C.png]] 

Detailed PSRV status is displayed. (In this example remote)

[[Image:StatusM3UA_D.png]] 

9- Select a Route to view its detailed status.

[[Image:StatusM3UA_E.png]] 

Detailed Route status is displayed. (In this example local)

[[Image:StatusM3UA_F.png]] 

Detailed Route status is displayed. (In this example remote)

[[Image:StatusM3UA_G.png]] 

<div class="mw-collapsible mw-collapsed" data-collapsetext="Northbound Interface" data-expandtext="Northbound Interface" style="width: 400px;">

'''Path'''
<pre>
/configurations/@[configuration_name]/m3ua_stacks/@[m3ua_name]/status
</pre>
</div>

Toolpack:SIGTRAN M2paStatus B

2021-01-04T08:38:43Z

William Wong: /* Status menu */

__FORCETOC__
<div style="font-size: 132%;">'''''Applies to version(s): v2.9, v2.10'''''</div>

{{DISPLAYTITLE:M2PA Status}}

This article illustrates how to view M2PA status and to set a periodic refresh of the M2UA protocol stack. This is done from the Status menu and the Navigation panel.

===Status menu===

1- Click '''Status''' in the navigation panel.

[[Image:Status_0.png]] 

2- Click the '''SIGTRAN M2pa''' tab.

[[Image:TabM2PA_0.png]] 

* Note that SCTP is used as the Transport Layer for this protocol, if any firewall is sitting between Telcobridges device and Sigtran network, this firewall needs to be set to allow SCTP traffic to pass through, and enabling the basic connectivity for sigtran link to be up

The status of the M2UA protocol stack is displayed.

[[Image:StatusM2PA_0.png]]

===Navigation Panel===
1- Click '''M2PA''' from the navigation panel.

[[Image:StatusM2PA_1.png]] 

2- Click the '''Status''' tab.

[[Image:StatusM2PA_2.png]] 

3- The status of the M2PA SAPs is shown.

*To configure a periodic refresh of the M2PA status, select a value from '''Refresh Every'''.
*Select '''Links''' to view Saps and Links status.

[[Image:StatusM2PA_3.png]] 

4- The M2PA SAPS links are shown.
* Select any SAP link to view further status.

[[Image:StatusM2PA_4.png]] 

5-To view the SAP links, click the '''Links''' tab.

[[Image:StatusM2PA_5.png]] 

<div class="mw-collapsible mw-collapsed" data-collapsetext="Northbound Interface" data-expandtext="Northbound Interface" style="width: 400px;">

'''Path'''
<pre>
/configurations/@[configuration_name]/hardware_units/@[hardware_name]/m2pa_stacks/@[m2pa_name]/status
</pre>
</div>

Toolpack:SIGTRAN M2uaStatus B

2021-01-04T08:38:21Z

William Wong: /* Status menu */

__FORCETOC__
<div style="font-size: 132%;">'''''Applies to version(s): v2.9, v2.10'''''</div>

{{DISPLAYTITLE:M2UA Status}}

This article illustrates how to view M2UA status and to set a periodic refresh of the M2UA protocol stack. This is done from the Status menu and the Navigation panel.

===Status menu===

1- Click '''Status''' in the navigation panel.

[[Image:Status_0.png]] 

2- Click the '''SIGTRAN M2ua''' tab.

[[Image:TabM2UA_0.png]] 

* Note that SCTP is used as the Transport Layer for this protocol, if any firewall is sitting between Telcobridges device and Sigtran network, this firewall needs to be set to allow SCTP traffic to pass through, and enabling the basic connectivity for sigtran link to be up

The status of the M2UA protocol stack is displayed.

[[Image:StatusM2UA_0.png]]

===Navigation Panel===
1- Click '''M2UA''' from the navigation panel.

[[Image:StatusM2UA_1.png]] 

2- Click the '''Status''' tab.

[[Image:StatusM2UA_2.png]] 

3- To configure a periodic refresh of the M2UA status, select a value from '''Refresh Every'''.
* To view the SAP status select a SAP link.

[[Image:StatusM2UA_3.png]] 

4- The SAP status is displayed
* To view the Clusters status, click the '''Clusters''' tab.

[[Image:StatusM2UA_4.png]] 

5- Select an M2UA Cluster to view the status of its links and peers.

[[Image:StatusM2UA_5.png]] 

6- To view the status of the peers, select the '''Peers''' tab.

[[Image:StatusM2UA_6.png]] 

The status of the peers is displayed.

[[Image:StatusM2UA_7.png]] 

<div class="mw-collapsible mw-collapsed" data-collapsetext="Northbound Interface" data-expandtext="Northbound Interface" style="width: 400px;">

'''Path'''
<pre>
/configurations/@[configuration_name]/hardware_units/@[hardware_name]/m2ua_stacks/@[m2ua_stack]/status
</pre>
</div>

Toolpack:SIGTRAN M3uaASPStatus A

2021-01-04T08:36:39Z

William Wong: /* Status menu */

__FORCETOC__
<div style="font-size: 132%;">'''''Applies to version(s): v2.8.'''''</div>

{{DISPLAYTITLE:M3UA ASP Status}}

This article illustrates how to view M3UA ASP status. This can be done from either the Status menu or the Navigation panel.

===Status menu===

1- Click '''Status''' in the navigation panel.

[[Image:Status_0.png]] 

2- Click the '''SIGTRAN M3ua''' tab.

[[Image:TabM3UA_0.png]] 

* Note that SCTP is used as the Transport Layer for this protocol, if any firewall is sitting between Telcobridges device and Sigtran network, this firewall needs to be set to allow SCTP traffic to pass through, and enabling the basic connectivity for sigtran link to be up

The status of the M3UA protocol stack is displayed.

[[Image:StatusM3UA_SGP_0.png]]

===Navigation Panel===
1- Click '''M3UA''' from the navigation panel.

[[Image:StatusM3UA_1.png]] 

2- Click the '''Status''' tab.

[[Image:StatusM3UA_2.png]] 

3- To configure a periodic refresh of the M3UA status, select a value from '''Refresh Every'''.
* To view the M3UA SAPS, click the '''Saps''' tab.

[[Image:StatusM3UA_3.png]] 

4- To view the status of an M3UA SAP, select an M3UA SAP.

[[Image:StatusM3UA_4.png]] 

The M3UA SAP status is displayed.

[[Image:StatusM3UA_5.png]] 

5- To view the status of an M3UA Network, select an M3UA Network.

[[Image:StatusM3UA_6.png]] 

6- Select a userpart to view its detailed status.

[[Image:StatusM3UA_7.png]] 

Detailed status is displayed.

[[Image:StatusM3UA_8.png]] 

7- Select a PSP to view its detailed status.

[[Image:StatusM3UA_9.png]] 

Detailed PSP status is displayed.

[[Image:StatusM3UA_A.png]] 

8- Select a PRSV to view its detailed status.

[[Image:StatusM3UA_B.png]] 

Detailed PSRV status is displayed. (In this example local)

[[Image:StatusM3UA_C.png]] 

Detailed PSRV status is displayed. (In this example remote)

[[Image:StatusM3UA_D.png]] 

9- Select a Route to view its detailed status.

[[Image:StatusM3UA_E.png]] 

Detailed Route status is displayed. (In this example local)

[[Image:StatusM3UA_F.png]] 

Detailed Route status is displayed. (In this example remote)

[[Image:StatusM3UA_G.png]]

Toolpack:SIGTRAN M3uaSgpStatus A

2021-01-04T08:36:08Z

William Wong: /* Status menu */

__FORCETOC__
<div style="font-size: 132%;">'''''Applies to version(s): v2.8.'''''</div>

{{DISPLAYTITLE:M3UA SGP Status}}

This article illustrates how to view M3UA SGP status. This can be done from either the Status menu or the Navigation panel.

===Status menu===

1- Click '''Status''' in the navigation panel.

[[Image:Status_0.png]] 

2- Click the '''SIGTRAN M3ua''' tab.

[[Image:TabM3UA_0.png]] 

* Note that SCTP is used as the Transport Layer for this protocol, if any firewall is sitting between Telcobridges device and Sigtran network, this firewall needs to be set to allow SCTP traffic to pass through, and enabling the basic connectivity for sigtran link to be up

The status of the M3UA protocol stack is displayed.

[[Image:StatusM3UA_SGP_0.png]]

===Navigation Panel===
1- Click '''M3UA''' from the navigation panel.

[[Image:StatusM3UA_1.png]] 

2- Click the '''Status''' tab.

[[Image:StatusM3UA_2.png]] 

3- To configure a periodic refresh of the M3UA status, select a value from '''Refresh Every'''.
* To view the M3UA SAPS, click the '''Saps''' tab.

[[Image:StatusM3UA_3.png]] 

4- To view the status of an M3UA SAP, select an M3UA SAP.

[[Image:StatusM3UA_4.png]] 

The M3UA SAP status is displayed.

[[Image:StatusM3UA_5.png]] 

5- To view the status of an M3UA Network, select an M3UA Network.

[[Image:StatusM3UA_6.png]] 

6- Select a userpart to view its detailed status.

[[Image:StatusM3UA_7.png]] 

Detailed status is displayed.

[[Image:StatusM3UA_8.png]] 

7- Select a PSP to view its detailed status.

[[Image:StatusM3UA_9.png]] 

Detailed PSP status is displayed.

[[Image:StatusM3UA_A.png]] 

8- Select a PRSV to view its detailed status.

[[Image:StatusM3UA_B.png]] 

Detailed PSRV status is displayed. (In this example local)

[[Image:StatusM3UA_C.png]] 

Detailed PSRV status is displayed. (In this example remote)

[[Image:StatusM3UA_D.png]] 

9- Select a Route to view its detailed status.

[[Image:StatusM3UA_E.png]] 

Detailed Route status is displayed. (In this example local)

[[Image:StatusM3UA_F.png]] 

Detailed Route status is displayed. (In this example remote)

[[Image:StatusM3UA_G.png]]

Toolpack:SIGTRAN M3uaIPSPStatus A

2021-01-04T08:35:38Z

William Wong: /* Status menu */

__FORCETOC__
<div style="font-size: 132%;">'''''Applies to version(s): v2.8.'''''</div>

{{DISPLAYTITLE:M3UA IPSP Status}}

This article illustrates how to view M3UA IPSP status. This can be done from either the Status menu or the Navigation panel.

===Status menu===

1- Click '''Status''' in the navigation panel.

[[Image:Status_0.png]] 

2- Click the '''SIGTRAN M3ua''' tab.

[[Image:TabM3UA_0.png]] 

* Note that SCTP is used as the Transport Layer for this protocol, if any firewall is sitting between Telcobridges device and Sigtran network, this firewall needs to be set to allow SCTP traffic to pass through, and enabling the basic connectivity for sigtran link to be up

The status of the M3UA protocol stack is displayed.

[[Image:StatusM3UA_0.png]]

===Navigation Panel===
1- Click '''M3UA''' from the navigation panel.

[[Image:StatusM3UA_1.png]] 

2- Click the '''Status''' tab.

[[Image:StatusM3UA_2.png]] 

3- To configure a periodic refresh of the M3UA status, select a value from '''Refresh Every'''.
* To view the M3UA SAPS, click the '''Saps''' tab.

[[Image:StatusM3UA_3.png]] 

4- To view the status of an M3UA SAP, select an M3UA SAP.

[[Image:StatusM3UA_4.png]] 

The M3UA SAP status is displayed.

[[Image:StatusM3UA_5.png]] 

5- To view the status of an M3UA Network, select an M3UA Network.

[[Image:StatusM3UA_6.png]] 

6- Select a userpart to view its detailed status.

[[Image:StatusM3UA_7.png]] 

Detailed status is displayed.

[[Image:StatusM3UA_8.png]] 

7- Select a PSP to view its detailed status.

[[Image:StatusM3UA_9.png]] 

Detailed PSP status is displayed.

[[Image:StatusM3UA_A.png]] 

8- Select a PRSV to view its detailed status.

[[Image:StatusM3UA_B.png]] 

Detailed PSRV status is displayed. (In this example local)

[[Image:StatusM3UA_C.png]] 

Detailed PSRV status is displayed. (In this example remote)

[[Image:StatusM3UA_D.png]] 

9- Select a Route to view its detailed status.

[[Image:StatusM3UA_E.png]] 

Detailed Route status is displayed. (In this example local)

[[Image:StatusM3UA_F.png]] 

Detailed Route status is displayed. (In this example remote)

[[Image:StatusM3UA_G.png]]

Toolpack:SIGTRAN M2paStatus A

2021-01-04T08:34:53Z

William Wong: /* Status menu */

__FORCETOC__
<div style="font-size: 132%;">'''''Applies to version(s): v2.8.'''''</div>

{{DISPLAYTITLE:M2PA Status}}

This article illustrates how to view M2PA status and to set a periodic refresh of the M2UA protocol stack. This is done from the Status menu and the Navigation panel.

===Status menu===

1- Click '''Status''' in the navigation panel.

[[Image:Status_0.png]] 

2- Click the '''SIGTRAN M2pa''' tab.

[[Image:TabM2PA_0.png]] 

* Note that SCTP is used as the Transport Layer for this protocol, if any firewall is sitting between Telcobridges device and Sigtran network, this firewall needs to be set to allow SCTP traffic to pass through, and enabling the basic connectivity for sigtran link to be up

The status of the M2UA protocol stack is displayed.

[[Image:StatusM2PA_0.png]]

===Navigation Panel===
1- Click '''M2PA''' from the navigation panel.

[[Image:StatusM2PA_1.png]] 

2- Click the '''Status''' tab.

[[Image:StatusM2PA_2.png]] 

3- The status of the M2PA SAPs is shown.

*To configure a periodic refresh of the M2PA status, select a value from '''Refresh Every'''.
*Select '''Links''' to view Saps and Links status.

[[Image:StatusM2PA_3.png]] 

4- The M2PA SAPS links are shown.
* Select any SAP link to view further status.

[[Image:StatusM2PA_4.png]] 

5-To view the SAP links, click the '''Links''' tab.

[[Image:StatusM2PA_5.png]]

Toolpack:SIGTRAN M2paStatus A

2021-01-04T08:34:18Z

William Wong: /* Status menu */

__FORCETOC__
<div style="font-size: 132%;">'''''Applies to version(s): v2.8.'''''</div>

{{DISPLAYTITLE:M2PA Status}}

This article illustrates how to view M2PA status and to set a periodic refresh of the M2UA protocol stack. This is done from the Status menu and the Navigation panel.

===Status menu===

1- Click '''Status''' in the navigation panel.

[[Image:Status_0.png]] 

2- Click the '''SIGTRAN M2pa''' tab.

* Note that SCTP is used as the Transport Layer for this protocol, if any firewall is sitting between Telcobridges device and Sigtran network, this firewall needs to be set to allow SCTP traffic to pass through, and enabling the basic connectivity for sigtran link to be up

[[Image:TabM2PA_0.png]] 

The status of the M2UA protocol stack is displayed.

[[Image:StatusM2PA_0.png]]

===Navigation Panel===
1- Click '''M2PA''' from the navigation panel.

[[Image:StatusM2PA_1.png]] 

2- Click the '''Status''' tab.

[[Image:StatusM2PA_2.png]] 

3- The status of the M2PA SAPs is shown.

*To configure a periodic refresh of the M2PA status, select a value from '''Refresh Every'''.
*Select '''Links''' to view Saps and Links status.

[[Image:StatusM2PA_3.png]] 

4- The M2PA SAPS links are shown.
* Select any SAP link to view further status.

[[Image:StatusM2PA_4.png]] 

5-To view the SAP links, click the '''Links''' tab.

[[Image:StatusM2PA_5.png]]

Toolpack:SIGTRAN M2paStatus A

2021-01-04T08:34:00Z

William Wong: /* Status menu */

__FORCETOC__
<div style="font-size: 132%;">'''''Applies to version(s): v2.8.'''''</div>

{{DISPLAYTITLE:M2PA Status}}

This article illustrates how to view M2PA status and to set a periodic refresh of the M2UA protocol stack. This is done from the Status menu and the Navigation panel.

===Status menu===

1- Click '''Status''' in the navigation panel.

[[Image:Status_0.png]] 

2- Click the '''SIGTRAN M2pa''' tab.

* Note that SCTP is used as the Transport Layer for this protocol, if any firewall is sitting between Telcobridges device and Sigtran network, this firewall needs to be set to allow SCTP traffic to pass through, and enabling the basic connectivity for sigtran link to be up

[[Image:TabM2PA_0.png]] 

The status of the M2UA protocol stack is displayed.

[[Image:StatusM2PA_0.png]]

===Navigation Panel===
1- Click '''M2PA''' from the navigation panel.

[[Image:StatusM2PA_1.png]] 

2- Click the '''Status''' tab.

[[Image:StatusM2PA_2.png]] 

3- The status of the M2PA SAPs is shown.

*To configure a periodic refresh of the M2PA status, select a value from '''Refresh Every'''.
*Select '''Links''' to view Saps and Links status.

[[Image:StatusM2PA_3.png]] 

4- The M2PA SAPS links are shown.
* Select any SAP link to view further status.

[[Image:StatusM2PA_4.png]] 

5-To view the SAP links, click the '''Links''' tab.

[[Image:StatusM2PA_5.png]]

Toolpack:SIGTRAN M2uaStatus A

2021-01-04T08:32:42Z

William Wong: /* Status menu */

__FORCETOC__
<div style="font-size: 132%;">'''''Applies to version(s): v2.8.'''''</div>

{{DISPLAYTITLE:M2UA Status}}

This article illustrates how to view M2UA status and to set a periodic refresh of the M2UA protocol stack. This is done from the Status menu and the Navigation panel.

===Status menu===

1- Click '''Status''' in the navigation panel.

[[Image:Status_0.png]] 

2- Click the '''SIGTRAN M2ua''' tab.

[[Image:TabM2UA_0.png]] 

* Note that SCTP is used as the Transport Layer for this protocol, if any firewall is sitting between Telcobridges device and Sigtran network, this firewall needs to be set to allow SCTP traffic to pass through, and enabling the basic connectivity for sigtran link to be up

The status of the M2UA protocol stack is displayed.

[[Image:StatusM2UA_0.png]]

===Navigation Panel===
1- Click '''M2UA''' from the navigation panel.

[[Image:StatusM2UA_1.png]] 

2- Click the '''Status''' tab.

[[Image:StatusM2UA_2.png]] 

3- To configure a periodic refresh of the M2UA status, select a value from '''Refresh Every'''.
* To view the SAP status select a SAP link.

[[Image:StatusM2UA_3.png]] 

4- The SAP status is displayed
* To view the Clusters status, click the '''Clusters''' tab.

[[Image:StatusM2UA_4.png]] 

5- Select an M2UA Cluster to view the status of its links and peers.

[[Image:StatusM2UA_5.png]] 

6- To view the status of the peers, select the '''Peers''' tab.

[[Image:StatusM2UA_6.png]] 

The status of the peers is displayed.

[[Image:StatusM2UA_7.png]]

Toolpack:SIGTRAN M2uaStatus A

2021-01-04T08:31:49Z

William Wong: /* Status menu */

__FORCETOC__
<div style="font-size: 132%;">'''''Applies to version(s): v2.8.'''''</div>

{{DISPLAYTITLE:M2UA Status}}

This article illustrates how to view M2UA status and to set a periodic refresh of the M2UA protocol stack. This is done from the Status menu and the Navigation panel.

===Status menu===

1- Click '''Status''' in the navigation panel.

[[Image:Status_0.png]] 

2- Click the '''SIGTRAN M2ua''' tab.

[[Image:TabM2UA_0.png]] 

* Note that SCTP is used as the Transport Layer for this protocol, if any firewall is sitting between Telcobridges device and Sigtran network, this firewall needs to be set to allow SCTP traffic passing through, and enabling the basic connectivity for sigtran link to be up

The status of the M2UA protocol stack is displayed.

[[Image:StatusM2UA_0.png]]

===Navigation Panel===
1- Click '''M2UA''' from the navigation panel.

[[Image:StatusM2UA_1.png]] 

2- Click the '''Status''' tab.

[[Image:StatusM2UA_2.png]] 

3- To configure a periodic refresh of the M2UA status, select a value from '''Refresh Every'''.
* To view the SAP status select a SAP link.

[[Image:StatusM2UA_3.png]] 

4- The SAP status is displayed
* To view the Clusters status, click the '''Clusters''' tab.

[[Image:StatusM2UA_4.png]] 

5- Select an M2UA Cluster to view the status of its links and peers.

[[Image:StatusM2UA_5.png]] 

6- To view the status of the peers, select the '''Peers''' tab.

[[Image:StatusM2UA_6.png]] 

The status of the peers is displayed.

[[Image:StatusM2UA_7.png]]

FreeSBC Troubleshooting

2021-01-04T03:34:52Z

William Wong: /* One Way Audio/ No Audio Problems */

{{DISPLAYTITLE:FreeSBC Troubleshooting Guide}}

= Scheduling Problems =
You may see the SBC tab in the general status as yellow. When you select it: 
Status -> SBC
You may see “Scheduling problem sbc list” with the hostname of the FreeSBC. If you select it, you may see “Scheduling problem alarm” set to “true” 
Status containing “scheduling” problem may point to:
*Non-dedicated (or not ‘pinned’) CPU [Open-stack]
*Not enough CPU reservation [Vmware]
*Memory is not dedicated to the Virtual Machine (VM)
*Too many active virtual machines fighting for resources on the host
*On KVM based installations (proxmox,virtmanager) CPU type must set to “host”

You need to be sure other VMs are not taking resources from FreeSBC VM instance.
Follow requirements shown here: [[FreeSBC:Cloud:VmWare_Installation_A#Requirements|FreeSBC requirements]]

[[Image:SBC_Status.png|400px| ]]

= Registration Errors =

== Endpoint sends the register request to FreeSBC however the FreeSBC is not forwarding it to the registrar ==
*Check if SIP domain configured correctly 
[[Toolpack:Creating_a_SIP_Domain_SBC_A|Creating a SIP domain]]
*Check the SIP domain Status. Be sure domain registrar can be reached by FreeSBC
Go to Status -> SIP -> SIP Domain -> Status -> SIP Registration Domains
*Check if SIP client sends correct Domain name to FreeSBC. You can capture a SIP trace (see [[Toolpack_Debug_Application:Tbsigtrace|Signaling trace capture tool]] ) and use Wireshark to analyze the trace. Look at the “To:” SIP header: it must match what is in the Sıp Domain configuration of the FreeSBC.

== FreeSBC forwards incoming registration messages to the registrar but registrar returns an error ==

*Check if SIP domain configured with correct registrar setting
[[Toolpack:Creating_a_SIP_Registrar_SBC_A|Creating a SIP Registrar]]
*Check if Registrar NAP configured with a correct IP address
[[Toolpack:Allocating_a_SIP_Network_Access_Point_(NAP)_SBC_A|Allocating a SIP NAP]]
*Check if your client configured with the correct username and password
*Check the forwarding modes on FreeSBC, and select the correct one
===Forwarding Modes===
FreeSBC always modifies the contact URI in SIP register requests to remain on the path between SIP User Agents and registrars. FreeSBC supports two different SIP registration forwarding modes (i.e. "Contact Remapping" or "Contact Passthrough").

*The "Contact Passthrough" forwarding mode makes the contact username portion of the Contact URI in SIP register requests to pass through unchanged.
*The "Contact Remapping" forwarding mode modifies the contact username portion of the Contact URI in SIP register requests and make it unique.
[[Image:Forwarding_Modes.png|400px| ]]

===Forward original headers from incoming request ===
If your registrar wants to receive original headers from incoming requests, Forward the domain without any modification in From/To/Contact/P-Asserted-Identity SIP headers. If you want to forward incoming SIP from/to header domain to the outgoing leg you need to upgrade your FreeSBC to the minimum Release 3.0.114. With Release 3.0.114 we added forward_sip_domain script to our routing scripts.

This filter is used to forward the domain name (or IP address and port) from the incoming call to the outgoing call, for the following SIP headers: 
* from (update call attribute :calling by appending :calling_sip_host and calling_sip_port) 
* to (update call attribute :called by appending :called_sip_host and called_sip_port) 
* P-asserted-identity (update call attribute :private_address by appending :private_address_sip_host and private_address_sip_port) 

* To set up a Filter, the main script needs to be modified. The main script can be either simple_routing.rb, or any other script. 

First, go to the routing script section of the Web portal
<pre>Gateway -> Routing scripts -> Example Scripts -> simple_routing.rb [Edit]
</pre>
Three things need to be added. At the start of the script: 
<pre>require 'forward_sip_domain' unless defined?(ForwardSipDomain)</pre>
In the main class: 
<pre>include ForwardSipDomain
</pre>
<pre>
route_remap :method => :forward_sip_domain
</pre>
 
The final script will look like this (with possibly other filters also included):
<pre>
require 'base_routing'
require 'forward_sip_domain' unless defined?(ForwardSipDomain)

class SimpleRouting < BaseRouting
include ForwardSipDomain

route_match :call_field_name => :called
route_match :call_field_name => :calling
route_match :call_field_name => :private_address

route_match :call_field_name => :nap
route_remap :call_field_name => :called, :route_field_name => :remapped_called
route_remap :call_field_name => :calling, :route_field_name => :remapped_calling
route_remap :call_field_name => :private_address, :route_field_name => :remapped_private_address
route_remap :call_field_name => :nap, :route_field_name => :remapped_nap
route_remap :method => :forward_sip_domain

end
</pre>

* This script requires the routes to have a custom column named "forward_sip_domain", type boolean.

<pre>Gateway -> Routes -> Create New Route Column</pre>
'''Name''': forward_sip_domain '''Type attributes''': boolean 

See FreeSBC uses case: [[FreeSBC:Remote_Workers|Remote Workers]]

'''Note:''' This script is coming as default after Rls. 3.0.116. You just need to set the forward_sip_domain route column as enabled.

= One Way Audio/ No Audio Problems =

* Can place a call, but one way or no audio

One-way audio is a common VOIP problem. It is one of the most frequent support questions we receive. There are many possible causes;

* Outdated firmware in routers, VOIP phones, Firewalls, etc. can cause one-way audio. Ensure you have the latest updates for all the devices in the call path.
* Firewall mistakenly blocking RTP, be sure firewalls configured correctly.
* Particularly if Network Address Translation (NAT) is involved in the call path, configuration of the various devices may be a problem. Please check the link for [[NAT]]
* Another reason for one way audio is having your system set to offering unsupported codecs within your other SIP systems.
** Please check the following link for TB supported codecs [[Voice_codecs|Voice codecs]]
** Check if the codecs are configured correctly in the SDP [[Toolpack:Profile_SDP_Description|SDP Description]]
** Capture SIP and RTP traffic to see which codecs are used: [[VoIP_Ethernet_Capture_FreeSBC_A|FreeSBC VoIP Capture]]
* Just as each side of a call must send RTP within the same codec, each side must also have the same phase timing (or ptime value). See [[Toolpack:Profile_SDP_Description|SDP Description]]
* Configured RTP port ranges can cause a problem too. Check FreeSBC and endpoints (Clients, SIP Trunks) are using correct RTP ports. See [[Toolpack:Creating_an_IP_Port_Range_E|Creating an IP Port Range]]
* High one-way packet loss. If sufficient packet loss occurs in one direction on a call, that half of the conversion may break down, but not cause the entire call to drop. Packet loss can occur due to a number of reasons:
** High utilization on a link with no QoS.
** Misconfigured interface: Half-duplex or duplex mismatch.
** Underperforming network devices.
** Cabling faults.
* Wrong threshold configuration under SBC Advanced parameter can cause FreeSBC to block RTP-Audio by the firewall due to thresholds. You can see the following error in your call trace;
[[Image:data_path_errors.png|600px| ]] 
Please double check your thresholds configuration from;
<pre>SBC -> Advanced parameters
</pre>
* For Two-way no audio, sometimes phone/phone system is behind the external firewall, user may find that if they just establish the call, two-way audio is fine, but if the call is placed on hold for certain time, and when resuming the call, there is no more two-way audio. This could be due to the external firewall is set to shutdown the RTP path after certain time of being idle (silence) as in the case of call on hold. The solution is to optimize this setting at external firewall to prevent it from shutting down the RTP over a reasonable or defined time.

= Call Drops Problems =
My call dropped while I was talking, I might hear fast busy or just dead air.

* Call dropping within the first minute – missing ACK in Invite. Check the NAT configuration. [[NAT]]
* Maximum Call time exceeded. Many service providers set a limit on the maximum duration for any call passing through their system. Double-check with your service provider.

== Calls dropping at specific intervals (10 minutes, 30 minutes) ==
** Check the SIP session timers. [[SIP_session_timers|SIP Session timers]
** Uncheck “Use Session Timer” from SIP configuration. SIP -> Select the name from SIP Configuration menu -> Session Timers
[[Image:Session_timers.png|600px| ]]
* Call dropping within the first minute – missing ACK in Invite. Check the NAT configuration. [[NAT]]
* Maximum Call time exceeded. Many service providers set a limit on the maximum duration for any call passing through their system. Double-check with your service provider.

== Calls dropping because of Congestions ==
The calls can be dropped because of the congestion too. You can check the NAP status for congestions.

[[Image:congestion.png|800px| ]]

If you are receiving a congestion error,
* You first need to check the concurrent call counts. If your concurrent call count exceeded the license, you would have a congestion error. Please check your license.
* Check if you set any Call Rate Limiting in the NAP. Go to NAPS -> Select The NAP -> Advanced Parameters -> Call Rate Limiting
* For other congestion problems please contact support@telcobridges.com

= Sub-optimal config sbc list warning =

The reason for this is that some Nics (Azure, vmxnet3, e1000, ixgbe vf) report a RETA size of 0 while reporting multiple rx queues, meaning they can not support multiple CPUs. However TBRouter is not aware of this limit on those Nics then initiate the Cores on the Nics

Please check the following link for how to solve this issue [[Toolpack: How to Get Rid of Sub Optimal]]

= Log procedure for Signaling/Routing problems =

* Set trace levels to 1 using by tbx_cli_tools_remote command
** Connect to ssh and use tbx_cli_tools_remote command
** Select the application
** Set the trace level to 1 by typing "T"
** When you finished the test call please disconnect from the application, press Escape twice.
** Do this for the gateway, tbsyslog, toolpack_engine applications.
** You can get more detail about how to use tbx_cli_tools_remote from the following link [[How_to_use_tbx_cli_tools_remote_program]]
* Run tbsigtrace capture. For more details please check the following link [[Toolpack_Debug_Application:Tbsigtrace]]
* Make a test call or simulate the problem
* Stop the tbsigtrace capture
* Export the call trace. For more details please check the following link [[Toolpack:Retrieving_Call_Trace_C]]
* Generate a tbreport for the date/time you made the test call For more details please check the following link [[TBReport]]

= Log procedure for Voice/RTP problems =

* Set trace levels to 1 using by tbx_cli_tools_remote command
** Connect to ssh and use tbx_cli_tools_remote command
** Select the application
** Set the trace level to 1 by typing "T"
** When you finished the test call please disconnect from the application, press Escape twice.
** Do this for the gateway, tbsyslog, toolpaack_engine, and tbrouter applications.
** You can get more detail about how to use tbx_cli_tools_remote from the following link [[How_to_use_tbx_cli_tools_remote_program]]
* Run tbrouter capture. For more details please check the following link [[VoIP_Ethernet_Capture_FreeSBC_A]]
* Make a test call or simulate the problem
* Stop the tbrouter capture
* Export the call trace. For more details please check the following link [[Toolpack:Retrieving_Call_Trace_C]]
* Generate a tbreport for the date/time you made the test call For more details please check the following link [[TBReport]]

FreeSBC Troubleshooting

2021-01-04T03:33:30Z

William Wong: /* One Way Audio/ No Audio Problems */

{{DISPLAYTITLE:FreeSBC Troubleshooting Guide}}

= Scheduling Problems =
You may see the SBC tab in the general status as yellow. When you select it: 
Status -> SBC
You may see “Scheduling problem sbc list” with the hostname of the FreeSBC. If you select it, you may see “Scheduling problem alarm” set to “true” 
Status containing “scheduling” problem may point to:
*Non-dedicated (or not ‘pinned’) CPU [Open-stack]
*Not enough CPU reservation [Vmware]
*Memory is not dedicated to the Virtual Machine (VM)
*Too many active virtual machines fighting for resources on the host
*On KVM based installations (proxmox,virtmanager) CPU type must set to “host”

You need to be sure other VMs are not taking resources from FreeSBC VM instance.
Follow requirements shown here: [[FreeSBC:Cloud:VmWare_Installation_A#Requirements|FreeSBC requirements]]

[[Image:SBC_Status.png|400px| ]]

= Registration Errors =

== Endpoint sends the register request to FreeSBC however the FreeSBC is not forwarding it to the registrar ==
*Check if SIP domain configured correctly 
[[Toolpack:Creating_a_SIP_Domain_SBC_A|Creating a SIP domain]]
*Check the SIP domain Status. Be sure domain registrar can be reached by FreeSBC
Go to Status -> SIP -> SIP Domain -> Status -> SIP Registration Domains
*Check if SIP client sends correct Domain name to FreeSBC. You can capture a SIP trace (see [[Toolpack_Debug_Application:Tbsigtrace|Signaling trace capture tool]] ) and use Wireshark to analyze the trace. Look at the “To:” SIP header: it must match what is in the Sıp Domain configuration of the FreeSBC.

== FreeSBC forwards incoming registration messages to the registrar but registrar returns an error ==

*Check if SIP domain configured with correct registrar setting
[[Toolpack:Creating_a_SIP_Registrar_SBC_A|Creating a SIP Registrar]]
*Check if Registrar NAP configured with a correct IP address
[[Toolpack:Allocating_a_SIP_Network_Access_Point_(NAP)_SBC_A|Allocating a SIP NAP]]
*Check if your client configured with the correct username and password
*Check the forwarding modes on FreeSBC, and select the correct one
===Forwarding Modes===
FreeSBC always modifies the contact URI in SIP register requests to remain on the path between SIP User Agents and registrars. FreeSBC supports two different SIP registration forwarding modes (i.e. "Contact Remapping" or "Contact Passthrough").

*The "Contact Passthrough" forwarding mode makes the contact username portion of the Contact URI in SIP register requests to pass through unchanged.
*The "Contact Remapping" forwarding mode modifies the contact username portion of the Contact URI in SIP register requests and make it unique.
[[Image:Forwarding_Modes.png|400px| ]]

===Forward original headers from incoming request ===
If your registrar wants to receive original headers from incoming requests, Forward the domain without any modification in From/To/Contact/P-Asserted-Identity SIP headers. If you want to forward incoming SIP from/to header domain to the outgoing leg you need to upgrade your FreeSBC to the minimum Release 3.0.114. With Release 3.0.114 we added forward_sip_domain script to our routing scripts.

This filter is used to forward the domain name (or IP address and port) from the incoming call to the outgoing call, for the following SIP headers: 
* from (update call attribute :calling by appending :calling_sip_host and calling_sip_port) 
* to (update call attribute :called by appending :called_sip_host and called_sip_port) 
* P-asserted-identity (update call attribute :private_address by appending :private_address_sip_host and private_address_sip_port) 

* To set up a Filter, the main script needs to be modified. The main script can be either simple_routing.rb, or any other script. 

First, go to the routing script section of the Web portal
<pre>Gateway -> Routing scripts -> Example Scripts -> simple_routing.rb [Edit]
</pre>
Three things need to be added. At the start of the script: 
<pre>require 'forward_sip_domain' unless defined?(ForwardSipDomain)</pre>
In the main class: 
<pre>include ForwardSipDomain
</pre>
<pre>
route_remap :method => :forward_sip_domain
</pre>
 
The final script will look like this (with possibly other filters also included):
<pre>
require 'base_routing'
require 'forward_sip_domain' unless defined?(ForwardSipDomain)

class SimpleRouting < BaseRouting
include ForwardSipDomain

route_match :call_field_name => :called
route_match :call_field_name => :calling
route_match :call_field_name => :private_address

route_match :call_field_name => :nap
route_remap :call_field_name => :called, :route_field_name => :remapped_called
route_remap :call_field_name => :calling, :route_field_name => :remapped_calling
route_remap :call_field_name => :private_address, :route_field_name => :remapped_private_address
route_remap :call_field_name => :nap, :route_field_name => :remapped_nap
route_remap :method => :forward_sip_domain

end
</pre>

* This script requires the routes to have a custom column named "forward_sip_domain", type boolean.

<pre>Gateway -> Routes -> Create New Route Column</pre>
'''Name''': forward_sip_domain '''Type attributes''': boolean 

See FreeSBC uses case: [[FreeSBC:Remote_Workers|Remote Workers]]

'''Note:''' This script is coming as default after Rls. 3.0.116. You just need to set the forward_sip_domain route column as enabled.

= One Way Audio/ No Audio Problems =

* Can place a call, but one way or no audio

One-way audio is a common VOIP problem. It is one of the most frequent support questions we receive. There are many possible causes;

* Outdated firmware in routers, VOIP phones, Firewalls, etc. can cause one-way audio. Ensure you have the latest updates for all the devices in the call path.
* Firewall mistakenly blocking RTP, be sure firewalls configured correctly.
* Particularly if Network Address Translation (NAT) is involved in the call path, configuration of the various devices may be a problem. Please check the link for [[NAT]]
* Another reason for one way audio is having your system set to offering unsupported codecs within your other SIP systems.
** Please check the following link for TB supported codecs [[Voice_codecs|Voice codecs]]
** Check if the codecs are configured correctly in the SDP [[Toolpack:Profile_SDP_Description|SDP Description]]
** Capture SIP and RTP traffic to see which codecs are used: [[VoIP_Ethernet_Capture_FreeSBC_A|FreeSBC VoIP Capture]]
* Just as each side of a call must send RTP within the same codec, each side must also have the same phase timing (or ptime value). See [[Toolpack:Profile_SDP_Description|SDP Description]]
* Configured RTP port ranges can cause a problem too. Check FreeSBC and endpoints (Clients, SIP Trunks) are using correct RTP ports. See [[Toolpack:Creating_an_IP_Port_Range_E|Creating an IP Port Range]]
* High one-way packet loss. If sufficient packet loss occurs in one direction on a call, that half of the conversion may break down, but not cause the entire call to drop. Packet loss can occur due to a number of reasons:
** High utilization on a link with no QoS.
** Misconfigured interface: Half-duplex or duplex mismatch.
** Underperforming network devices.
** Cabling faults.
* Wrong threshold configuration under SBC Advanced parameter can cause FreeSBC to block RTP-Audio by the firewall due to thresholds. You can see the following error in your call trace;
[[Image:data_path_errors.png|600px| ]] 
Please double check your thresholds configuration from;
<pre>SBC -> Advanced parameters
</pre>
* For Two-way no audio, sometimes phone/phone system is behind the firewall, user may find that if they just establish the call, two-way audio is fine, but if the call is placed on hold for certain time, and when resuming the call, there is no more two-way audio. This could be due to the the external firewall is set to shutdown the RTP path after certain time of being idle (silence) as in the case of call on hold. The solution is to optimize this setting at external firewall to prevent it from shutting down the RTP over a reasonable or defined time.

= Call Drops Problems =
My call dropped while I was talking, I might hear fast busy or just dead air.

* Call dropping within the first minute – missing ACK in Invite. Check the NAT configuration. [[NAT]]
* Maximum Call time exceeded. Many service providers set a limit on the maximum duration for any call passing through their system. Double-check with your service provider.

== Calls dropping at specific intervals (10 minutes, 30 minutes) ==
** Check the SIP session timers. [[SIP_session_timers|SIP Session timers]
** Uncheck “Use Session Timer” from SIP configuration. SIP -> Select the name from SIP Configuration menu -> Session Timers
[[Image:Session_timers.png|600px| ]]
* Call dropping within the first minute – missing ACK in Invite. Check the NAT configuration. [[NAT]]
* Maximum Call time exceeded. Many service providers set a limit on the maximum duration for any call passing through their system. Double-check with your service provider.

== Calls dropping because of Congestions ==
The calls can be dropped because of the congestion too. You can check the NAP status for congestions.

[[Image:congestion.png|800px| ]]

If you are receiving a congestion error,
* You first need to check the concurrent call counts. If your concurrent call count exceeded the license, you would have a congestion error. Please check your license.
* Check if you set any Call Rate Limiting in the NAP. Go to NAPS -> Select The NAP -> Advanced Parameters -> Call Rate Limiting
* For other congestion problems please contact support@telcobridges.com

= Sub-optimal config sbc list warning =

The reason for this is that some Nics (Azure, vmxnet3, e1000, ixgbe vf) report a RETA size of 0 while reporting multiple rx queues, meaning they can not support multiple CPUs. However TBRouter is not aware of this limit on those Nics then initiate the Cores on the Nics

Please check the following link for how to solve this issue [[Toolpack: How to Get Rid of Sub Optimal]]

= Log procedure for Signaling/Routing problems =

* Set trace levels to 1 using by tbx_cli_tools_remote command
** Connect to ssh and use tbx_cli_tools_remote command
** Select the application
** Set the trace level to 1 by typing "T"
** When you finished the test call please disconnect from the application, press Escape twice.
** Do this for the gateway, tbsyslog, toolpack_engine applications.
** You can get more detail about how to use tbx_cli_tools_remote from the following link [[How_to_use_tbx_cli_tools_remote_program]]
* Run tbsigtrace capture. For more details please check the following link [[Toolpack_Debug_Application:Tbsigtrace]]
* Make a test call or simulate the problem
* Stop the tbsigtrace capture
* Export the call trace. For more details please check the following link [[Toolpack:Retrieving_Call_Trace_C]]
* Generate a tbreport for the date/time you made the test call For more details please check the following link [[TBReport]]

= Log procedure for Voice/RTP problems =

* Set trace levels to 1 using by tbx_cli_tools_remote command
** Connect to ssh and use tbx_cli_tools_remote command
** Select the application
** Set the trace level to 1 by typing "T"
** When you finished the test call please disconnect from the application, press Escape twice.
** Do this for the gateway, tbsyslog, toolpaack_engine, and tbrouter applications.
** You can get more detail about how to use tbx_cli_tools_remote from the following link [[How_to_use_tbx_cli_tools_remote_program]]
* Run tbrouter capture. For more details please check the following link [[VoIP_Ethernet_Capture_FreeSBC_A]]
* Make a test call or simulate the problem
* Stop the tbrouter capture
* Export the call trace. For more details please check the following link [[Toolpack:Retrieving_Call_Trace_C]]
* Generate a tbreport for the date/time you made the test call For more details please check the following link [[TBReport]]

FreeSBC:Hosted PBX:Configuration A

2020-10-21T08:40:57Z

William Wong: /* SBC Use Cases */

=== '''''Applies to version: v3.0''''' ===

{{DISPLAYTITLE:FreeSBC:Hosted PBX:Example}}
=Introduction=
FreeSBC Hosted PBX Example Configuration provides you with a step by step Hosted PBX Configuration of [[FreeSBC|FreeSbc]] systems, using the Web Portal configuration tool. Complete general installation configuration steps, before you begin configuring your specific scenario.

=Hosted PBX Example=

[[Image:Hosted_PBX_topo_1.png]] 

==Prerequisites==
[[FreeSBC|FreeSBC]] devices must be installed as described in their respective [[Product_Installation_SBC|installation guides]].

=IP Network Configuration=

==Virtual Port Configuration for Wide Area Network==
1. Select '''IP Interfaces''' from the navigation panel

[[Image:Create Voip Interface_Tsbc_0.png]] 
 

2. Click the '''Virtual Ports''' tab.
*Click '''Create New Virtual Port'''

[[Image:SIP_Trunking_VP_Create.png]] 

3. Configure the virtual port.

*Enter a name for the virtual port
*Select the host(s) to which the virtual port is assigned
*Select a physical port to which the virtual port is assigned

*Click '''Create'''

[[Image:SIP_Trunking_VP_WAN.png]] 

4. Create a VLAN that uses this virtual port
*Click '''Create new Host VLAN'''

[[Image:SIP_Trunking_Vlan_WAN.png]] 

5. Configure the new VLAN

*Enter a name for the VLAN
*If the port is to be used untagged, make sure '''Untagged''' is checked.
*If the port is to be used with a 802.1Q tag, uncheck '''Untagged''' and enter a VLAN ID.
*Click '''Create'''

[[Image:SIP_Trunking_Vlan_WAN_Interface_1.png]] 

OR

[[Image:SIP_Trunking_Vlan_WAN_Interface_2.png]] 

==Virtual Port Configuration for Local Area Network==
1. Select '''IP Interfaces''' from the navigation panel

[[Image:Create Voip Interface_Tsbc_0.png]] 
 

2. 2. Click the '''Virtual Ports''' tab.
*Click '''Create New Virtual Port'''

[[Image:SIP_Trunking_VP_Create_LAN.png]] 

3. Configure the virtual port.

*Enter a name for the virtual port
*Select the host(s) to which the virtual port is assigned
*Select a physical port to which the virtual port is assigned

*Click '''Create'''

[[Image:SIP_Trunking_VP_LAN.png]] 

4. Create a VLAN that uses this virtual port
*Click '''Create new Host VLAN'''

[[Image:SIP_Trunking_Vlan_LAN.png]] 

5. Configure the new VLAN

*Enter a name for the VLAN
*If the port is to be used untagged, make sure '''Untagged''' is checked.
*If the port is to be used with a 802.1Q tag, uncheck '''Untagged''' and enter a VLAN ID.
*Click '''Create'''

[[Image:SIP_Trunking_Vlan_LAN_Interface_1.png]] 

OR

[[Image:SIP_Trunking_Vlan_LAN_Interface_2.png]] 

==Configuring IP Interface for Wide Area Network==

1. Select '''IP Interfaces''' from the navigation panel:

[[Image:Create_IP_Interface_Tsbc_0.png]] 
 

2. Click the '''IP Interfaces''' tab.
*Click '''Create New IP Interface'''

[[Image:Create_IP_if_WAN_1.png]] 
 

3. Configure the IP interface:

*Enter a name for the interface
*Select 1 or more services to use for the IP interface (RTP and SIP).
*Select the '''Host VLAN''' from which IP packets will exit.
*Indicate whether or not to use DHCP to automatically get an IP address for this port. (selecting this option removes the need to enter and IP address, Netmask, and Gateway)
*Enter an '''IP address'''
*Enter a '''Netmask'''
*Enter a '''gateway address'''
*Click '''Save'''

[[Image:Create_IPif_WAN_1.png]] 
 

==Configuring IP Interface for Local Area Network==

1. Select '''IP Interfaces''' from the navigation panel:

[[Image:Create_IP_Interface_Tsbc_0.png]] 

2. Click the '''IP Interfaces''' tab.
*Click '''Create New IP Interface'''

[[Image:Create_IPif_LAN.png]] 

3. Configure the IP interface:

*Enter a name for the interface
*Select 1 or more services to use for the IP interface (RTP and SIP).
*Select the '''Host VLAN''' from which IP packets will exit.
*Indicate whether or not to use DHCP to automatically get an IP address for this port. (selecting this option removes the need to enter and IP address, Netmask, and Gateway)
*Enter an '''IP address'''
*Enter a '''Netmask'''
*Enter a '''gateway address'''
*Click '''Save'''

[[Image:Create_IP_if_LAN_1.png]] 

=SIP Stack Configuration=

You must configure SIP signaling for your system. The first step in doing so is to create a SIP stack:

1- Click '''SIP''' in the navigation panel

[[Image:SIP_NavigationMenu.png]]

2- Click '''Create New Sip'''

[[Image:Create_New_SIP.png]]

3- Create the new SIP stack:

* Verify that the box labeled '''Enabled''' is checked
* Enter a '''name''' for the stack
* Click '''Create'''

[[Image:SIP_Trunking_SIP_Stack.png]]

==SIP Transport Server Configuration for Wide Area Network==

1- Click '''SIP''' in the navigation panel

[[Image:SIP_NavigationMenu.png]]

2- Select a SIP stack for which you wish to create a transport server

[[Image:SIP_Config_List.png]]

3- Click '''Create New Transport Server'''

[[Image:WAN_SIP_TransportServer.png]]

4- Create the new SIP transport server:

* Enter a '''name''' for the server
* Select an appropriate '''port type'''
* Select an appropriate host '''IP interface'''
* Click '''Create'''

[[Image:WAN_SIP_TransportServer1.png]]

==SIP Transport Server Configuration for Local Area Network==

1- Click '''SIP''' in the navigation panel

[[Image:SIP_NavigationMenu.png]]

2- Select a SIP stack for which you wish to create a transport server

[[Image:SIP_Config_List.png]]

3- Click '''Create New Transport Server'''

[[Image:WAN_SIP_TransportServer.png]]

4- Create the new SIP transport server:

* Enter a '''name''' for the server
* Select an appropriate '''port type'''
* Select an appropriate host '''IP interface'''
* Click '''Create'''

[[Image:LAN_SIP_TransportServer1.png]]

=SIP NAP=

A Network Access Point or NAP represents the entry point to another network or destination peer (e.g. SIP proxy, SIP trunk, etc)

==SIP NAP Configuration for Wide Area Network==

To create a new NAP:

1- Click '''NAPs''' in the navigation panel

[[Image:NAP_NavigationMenu.png]]

2- Click '''Create New NAP'''

[[Image:Create_New_NAP_SIP.png]]

3- Create the new NAP:

* Enter a '''name''' for the NAP
* Click '''Create'''

[[Image:Open_NAP_Create.png]]

4- Verify that the '''NAP was successfully created''' message appears

[[Image:Create_New_NAP_SIP2.png]]

5- Associate a SIP transport server with the new NAP:

* Select a '''SIP Transport Server''' from the '''Available''' list
* Click '''"<<"''' to associate the '''WAN_SIP_TS''' with the NAP

[[Image:Open_NAP_Create_1.png]]

6- Disable proxy address:

[[Image:Open_NAP_Create_NoProxy.png]]

7- Enter full access in the access control list

* Enter an '''IP/MASK''' (use 0.0.0.0/0 to accept any addresses)
* Click '''"<<"''' to add in the list of Access Control

[[Image:Open_NAP_Create_ACL.png]]

8- Associate a Port range with the new NAP:

* Select a '''port range''' from the '''Available''' list
* Click '''"<<"''' to associate '''WAN_Vlan:0''' Port range with the NAP

[[Image:Open_NAP_Create_2.png]]

9 - Configure settings for the following parameter groups as required:
*Check '''Accept only authorized users''' option. With this option unchecked, FreeSBC will forward any INVITEs sent to the "Open NAP" to the PBX without first authenticating the user.
*Registration Parameters
*Authentication Parameters
*Network Address Translation
*SIP-I Parameters
*Advanced Parameters

*Click '''Save'''

[[Image:Open_NAP_Create_4_21.png]]

==SIP NAP Configuration for Local Area Network==

To create a new NAP:

1- Click '''NAPs''' in the navigation panel

[[Image:NAP_NavigationMenu.png]]

2- Click '''Create New NAP'''

[[Image:Create_New_NAP_SIP.png]]

3- Create the new NAP:

* Enter a '''name''' for the NAP
* Click '''Create'''

[[Image:SIP_Server_NAP_Create.png]]

4- Verify that the '''NAP was successfully created''' message appears

[[Image:Create_New_NAP_SIP2.png]]

5- Associate a SIP transport server with the new NAP:

* Select a '''SIP Transport Server''' from the '''Available''' list
* Click '''"<<"''' to associate the '''LAN_SIP_TS''' with the NAP

[[Image:SIP_Server_NAP_Create_1.png]]

6- Enter SIP Server proxy address:

[[Image:SIP_Server_NAP_Create_2.png]]

7- Associate a Port range with the new NAP:

* Select a '''port range''' from the '''Available''' list
* Click '''"<<"''' to associate '''LAN_Vlan:0''' Port range with the NAP

[[Image:SIP_Server_NAP_Create_3.png]]

9- Configure settings for the following parameter groups as required:
*Registration Parameters
*Authentication Parameters
*Network Address Translation
*SIP-I Parameters
*Advanced Parameters

*Click '''Save'''

[[Image:SIP_Server_NAP_Create_4.png]]

==Access Control List==

FreeSBC will automatically create Access Control List for each NAP you created.

[[Image:Remote_Workers_Access_Control_List_1.png]]

If you double-click one of the created ACL, you will see FreeSBC only accept the calls if source IP matches. In this sample; the calls from 192.168.1.10 will accepted only.

[[Image:SIP_Access_Control_List_2.png]]

=SIP DOMAIN=

A SIP domain represents a grouping of devices (or users) that can communicate with one another.
You must configure SIP Registration Domain for your system. The first step in doing so is to create a SIP Domain:

==Create New SIP Registration Domain==

1- Click '''SIP Domain''' in the navigation panel

[[Image:SIP_Domain_NavigationMenu.png]]

2- Click '''Create New Domain'''

[[Image:Create_SIP_Domain.png]]

3- Create the new Domain:

* Enter a configuration '''Name''' for this domain.
* Enter a '''Domain Name''' for the SIP Registration Domain (the domain can be an FQDN or an IP address)
* Set the number '''Maximum Registered Users''' for this domain
* Set the Expires value used by SBC when the remote device doesn't supply one ('''Default Contact Expire''')
* Select '''Routing Method''' the system will use to route calls to registered users (if enabled in routing scripts).
** '''Register source''': Sends SIP Invite to the registering source IP address.
** '''Contact''': Sends SIP Invite to the 'contact' from the Register message.
* Set the '''Default Contact Expiration''', this value will be used when no ''Expires'' value is supplied by the user agent.
* Set the '''Minimum Contact Expiration''', this is the minimum ''Expires'' value that can be supplied by a user agent. Lower values will be rejected with a 423 'Interval too brief' response.
* Set the '''Maximum Contact Expiration''', this is the maximum ''Expires'' value that can be supplied by a user agent. The higher value is replaced by this parameter.
* Forwarding Parameters:
** Select the Registration '''Forwarding Mode''' to the registrar:
*** '''Contact Remapping''': Changes the user and the IP address.
*** '''Contact Passthrough''': Doesn't change anything. Enables devices to be contacted directly without going through the SBC.
** Set '''Minimum Registrar Expiration''', this is the minimum ''Expires'' value sent by the SBC to the registrar. If a user agent 'Expires' value is greater than this parameter, the SBC will do rate adaptation between the user agent and the Registrar.
** Set '''Maximum Pending Register Forward''', the maximum number of simultaneous pending register requests allowed for this domain. New REGISTER request is being refused passed this threshold.
** Set '''Maximum Simultaneous Register Forward''', the maximum number of simultaneous active register requests allowed for this domain. New REGISTER request is being refused passed this threshold.
* Click '''Create'''

[[Image:register_domain_1.png]]

4- Verify that the '''Sip registration domain was successfully created''' message appears

[[Image:register_domain_2.png]]
 

5- If your registrar has multiple domains, you need to create all the domains one by one.

==Create New SIP Registration Registrar==

A SIP registrar represents a SIP endpoint that provides a location service.
You must configure SIP Registrar for your system. The first step in doing so is to select your SIP Domain:

1- Click on your domain in the '''SIP Domain List'''

[[Image:SIP_Domain_Select.png]]

2- Click '''Create New SIP Registration Registrar'''

[[Image:New_SIP_Registrar_Create.png]]

3- Create the new SIP Registration Registrar

* Enter a '''Name''' for the SIP Registration Registrar
* Select pre defined SIP Proxy NAP from drop-down menu
* Click '''Create'''

[[Image:SIP_Registrar_Select.png]]

4- Verify that the '''Sip registration registrar was successfully created''' message appears

[[Image:Create_New_SIP_Registrar2.png]]

==Associate a SIP Domain with the new NAP==
Associate a SIP Domain with the new NAP. If you have more then 1 registrar domain using the same registrar you can associate all of them with the NAP:

* Select a '''sip domain''' from the '''Available''' list
* Click '''"<<"''' to associate the SIP Domain with the NAP
[[Image:Associate_SIP_NAP.png]]

=Call Route=

You must set up call routing for your system. [[Call routing]] refers to the ability to route calls based on criteria such as origin, destination, time of day, service provider rates, and more.

==Route Configuration for Remote Phones to SIP Server==

1- Click '''Routes''' in the navigation panel

[[Image:CreateCallRoute_0.png]]

2- Click '''Create New Static Route'''

[[Image:CreateCallRoute_1.png]]

3- Create the new route:

* Enter a '''RoutesetName''' for the route
* Select '''SIP_REMOTE''', to match calls from Trunk NAP
* Select '''SIP_SERVER_NAP'''
* Click '''Create'''

[[Image:Route_Remote_to_Server.png]]

4- Verify that the '''" Route was successfully created"''' message appears and that the new route is listed in the Routeset list

[[Image:Route_Remote_to_Server_1.png]]

==Route Configuration for SIP Server to Remote Phones==

1- Click '''Routes''' in the navigation panel

[[Image:CreateCallRoute_0.png]]

2- Click '''Create New Static Route'''

[[Image:CreateCallRoute_1.png]]

3- Create the new route:

* Enter a '''RoutesetName''' for the route
* Select '''SIP_SERVER_NAP''', to match calls from Trunk NAP
* Select ''' (By registered user) '''
* Click '''Create'''

[[Image:Route_Server_to_Remote_2.png]]

4- Verify that the '''" Route was successfully created"''' message appears and that the new route is listed in the Routeset list

[[Image:Route_Server_to_Remote_3.png]]

=Activating the Configuration=

Changes made to the configuration of the FreeSBC units are stored in the OAM&P Configuration and Logging database. In order for changes to be used by the system, they must first be activated. This is done at the system level and accessed from the Navigation panel.

Check the following link for activating the configuration;

[[Toolpack:Activating_the_Configuration_D]]

=SBC Use Cases=
[[Toolpack:Tsbc Use Cases A|SBC Use Cases]] 
[[FreeSBC_Use_Cases|FreeSBC Use Cases]]

[[FreeSBC: Multiple Domains/Hosted PBXs Configuration|Multiple Domains/PBXs scenario]]

=SBC Tutorial Guide V3.0=
[[Tmedia Tsig Tdev Tutorial Guide v3.0|Version 3.0]]

[[Category:Revise on Major]]

FreeSBC:Hosted PBX:Configuration A

2020-10-21T08:40:43Z

William Wong: /* SBC Use Cases */

=== '''''Applies to version: v3.0''''' ===

{{DISPLAYTITLE:FreeSBC:Hosted PBX:Example}}
=Introduction=
FreeSBC Hosted PBX Example Configuration provides you with a step by step Hosted PBX Configuration of [[FreeSBC|FreeSbc]] systems, using the Web Portal configuration tool. Complete general installation configuration steps, before you begin configuring your specific scenario.

=Hosted PBX Example=

[[Image:Hosted_PBX_topo_1.png]] 

==Prerequisites==
[[FreeSBC|FreeSBC]] devices must be installed as described in their respective [[Product_Installation_SBC|installation guides]].

=IP Network Configuration=

==Virtual Port Configuration for Wide Area Network==
1. Select '''IP Interfaces''' from the navigation panel

[[Image:Create Voip Interface_Tsbc_0.png]] 
 

2. Click the '''Virtual Ports''' tab.
*Click '''Create New Virtual Port'''

[[Image:SIP_Trunking_VP_Create.png]] 

3. Configure the virtual port.

*Enter a name for the virtual port
*Select the host(s) to which the virtual port is assigned
*Select a physical port to which the virtual port is assigned

*Click '''Create'''

[[Image:SIP_Trunking_VP_WAN.png]] 

4. Create a VLAN that uses this virtual port
*Click '''Create new Host VLAN'''

[[Image:SIP_Trunking_Vlan_WAN.png]] 

5. Configure the new VLAN

*Enter a name for the VLAN
*If the port is to be used untagged, make sure '''Untagged''' is checked.
*If the port is to be used with a 802.1Q tag, uncheck '''Untagged''' and enter a VLAN ID.
*Click '''Create'''

[[Image:SIP_Trunking_Vlan_WAN_Interface_1.png]] 

OR

[[Image:SIP_Trunking_Vlan_WAN_Interface_2.png]] 

==Virtual Port Configuration for Local Area Network==
1. Select '''IP Interfaces''' from the navigation panel

[[Image:Create Voip Interface_Tsbc_0.png]] 
 

2. 2. Click the '''Virtual Ports''' tab.
*Click '''Create New Virtual Port'''

[[Image:SIP_Trunking_VP_Create_LAN.png]] 

3. Configure the virtual port.

*Enter a name for the virtual port
*Select the host(s) to which the virtual port is assigned
*Select a physical port to which the virtual port is assigned

*Click '''Create'''

[[Image:SIP_Trunking_VP_LAN.png]] 

4. Create a VLAN that uses this virtual port
*Click '''Create new Host VLAN'''

[[Image:SIP_Trunking_Vlan_LAN.png]] 

5. Configure the new VLAN

*Enter a name for the VLAN
*If the port is to be used untagged, make sure '''Untagged''' is checked.
*If the port is to be used with a 802.1Q tag, uncheck '''Untagged''' and enter a VLAN ID.
*Click '''Create'''

[[Image:SIP_Trunking_Vlan_LAN_Interface_1.png]] 

OR

[[Image:SIP_Trunking_Vlan_LAN_Interface_2.png]] 

==Configuring IP Interface for Wide Area Network==

1. Select '''IP Interfaces''' from the navigation panel:

[[Image:Create_IP_Interface_Tsbc_0.png]] 
 

2. Click the '''IP Interfaces''' tab.
*Click '''Create New IP Interface'''

[[Image:Create_IP_if_WAN_1.png]] 
 

3. Configure the IP interface:

*Enter a name for the interface
*Select 1 or more services to use for the IP interface (RTP and SIP).
*Select the '''Host VLAN''' from which IP packets will exit.
*Indicate whether or not to use DHCP to automatically get an IP address for this port. (selecting this option removes the need to enter and IP address, Netmask, and Gateway)
*Enter an '''IP address'''
*Enter a '''Netmask'''
*Enter a '''gateway address'''
*Click '''Save'''

[[Image:Create_IPif_WAN_1.png]] 
 

==Configuring IP Interface for Local Area Network==

1. Select '''IP Interfaces''' from the navigation panel:

[[Image:Create_IP_Interface_Tsbc_0.png]] 

2. Click the '''IP Interfaces''' tab.
*Click '''Create New IP Interface'''

[[Image:Create_IPif_LAN.png]] 

3. Configure the IP interface:

*Enter a name for the interface
*Select 1 or more services to use for the IP interface (RTP and SIP).
*Select the '''Host VLAN''' from which IP packets will exit.
*Indicate whether or not to use DHCP to automatically get an IP address for this port. (selecting this option removes the need to enter and IP address, Netmask, and Gateway)
*Enter an '''IP address'''
*Enter a '''Netmask'''
*Enter a '''gateway address'''
*Click '''Save'''

[[Image:Create_IP_if_LAN_1.png]] 

=SIP Stack Configuration=

You must configure SIP signaling for your system. The first step in doing so is to create a SIP stack:

1- Click '''SIP''' in the navigation panel

[[Image:SIP_NavigationMenu.png]]

2- Click '''Create New Sip'''

[[Image:Create_New_SIP.png]]

3- Create the new SIP stack:

* Verify that the box labeled '''Enabled''' is checked
* Enter a '''name''' for the stack
* Click '''Create'''

[[Image:SIP_Trunking_SIP_Stack.png]]

==SIP Transport Server Configuration for Wide Area Network==

1- Click '''SIP''' in the navigation panel

[[Image:SIP_NavigationMenu.png]]

2- Select a SIP stack for which you wish to create a transport server

[[Image:SIP_Config_List.png]]

3- Click '''Create New Transport Server'''

[[Image:WAN_SIP_TransportServer.png]]

4- Create the new SIP transport server:

* Enter a '''name''' for the server
* Select an appropriate '''port type'''
* Select an appropriate host '''IP interface'''
* Click '''Create'''

[[Image:WAN_SIP_TransportServer1.png]]

==SIP Transport Server Configuration for Local Area Network==

1- Click '''SIP''' in the navigation panel

[[Image:SIP_NavigationMenu.png]]

2- Select a SIP stack for which you wish to create a transport server

[[Image:SIP_Config_List.png]]

3- Click '''Create New Transport Server'''

[[Image:WAN_SIP_TransportServer.png]]

4- Create the new SIP transport server:

* Enter a '''name''' for the server
* Select an appropriate '''port type'''
* Select an appropriate host '''IP interface'''
* Click '''Create'''

[[Image:LAN_SIP_TransportServer1.png]]

=SIP NAP=

A Network Access Point or NAP represents the entry point to another network or destination peer (e.g. SIP proxy, SIP trunk, etc)

==SIP NAP Configuration for Wide Area Network==

To create a new NAP:

1- Click '''NAPs''' in the navigation panel

[[Image:NAP_NavigationMenu.png]]

2- Click '''Create New NAP'''

[[Image:Create_New_NAP_SIP.png]]

3- Create the new NAP:

* Enter a '''name''' for the NAP
* Click '''Create'''

[[Image:Open_NAP_Create.png]]

4- Verify that the '''NAP was successfully created''' message appears

[[Image:Create_New_NAP_SIP2.png]]

5- Associate a SIP transport server with the new NAP:

* Select a '''SIP Transport Server''' from the '''Available''' list
* Click '''"<<"''' to associate the '''WAN_SIP_TS''' with the NAP

[[Image:Open_NAP_Create_1.png]]

6- Disable proxy address:

[[Image:Open_NAP_Create_NoProxy.png]]

7- Enter full access in the access control list

* Enter an '''IP/MASK''' (use 0.0.0.0/0 to accept any addresses)
* Click '''"<<"''' to add in the list of Access Control

[[Image:Open_NAP_Create_ACL.png]]

8- Associate a Port range with the new NAP:

* Select a '''port range''' from the '''Available''' list
* Click '''"<<"''' to associate '''WAN_Vlan:0''' Port range with the NAP

[[Image:Open_NAP_Create_2.png]]

9 - Configure settings for the following parameter groups as required:
*Check '''Accept only authorized users''' option. With this option unchecked, FreeSBC will forward any INVITEs sent to the "Open NAP" to the PBX without first authenticating the user.
*Registration Parameters
*Authentication Parameters
*Network Address Translation
*SIP-I Parameters
*Advanced Parameters

*Click '''Save'''

[[Image:Open_NAP_Create_4_21.png]]

==SIP NAP Configuration for Local Area Network==

To create a new NAP:

1- Click '''NAPs''' in the navigation panel

[[Image:NAP_NavigationMenu.png]]

2- Click '''Create New NAP'''

[[Image:Create_New_NAP_SIP.png]]

3- Create the new NAP:

* Enter a '''name''' for the NAP
* Click '''Create'''

[[Image:SIP_Server_NAP_Create.png]]

4- Verify that the '''NAP was successfully created''' message appears

[[Image:Create_New_NAP_SIP2.png]]

5- Associate a SIP transport server with the new NAP:

* Select a '''SIP Transport Server''' from the '''Available''' list
* Click '''"<<"''' to associate the '''LAN_SIP_TS''' with the NAP

[[Image:SIP_Server_NAP_Create_1.png]]

6- Enter SIP Server proxy address:

[[Image:SIP_Server_NAP_Create_2.png]]

7- Associate a Port range with the new NAP:

* Select a '''port range''' from the '''Available''' list
* Click '''"<<"''' to associate '''LAN_Vlan:0''' Port range with the NAP

[[Image:SIP_Server_NAP_Create_3.png]]

9- Configure settings for the following parameter groups as required:
*Registration Parameters
*Authentication Parameters
*Network Address Translation
*SIP-I Parameters
*Advanced Parameters

*Click '''Save'''

[[Image:SIP_Server_NAP_Create_4.png]]

==Access Control List==

FreeSBC will automatically create Access Control List for each NAP you created.

[[Image:Remote_Workers_Access_Control_List_1.png]]

If you double-click one of the created ACL, you will see FreeSBC only accept the calls if source IP matches. In this sample; the calls from 192.168.1.10 will accepted only.

[[Image:SIP_Access_Control_List_2.png]]

=SIP DOMAIN=

A SIP domain represents a grouping of devices (or users) that can communicate with one another.
You must configure SIP Registration Domain for your system. The first step in doing so is to create a SIP Domain:

==Create New SIP Registration Domain==

1- Click '''SIP Domain''' in the navigation panel

[[Image:SIP_Domain_NavigationMenu.png]]

2- Click '''Create New Domain'''

[[Image:Create_SIP_Domain.png]]

3- Create the new Domain:

* Enter a configuration '''Name''' for this domain.
* Enter a '''Domain Name''' for the SIP Registration Domain (the domain can be an FQDN or an IP address)
* Set the number '''Maximum Registered Users''' for this domain
* Set the Expires value used by SBC when the remote device doesn't supply one ('''Default Contact Expire''')
* Select '''Routing Method''' the system will use to route calls to registered users (if enabled in routing scripts).
** '''Register source''': Sends SIP Invite to the registering source IP address.
** '''Contact''': Sends SIP Invite to the 'contact' from the Register message.
* Set the '''Default Contact Expiration''', this value will be used when no ''Expires'' value is supplied by the user agent.
* Set the '''Minimum Contact Expiration''', this is the minimum ''Expires'' value that can be supplied by a user agent. Lower values will be rejected with a 423 'Interval too brief' response.
* Set the '''Maximum Contact Expiration''', this is the maximum ''Expires'' value that can be supplied by a user agent. The higher value is replaced by this parameter.
* Forwarding Parameters:
** Select the Registration '''Forwarding Mode''' to the registrar:
*** '''Contact Remapping''': Changes the user and the IP address.
*** '''Contact Passthrough''': Doesn't change anything. Enables devices to be contacted directly without going through the SBC.
** Set '''Minimum Registrar Expiration''', this is the minimum ''Expires'' value sent by the SBC to the registrar. If a user agent 'Expires' value is greater than this parameter, the SBC will do rate adaptation between the user agent and the Registrar.
** Set '''Maximum Pending Register Forward''', the maximum number of simultaneous pending register requests allowed for this domain. New REGISTER request is being refused passed this threshold.
** Set '''Maximum Simultaneous Register Forward''', the maximum number of simultaneous active register requests allowed for this domain. New REGISTER request is being refused passed this threshold.
* Click '''Create'''

[[Image:register_domain_1.png]]

4- Verify that the '''Sip registration domain was successfully created''' message appears

[[Image:register_domain_2.png]]
 

5- If your registrar has multiple domains, you need to create all the domains one by one.

==Create New SIP Registration Registrar==

A SIP registrar represents a SIP endpoint that provides a location service.
You must configure SIP Registrar for your system. The first step in doing so is to select your SIP Domain:

1- Click on your domain in the '''SIP Domain List'''

[[Image:SIP_Domain_Select.png]]

2- Click '''Create New SIP Registration Registrar'''

[[Image:New_SIP_Registrar_Create.png]]

3- Create the new SIP Registration Registrar

* Enter a '''Name''' for the SIP Registration Registrar
* Select pre defined SIP Proxy NAP from drop-down menu
* Click '''Create'''

[[Image:SIP_Registrar_Select.png]]

4- Verify that the '''Sip registration registrar was successfully created''' message appears

[[Image:Create_New_SIP_Registrar2.png]]

==Associate a SIP Domain with the new NAP==
Associate a SIP Domain with the new NAP. If you have more then 1 registrar domain using the same registrar you can associate all of them with the NAP:

* Select a '''sip domain''' from the '''Available''' list
* Click '''"<<"''' to associate the SIP Domain with the NAP
[[Image:Associate_SIP_NAP.png]]

=Call Route=

You must set up call routing for your system. [[Call routing]] refers to the ability to route calls based on criteria such as origin, destination, time of day, service provider rates, and more.

==Route Configuration for Remote Phones to SIP Server==

1- Click '''Routes''' in the navigation panel

[[Image:CreateCallRoute_0.png]]

2- Click '''Create New Static Route'''

[[Image:CreateCallRoute_1.png]]

3- Create the new route:

* Enter a '''RoutesetName''' for the route
* Select '''SIP_REMOTE''', to match calls from Trunk NAP
* Select '''SIP_SERVER_NAP'''
* Click '''Create'''

[[Image:Route_Remote_to_Server.png]]

4- Verify that the '''" Route was successfully created"''' message appears and that the new route is listed in the Routeset list

[[Image:Route_Remote_to_Server_1.png]]

==Route Configuration for SIP Server to Remote Phones==

1- Click '''Routes''' in the navigation panel

[[Image:CreateCallRoute_0.png]]

2- Click '''Create New Static Route'''

[[Image:CreateCallRoute_1.png]]

3- Create the new route:

* Enter a '''RoutesetName''' for the route
* Select '''SIP_SERVER_NAP''', to match calls from Trunk NAP
* Select ''' (By registered user) '''
* Click '''Create'''

[[Image:Route_Server_to_Remote_2.png]]

4- Verify that the '''" Route was successfully created"''' message appears and that the new route is listed in the Routeset list

[[Image:Route_Server_to_Remote_3.png]]

=Activating the Configuration=

Changes made to the configuration of the FreeSBC units are stored in the OAM&P Configuration and Logging database. In order for changes to be used by the system, they must first be activated. This is done at the system level and accessed from the Navigation panel.

Check the following link for activating the configuration;

[[Toolpack:Activating_the_Configuration_D]]

=SBC Use Cases=
[[Toolpack:Tsbc Use Cases A|SBC Use Cases]] 
[[FreeSBC_Use_Cases|FreeSBC Use Cases]]
[[FreeSBC: Multiple Domains/Hosted PBXs Configuration|Multiple Domains/PBXs scenario]]

=SBC Tutorial Guide V3.0=
[[Tmedia Tsig Tdev Tutorial Guide v3.0|Version 3.0]]

[[Category:Revise on Major]]

FreeSBC:Hosted PBX:Configuration A

2020-10-21T08:40:08Z

William Wong: /* SBC Use Cases */

FreeSBC:Hosted PBX:Configuration A

2020-10-21T08:39:42Z

William Wong: /* SBC Use Cases */

=== '''''Applies to version: v3.0''''' ===

{{DISPLAYTITLE:FreeSBC:Hosted PBX:Example}}
=Introduction=
FreeSBC Hosted PBX Example Configuration provides you with a step by step Hosted PBX Configuration of [[FreeSBC|FreeSbc]] systems, using the Web Portal configuration tool. Complete general installation configuration steps, before you begin configuring your specific scenario.

=Hosted PBX Example=

[[Image:Hosted_PBX_topo_1.png]] 

==Prerequisites==
[[FreeSBC|FreeSBC]] devices must be installed as described in their respective [[Product_Installation_SBC|installation guides]].

=IP Network Configuration=

==Virtual Port Configuration for Wide Area Network==
1. Select '''IP Interfaces''' from the navigation panel

[[Image:Create Voip Interface_Tsbc_0.png]] 
 

2. Click the '''Virtual Ports''' tab.
*Click '''Create New Virtual Port'''

[[Image:SIP_Trunking_VP_Create.png]] 

3. Configure the virtual port.

*Enter a name for the virtual port
*Select the host(s) to which the virtual port is assigned
*Select a physical port to which the virtual port is assigned

*Click '''Create'''

[[Image:SIP_Trunking_VP_WAN.png]] 

4. Create a VLAN that uses this virtual port
*Click '''Create new Host VLAN'''

[[Image:SIP_Trunking_Vlan_WAN.png]] 

5. Configure the new VLAN

*Enter a name for the VLAN
*If the port is to be used untagged, make sure '''Untagged''' is checked.
*If the port is to be used with a 802.1Q tag, uncheck '''Untagged''' and enter a VLAN ID.
*Click '''Create'''

[[Image:SIP_Trunking_Vlan_WAN_Interface_1.png]] 

OR

[[Image:SIP_Trunking_Vlan_WAN_Interface_2.png]] 

==Virtual Port Configuration for Local Area Network==
1. Select '''IP Interfaces''' from the navigation panel

[[Image:Create Voip Interface_Tsbc_0.png]] 
 

2. 2. Click the '''Virtual Ports''' tab.
*Click '''Create New Virtual Port'''

[[Image:SIP_Trunking_VP_Create_LAN.png]] 

3. Configure the virtual port.

*Enter a name for the virtual port
*Select the host(s) to which the virtual port is assigned
*Select a physical port to which the virtual port is assigned

*Click '''Create'''

[[Image:SIP_Trunking_VP_LAN.png]] 

4. Create a VLAN that uses this virtual port
*Click '''Create new Host VLAN'''

[[Image:SIP_Trunking_Vlan_LAN.png]] 

5. Configure the new VLAN

*Enter a name for the VLAN
*If the port is to be used untagged, make sure '''Untagged''' is checked.
*If the port is to be used with a 802.1Q tag, uncheck '''Untagged''' and enter a VLAN ID.
*Click '''Create'''

[[Image:SIP_Trunking_Vlan_LAN_Interface_1.png]] 

OR

[[Image:SIP_Trunking_Vlan_LAN_Interface_2.png]] 

==Configuring IP Interface for Wide Area Network==

1. Select '''IP Interfaces''' from the navigation panel:

[[Image:Create_IP_Interface_Tsbc_0.png]] 
 

2. Click the '''IP Interfaces''' tab.
*Click '''Create New IP Interface'''

[[Image:Create_IP_if_WAN_1.png]] 
 

3. Configure the IP interface:

*Enter a name for the interface
*Select 1 or more services to use for the IP interface (RTP and SIP).
*Select the '''Host VLAN''' from which IP packets will exit.
*Indicate whether or not to use DHCP to automatically get an IP address for this port. (selecting this option removes the need to enter and IP address, Netmask, and Gateway)
*Enter an '''IP address'''
*Enter a '''Netmask'''
*Enter a '''gateway address'''
*Click '''Save'''

[[Image:Create_IPif_WAN_1.png]] 
 

==Configuring IP Interface for Local Area Network==

1. Select '''IP Interfaces''' from the navigation panel:

[[Image:Create_IP_Interface_Tsbc_0.png]] 

2. Click the '''IP Interfaces''' tab.
*Click '''Create New IP Interface'''

[[Image:Create_IPif_LAN.png]] 

3. Configure the IP interface:

*Enter a name for the interface
*Select 1 or more services to use for the IP interface (RTP and SIP).
*Select the '''Host VLAN''' from which IP packets will exit.
*Indicate whether or not to use DHCP to automatically get an IP address for this port. (selecting this option removes the need to enter and IP address, Netmask, and Gateway)
*Enter an '''IP address'''
*Enter a '''Netmask'''
*Enter a '''gateway address'''
*Click '''Save'''

[[Image:Create_IP_if_LAN_1.png]] 

=SIP Stack Configuration=

You must configure SIP signaling for your system. The first step in doing so is to create a SIP stack:

1- Click '''SIP''' in the navigation panel

[[Image:SIP_NavigationMenu.png]]

2- Click '''Create New Sip'''

[[Image:Create_New_SIP.png]]

3- Create the new SIP stack:

* Verify that the box labeled '''Enabled''' is checked
* Enter a '''name''' for the stack
* Click '''Create'''

[[Image:SIP_Trunking_SIP_Stack.png]]

==SIP Transport Server Configuration for Wide Area Network==

1- Click '''SIP''' in the navigation panel

[[Image:SIP_NavigationMenu.png]]

2- Select a SIP stack for which you wish to create a transport server

[[Image:SIP_Config_List.png]]

3- Click '''Create New Transport Server'''

[[Image:WAN_SIP_TransportServer.png]]

4- Create the new SIP transport server:

* Enter a '''name''' for the server
* Select an appropriate '''port type'''
* Select an appropriate host '''IP interface'''
* Click '''Create'''

[[Image:WAN_SIP_TransportServer1.png]]

==SIP Transport Server Configuration for Local Area Network==

1- Click '''SIP''' in the navigation panel

[[Image:SIP_NavigationMenu.png]]

2- Select a SIP stack for which you wish to create a transport server

[[Image:SIP_Config_List.png]]

3- Click '''Create New Transport Server'''

[[Image:WAN_SIP_TransportServer.png]]

4- Create the new SIP transport server:

* Enter a '''name''' for the server
* Select an appropriate '''port type'''
* Select an appropriate host '''IP interface'''
* Click '''Create'''

[[Image:LAN_SIP_TransportServer1.png]]

=SIP NAP=

A Network Access Point or NAP represents the entry point to another network or destination peer (e.g. SIP proxy, SIP trunk, etc)

==SIP NAP Configuration for Wide Area Network==

To create a new NAP:

1- Click '''NAPs''' in the navigation panel

[[Image:NAP_NavigationMenu.png]]

2- Click '''Create New NAP'''

[[Image:Create_New_NAP_SIP.png]]

3- Create the new NAP:

* Enter a '''name''' for the NAP
* Click '''Create'''

[[Image:Open_NAP_Create.png]]

4- Verify that the '''NAP was successfully created''' message appears

[[Image:Create_New_NAP_SIP2.png]]

5- Associate a SIP transport server with the new NAP:

* Select a '''SIP Transport Server''' from the '''Available''' list
* Click '''"<<"''' to associate the '''WAN_SIP_TS''' with the NAP

[[Image:Open_NAP_Create_1.png]]

6- Disable proxy address:

[[Image:Open_NAP_Create_NoProxy.png]]

7- Enter full access in the access control list

* Enter an '''IP/MASK''' (use 0.0.0.0/0 to accept any addresses)
* Click '''"<<"''' to add in the list of Access Control

[[Image:Open_NAP_Create_ACL.png]]

8- Associate a Port range with the new NAP:

* Select a '''port range''' from the '''Available''' list
* Click '''"<<"''' to associate '''WAN_Vlan:0''' Port range with the NAP

[[Image:Open_NAP_Create_2.png]]

9 - Configure settings for the following parameter groups as required:
*Check '''Accept only authorized users''' option. With this option unchecked, FreeSBC will forward any INVITEs sent to the "Open NAP" to the PBX without first authenticating the user.
*Registration Parameters
*Authentication Parameters
*Network Address Translation
*SIP-I Parameters
*Advanced Parameters

*Click '''Save'''

[[Image:Open_NAP_Create_4_21.png]]

==SIP NAP Configuration for Local Area Network==

To create a new NAP:

1- Click '''NAPs''' in the navigation panel

[[Image:NAP_NavigationMenu.png]]

2- Click '''Create New NAP'''

[[Image:Create_New_NAP_SIP.png]]

3- Create the new NAP:

* Enter a '''name''' for the NAP
* Click '''Create'''

[[Image:SIP_Server_NAP_Create.png]]

4- Verify that the '''NAP was successfully created''' message appears

[[Image:Create_New_NAP_SIP2.png]]

5- Associate a SIP transport server with the new NAP:

* Select a '''SIP Transport Server''' from the '''Available''' list
* Click '''"<<"''' to associate the '''LAN_SIP_TS''' with the NAP

[[Image:SIP_Server_NAP_Create_1.png]]

6- Enter SIP Server proxy address:

[[Image:SIP_Server_NAP_Create_2.png]]

7- Associate a Port range with the new NAP:

* Select a '''port range''' from the '''Available''' list
* Click '''"<<"''' to associate '''LAN_Vlan:0''' Port range with the NAP

[[Image:SIP_Server_NAP_Create_3.png]]

9- Configure settings for the following parameter groups as required:
*Registration Parameters
*Authentication Parameters
*Network Address Translation
*SIP-I Parameters
*Advanced Parameters

*Click '''Save'''

[[Image:SIP_Server_NAP_Create_4.png]]

==Access Control List==

FreeSBC will automatically create Access Control List for each NAP you created.

[[Image:Remote_Workers_Access_Control_List_1.png]]

If you double-click one of the created ACL, you will see FreeSBC only accept the calls if source IP matches. In this sample; the calls from 192.168.1.10 will accepted only.

[[Image:SIP_Access_Control_List_2.png]]

=SIP DOMAIN=

A SIP domain represents a grouping of devices (or users) that can communicate with one another.
You must configure SIP Registration Domain for your system. The first step in doing so is to create a SIP Domain:

==Create New SIP Registration Domain==

1- Click '''SIP Domain''' in the navigation panel

[[Image:SIP_Domain_NavigationMenu.png]]

2- Click '''Create New Domain'''

[[Image:Create_SIP_Domain.png]]

3- Create the new Domain:

* Enter a configuration '''Name''' for this domain.
* Enter a '''Domain Name''' for the SIP Registration Domain (the domain can be an FQDN or an IP address)
* Set the number '''Maximum Registered Users''' for this domain
* Set the Expires value used by SBC when the remote device doesn't supply one ('''Default Contact Expire''')
* Select '''Routing Method''' the system will use to route calls to registered users (if enabled in routing scripts).
** '''Register source''': Sends SIP Invite to the registering source IP address.
** '''Contact''': Sends SIP Invite to the 'contact' from the Register message.
* Set the '''Default Contact Expiration''', this value will be used when no ''Expires'' value is supplied by the user agent.
* Set the '''Minimum Contact Expiration''', this is the minimum ''Expires'' value that can be supplied by a user agent. Lower values will be rejected with a 423 'Interval too brief' response.
* Set the '''Maximum Contact Expiration''', this is the maximum ''Expires'' value that can be supplied by a user agent. The higher value is replaced by this parameter.
* Forwarding Parameters:
** Select the Registration '''Forwarding Mode''' to the registrar:
*** '''Contact Remapping''': Changes the user and the IP address.
*** '''Contact Passthrough''': Doesn't change anything. Enables devices to be contacted directly without going through the SBC.
** Set '''Minimum Registrar Expiration''', this is the minimum ''Expires'' value sent by the SBC to the registrar. If a user agent 'Expires' value is greater than this parameter, the SBC will do rate adaptation between the user agent and the Registrar.
** Set '''Maximum Pending Register Forward''', the maximum number of simultaneous pending register requests allowed for this domain. New REGISTER request is being refused passed this threshold.
** Set '''Maximum Simultaneous Register Forward''', the maximum number of simultaneous active register requests allowed for this domain. New REGISTER request is being refused passed this threshold.
* Click '''Create'''

[[Image:register_domain_1.png]]

4- Verify that the '''Sip registration domain was successfully created''' message appears

[[Image:register_domain_2.png]]
 

5- If your registrar has multiple domains, you need to create all the domains one by one.

==Create New SIP Registration Registrar==

A SIP registrar represents a SIP endpoint that provides a location service.
You must configure SIP Registrar for your system. The first step in doing so is to select your SIP Domain:

1- Click on your domain in the '''SIP Domain List'''

[[Image:SIP_Domain_Select.png]]

2- Click '''Create New SIP Registration Registrar'''

[[Image:New_SIP_Registrar_Create.png]]

3- Create the new SIP Registration Registrar

* Enter a '''Name''' for the SIP Registration Registrar
* Select pre defined SIP Proxy NAP from drop-down menu
* Click '''Create'''

[[Image:SIP_Registrar_Select.png]]

4- Verify that the '''Sip registration registrar was successfully created''' message appears

[[Image:Create_New_SIP_Registrar2.png]]

==Associate a SIP Domain with the new NAP==
Associate a SIP Domain with the new NAP. If you have more then 1 registrar domain using the same registrar you can associate all of them with the NAP:

* Select a '''sip domain''' from the '''Available''' list
* Click '''"<<"''' to associate the SIP Domain with the NAP
[[Image:Associate_SIP_NAP.png]]

=Call Route=

You must set up call routing for your system. [[Call routing]] refers to the ability to route calls based on criteria such as origin, destination, time of day, service provider rates, and more.

==Route Configuration for Remote Phones to SIP Server==

1- Click '''Routes''' in the navigation panel

[[Image:CreateCallRoute_0.png]]

2- Click '''Create New Static Route'''

[[Image:CreateCallRoute_1.png]]

3- Create the new route:

* Enter a '''RoutesetName''' for the route
* Select '''SIP_REMOTE''', to match calls from Trunk NAP
* Select '''SIP_SERVER_NAP'''
* Click '''Create'''

[[Image:Route_Remote_to_Server.png]]

4- Verify that the '''" Route was successfully created"''' message appears and that the new route is listed in the Routeset list

[[Image:Route_Remote_to_Server_1.png]]

==Route Configuration for SIP Server to Remote Phones==

1- Click '''Routes''' in the navigation panel

[[Image:CreateCallRoute_0.png]]

2- Click '''Create New Static Route'''

[[Image:CreateCallRoute_1.png]]

3- Create the new route:

* Enter a '''RoutesetName''' for the route
* Select '''SIP_SERVER_NAP''', to match calls from Trunk NAP
* Select ''' (By registered user) '''
* Click '''Create'''

[[Image:Route_Server_to_Remote_2.png]]

4- Verify that the '''" Route was successfully created"''' message appears and that the new route is listed in the Routeset list

[[Image:Route_Server_to_Remote_3.png]]

=Activating the Configuration=

Changes made to the configuration of the FreeSBC units are stored in the OAM&P Configuration and Logging database. In order for changes to be used by the system, they must first be activated. This is done at the system level and accessed from the Navigation panel.

Check the following link for activating the configuration;

[[Toolpack:Activating_the_Configuration_D]]

=SBC Use Cases=
[[Toolpack:Tsbc Use Cases A|SBC Use Cases]] 
[[FreeSBC_Use_Cases|FreeSBC Use Cases]]
[[FreeSBC: Multiple Domains/Hosted PBXs Configuration|Multiple Domains/PBXs scenario]]

=SBC Tutorial Guide V3.0=
[[Tmedia Tsig Tdev Tutorial Guide v3.0|Version 3.0]]

[[Category:Revise on Major]]

FreeSBC: Multiple Domains/Hosted PBXs Configuration

2020-10-21T08:37:42Z

William Wong: /* Route Configuration for SIP Server to Remote Phones */

=== '''''Applies to version: v3.0, 3.1''''' ===

{{DISPLAYTITLE:FreeSBC:Multiple Domains/Hosted PBXs Configuration:Example}}
=Introduction=
FreeSBC Hosted PBX Example (see [[FreeSBC:Hosted PBX:Configuration A|FreeSBC:Hosted PBX]]) provides a step by step Hosted PBX Configuration of [[FreeSBC|FreeSBC]] systems, using the Web Portal configuration tool. What will be discussed here, with guide for example configuration, is when there are multiple domains per PBX or several PBXs with different domains. This multiple domains/PBXs configuration is basically the extension of what is configured in single domain/hosted PBX, that including the setting of different domains, followed by specific routing for different domain destinations.

=Hosted PBXs Example=

[[Image:Hosted_Multiple_PBX_topo_1.png]] 

From above, there will be several Hosted PBXs like Hosted PBX1, Hosted PBX2, Hosted PBX3, etc. And each Hosted PBX will use its own domain like example1.com for PBX1, example2.com for PBX2, and so on. This multiple domain scenario can also be applied to single Hosted PBX with different several domains that will be using the similar configuration.

With external path over the WAN, different phones belonging to different Hosted PBXs register to domain corresponding to the PBX/Registrar the phone is subscribing to.

Prerequisites and below configuration sections need to be done as listed in [[FreeSBC:Hosted PBX:Configuration A|single Hosted PBX configuration]] before proceeding:

* IP Network Configuration
* SIP Stack Configuration

=SIP NAP=

A Network Access Point or NAP represents the entry point to another network or destination peer (e.g. SIP proxy, SIP trunk, etc)

==SIP NAP Configuration for Local Area Network==

To create a new NAP:

1- Click '''NAPs''' in the navigation panel

[[Image:NAP_NavigationMenu.png]]

2- Click '''Create New NAP'''

[[Image:Create_New_NAP_SIP.png]]

3- Create the new NAP:

* Enter a '''name''' for the NAP
* Click '''Create'''

[[Image:SIP_Server_NAP_Create.png]]

4- Verify that the '''NAP was successfully created''' message appears

[[Image:Create_New_NAP_SIP2.png]]

5- Associate a SIP transport server with the new NAP:

* Select a '''SIP Transport Server''' from the '''Available''' list
* Click '''"<<"''' to associate the '''LAN_SIP_TS''' with the NAP

[[Image:SIP_Server_NAP_Create_1.png]]

6- Enter SIP Server proxy address:

[[Image:SIP_Server_NAP_Create_2.png]]

7- Associate a Port range with the new NAP:

* Select a '''port range''' from the '''Available''' list
* Click '''"<<"''' to associate '''LAN_Vlan:0''' Port range with the NAP

[[Image:SIP_Server_NAP_Create_3.png]]

9- Configure settings for the following parameter groups as required:
*Registration Parameters
*Authentication Parameters
*Network Address Translation
*SIP-I Parameters
*Advanced Parameters

*Click '''Save'''

[[Image:SIP_Server_NAP_Create_4.png]]

* Here, we need to repeat above steps and create NAP for every Hosted PBX, for example, NAP_PBX1 (instead of SIP_SERVER_NAP) for PBX1 (192.168.1.10), NAP_PBX2 for PBX2 (192.168.1.11), and NAP_PBX3 for PBX3 (192.168.1.12) and so on if more.

==Access Control List==

FreeSBC will automatically create Access Control List for each NAP you created.

[[Image:Remote_Workers_Access_Control_List_1.png]]

If you double-click one of the created ACL, you will see FreeSBC only accept the calls if source IP matches. In this sample; the calls from 192.168.1.10, 192.168.1.11, 192.168.1.12 will be accepted only.

[[Image:SIP_Access_Control_List_2.png]]

Rules for NAP_PBX1, NAP_PBX2, NAP_PBX3,.. will be created automatically in the ACL.

=SIP DOMAIN=

A SIP domain represents a grouping of devices (or users) that can communicate with one another.
You must configure SIP Registration Domain for your system. The first step in doing so is to create a SIP Domain:

==Create New SIP Registration Domain==

1- Click '''SIP Domain''' in the navigation panel

[[Image:SIP_Domain_NavigationMenu.png]]

2- Click '''Create New Domain'''

[[Image:Create_SIP_Domain.png]]

3- Create the new Domain:

* Enter a configuration '''Name''' for this domain.
* Enter a '''Domain Name''' for the SIP Registration Domain (the domain can be an FQDN or an IP address)
* Set the number '''Maximum Registered Users''' for this domain
* Set the Expires value used by SBC when the remote device doesn't supply one ('''Default Contact Expire''')
* Select '''Routing Method''' the system will use to route calls to registered users (if enabled in routing scripts).
** '''Register source''': Sends SIP Invite to the registering source IP address.
** '''Contact''': Sends SIP Invite to the 'contact' from the Register message.
* Set the '''Default Contact Expiration''', this value will be used when no ''Expires'' value is supplied by the user agent.
* Set the '''Minimum Contact Expiration''', this is the minimum ''Expires'' value that can be supplied by a user agent. Lower values will be rejected with a 423 'Interval too brief' response.
* Set the '''Maximum Contact Expiration''', this is the maximum ''Expires'' value that can be supplied by a user agent. The higher value is replaced by this parameter.
* Forwarding Parameters:
** Select the Registration '''Forwarding Mode''' to the registrar:
*** '''Contact Remapping''': Changes the user and the IP address.
*** '''Contact Passthrough''': Doesn't change anything. Enables devices to be contacted directly without going through the SBC.
** Set '''Minimum Registrar Expiration''', this is the minimum ''Expires'' value sent by the SBC to the registrar. If a user agent 'Expires' value is greater than this parameter, the SBC will do rate adaptation between the user agent and the Registrar.
** Set '''Maximum Pending Register Forward''', the maximum number of simultaneous pending register requests allowed for this domain. New REGISTER request is being refused passed this threshold.
** Set '''Maximum Simultaneous Register Forward''', the maximum number of simultaneous active register requests allowed for this domain. New REGISTER request is being refused passed this threshold.
* Click '''Create'''

[[Image:register_domain_1.png]]

4- Verify that the '''Sip registration domain was successfully created''' message appears

[[Image:register_domain_2.png]]
 

5- If your registrar has multiple domains, you need to create all the domains one by one.

6- Repeat above steps for every domain designated to every PBX, for example, example1.com for PBX1, example2.com for PBX2, and so on. This also applies to the case of multiple domains from the same PBX/Registrar as pointed out in step 5 above.

==Create New SIP Registration Registrar==

A SIP registrar represents a SIP endpoint that provides a location service.
You must configure SIP Registrar for your system. The first step in doing so is to select your SIP Domain:

1- Click on your domain in the '''SIP Domain List'''

[[Image:SIP_Domain_Select.png]]

2- Click '''Create New SIP Registration Registrar'''

[[Image:New_SIP_Registrar_Create.png]]

3- Create the new SIP Registration Registrar

* Enter a '''Name''' for the SIP Registration Registrar
* Select pre defined SIP Proxy NAP from drop-down menu
* Click '''Create'''

[[Image:SIP_Registrar_Select.png]]

4- Verify that the '''Sip registration registrar was successfully created''' message appears

[[Image:Create_New_SIP_Registrar2.png]]

5- Different Registrars need to be created, so do the above steps for SIP_Registrar_1 for example1.com and choose NAP_PBX1 (192.168.1.10), SIP_Registrar_2 for example2.com and choose NAP_PBX2 (192.168.1.11), and SIP_Registrar_3 for example3.com and choose NAP_PBX3 (192.168.1.12).

==Associate a SIP Domain with the new NAP==
Associate a SIP Domain with the new NAP. If you have more then 1 registrar domain using the same registrar you can associate all of them with the NAP:

* Select a '''sip domain''' from the '''Available''' list
* Click '''"<<"''' to associate the SIP Domain with the NAP
[[Image:Associate_SIP_NAP.png]]

* For each domain, for example, example1.com, choose OPEN_NAP and NAP_PBX1, and for example2.com, choose OPEN_NAP and NAP_PBX2, and so on.

=Call Route=

You must set up call routing for your system. [[Call routing]] refers to the ability to route calls based on criteria such as origin, destination, time of day, service provider rates, and more.

==Route Configuration for Remote Phones to SIP Server==

1- Click '''Routes''' in the navigation panel

[[Image:CreateCallRoute_0.png]]

2- Click '''Create New Static Route'''

[[Image:CreateCallRoute_1.png]]

3- Create the new route:

* Enter a '''RoutesetName''' for the route
* Select '''SIP_REMOTE''', to match calls from Trunk NAP
* Select '''SIP_SERVER_NAP'''
* Click '''Create'''

[[Image:Route_Remote_to_Server.png]]

4- Verify that the '''" Route was successfully created"''' message appears and that the new route is listed in the Routeset list

[[Image:Route_Remote_to_Server_1.png]]

==Route Configuration for SIP Server to Remote Phones==

1- Click '''Routes''' in the navigation panel

[[Image:CreateCallRoute_0.png]]

2- Click '''Create New Static Route'''

[[Image:CreateCallRoute_1.png]]

3- Create the new route:

* Enter a '''RoutesetName''' for the route
* Select '''SIP_SERVER_NAP''', to match calls from Trunk NAP
* Select ''' (By registered user) '''
* Click '''Create'''

[[Image:Route_Server_to_Remote_2.png]]

4- Verify that the '''" Route was successfully created"''' message appears and that the new route is listed in the Routeset list

[[Image:Route_Server_to_Remote_3.png]]

5- Here, we are creating 3 pairs of route entry from above for 3 domains. In order for Registration/call to terminate correctly to the right NAP/PBX/Registrar, we need to set some condition to match specifically the registration/call

* for incoming call belonging to example1.com, on the route "called" field, can input /^([0-9]*)@example1.com/
* for incoming call belonging to example2.com, on the route "called" field, can input /^([0-9]*)@example2.com/
* for incoming call belonging to example2.com, on the route "called" field, can input /^([0-9]*)@example2.com/

[[Image:Multiple domains routes.png|800px|]]

=Activating the Configuration=

Changes made to the configuration of the FreeSBC units are stored in the OAM&P Configuration and Logging database. In order for changes to be used by the system, they must first be activated. This is done at the system level and accessed from the Navigation panel.

Check the following link for activating the configuration;

[[Toolpack:Activating_the_Configuration_D]]

=SBC Use Cases=
[[Toolpack:Tsbc Use Cases A|SBC Use Cases]] 
[[FreeSBC_Use_Cases|FreeSBC Use Cases]]

=SBC Tutorial Guide V3.0=
[[Tmedia Tsig Tdev Tutorial Guide v3.0|Version 3.0]]

[[Category:Revise on Major]]

FreeSBC: Multiple Domains/Hosted PBXs Configuration

2020-10-21T08:36:25Z

William Wong: /* Route Configuration for SIP Server to Remote Phones */

=== '''''Applies to version: v3.0, 3.1''''' ===

{{DISPLAYTITLE:FreeSBC:Multiple Domains/Hosted PBXs Configuration:Example}}
=Introduction=
FreeSBC Hosted PBX Example (see [[FreeSBC:Hosted PBX:Configuration A|FreeSBC:Hosted PBX]]) provides a step by step Hosted PBX Configuration of [[FreeSBC|FreeSBC]] systems, using the Web Portal configuration tool. What will be discussed here, with guide for example configuration, is when there are multiple domains per PBX or several PBXs with different domains. This multiple domains/PBXs configuration is basically the extension of what is configured in single domain/hosted PBX, that including the setting of different domains, followed by specific routing for different domain destinations.

=Hosted PBXs Example=

[[Image:Hosted_Multiple_PBX_topo_1.png]] 

From above, there will be several Hosted PBXs like Hosted PBX1, Hosted PBX2, Hosted PBX3, etc. And each Hosted PBX will use its own domain like example1.com for PBX1, example2.com for PBX2, and so on. This multiple domain scenario can also be applied to single Hosted PBX with different several domains that will be using the similar configuration.

With external path over the WAN, different phones belonging to different Hosted PBXs register to domain corresponding to the PBX/Registrar the phone is subscribing to.

Prerequisites and below configuration sections need to be done as listed in [[FreeSBC:Hosted PBX:Configuration A|single Hosted PBX configuration]] before proceeding:

* IP Network Configuration
* SIP Stack Configuration

=SIP NAP=

A Network Access Point or NAP represents the entry point to another network or destination peer (e.g. SIP proxy, SIP trunk, etc)

==SIP NAP Configuration for Local Area Network==

To create a new NAP:

1- Click '''NAPs''' in the navigation panel

[[Image:NAP_NavigationMenu.png]]

2- Click '''Create New NAP'''

[[Image:Create_New_NAP_SIP.png]]

3- Create the new NAP:

* Enter a '''name''' for the NAP
* Click '''Create'''

[[Image:SIP_Server_NAP_Create.png]]

4- Verify that the '''NAP was successfully created''' message appears

[[Image:Create_New_NAP_SIP2.png]]

5- Associate a SIP transport server with the new NAP:

* Select a '''SIP Transport Server''' from the '''Available''' list
* Click '''"<<"''' to associate the '''LAN_SIP_TS''' with the NAP

[[Image:SIP_Server_NAP_Create_1.png]]

6- Enter SIP Server proxy address:

[[Image:SIP_Server_NAP_Create_2.png]]

7- Associate a Port range with the new NAP:

* Select a '''port range''' from the '''Available''' list
* Click '''"<<"''' to associate '''LAN_Vlan:0''' Port range with the NAP

[[Image:SIP_Server_NAP_Create_3.png]]

9- Configure settings for the following parameter groups as required:
*Registration Parameters
*Authentication Parameters
*Network Address Translation
*SIP-I Parameters
*Advanced Parameters

*Click '''Save'''

[[Image:SIP_Server_NAP_Create_4.png]]

* Here, we need to repeat above steps and create NAP for every Hosted PBX, for example, NAP_PBX1 (instead of SIP_SERVER_NAP) for PBX1 (192.168.1.10), NAP_PBX2 for PBX2 (192.168.1.11), and NAP_PBX3 for PBX3 (192.168.1.12) and so on if more.

==Access Control List==

FreeSBC will automatically create Access Control List for each NAP you created.

[[Image:Remote_Workers_Access_Control_List_1.png]]

If you double-click one of the created ACL, you will see FreeSBC only accept the calls if source IP matches. In this sample; the calls from 192.168.1.10, 192.168.1.11, 192.168.1.12 will be accepted only.

[[Image:SIP_Access_Control_List_2.png]]

Rules for NAP_PBX1, NAP_PBX2, NAP_PBX3,.. will be created automatically in the ACL.

=SIP DOMAIN=

A SIP domain represents a grouping of devices (or users) that can communicate with one another.
You must configure SIP Registration Domain for your system. The first step in doing so is to create a SIP Domain:

==Create New SIP Registration Domain==

1- Click '''SIP Domain''' in the navigation panel

[[Image:SIP_Domain_NavigationMenu.png]]

2- Click '''Create New Domain'''

[[Image:Create_SIP_Domain.png]]

3- Create the new Domain:

* Enter a configuration '''Name''' for this domain.
* Enter a '''Domain Name''' for the SIP Registration Domain (the domain can be an FQDN or an IP address)
* Set the number '''Maximum Registered Users''' for this domain
* Set the Expires value used by SBC when the remote device doesn't supply one ('''Default Contact Expire''')
* Select '''Routing Method''' the system will use to route calls to registered users (if enabled in routing scripts).
** '''Register source''': Sends SIP Invite to the registering source IP address.
** '''Contact''': Sends SIP Invite to the 'contact' from the Register message.
* Set the '''Default Contact Expiration''', this value will be used when no ''Expires'' value is supplied by the user agent.
* Set the '''Minimum Contact Expiration''', this is the minimum ''Expires'' value that can be supplied by a user agent. Lower values will be rejected with a 423 'Interval too brief' response.
* Set the '''Maximum Contact Expiration''', this is the maximum ''Expires'' value that can be supplied by a user agent. The higher value is replaced by this parameter.
* Forwarding Parameters:
** Select the Registration '''Forwarding Mode''' to the registrar:
*** '''Contact Remapping''': Changes the user and the IP address.
*** '''Contact Passthrough''': Doesn't change anything. Enables devices to be contacted directly without going through the SBC.
** Set '''Minimum Registrar Expiration''', this is the minimum ''Expires'' value sent by the SBC to the registrar. If a user agent 'Expires' value is greater than this parameter, the SBC will do rate adaptation between the user agent and the Registrar.
** Set '''Maximum Pending Register Forward''', the maximum number of simultaneous pending register requests allowed for this domain. New REGISTER request is being refused passed this threshold.
** Set '''Maximum Simultaneous Register Forward''', the maximum number of simultaneous active register requests allowed for this domain. New REGISTER request is being refused passed this threshold.
* Click '''Create'''

[[Image:register_domain_1.png]]

4- Verify that the '''Sip registration domain was successfully created''' message appears

[[Image:register_domain_2.png]]
 

5- If your registrar has multiple domains, you need to create all the domains one by one.

6- Repeat above steps for every domain designated to every PBX, for example, example1.com for PBX1, example2.com for PBX2, and so on. This also applies to the case of multiple domains from the same PBX/Registrar as pointed out in step 5 above.

==Create New SIP Registration Registrar==

A SIP registrar represents a SIP endpoint that provides a location service.
You must configure SIP Registrar for your system. The first step in doing so is to select your SIP Domain:

1- Click on your domain in the '''SIP Domain List'''

[[Image:SIP_Domain_Select.png]]

2- Click '''Create New SIP Registration Registrar'''

[[Image:New_SIP_Registrar_Create.png]]

3- Create the new SIP Registration Registrar

* Enter a '''Name''' for the SIP Registration Registrar
* Select pre defined SIP Proxy NAP from drop-down menu
* Click '''Create'''

[[Image:SIP_Registrar_Select.png]]

4- Verify that the '''Sip registration registrar was successfully created''' message appears

[[Image:Create_New_SIP_Registrar2.png]]

5- Different Registrars need to be created, so do the above steps for SIP_Registrar_1 for example1.com and choose NAP_PBX1 (192.168.1.10), SIP_Registrar_2 for example2.com and choose NAP_PBX2 (192.168.1.11), and SIP_Registrar_3 for example3.com and choose NAP_PBX3 (192.168.1.12).

==Associate a SIP Domain with the new NAP==
Associate a SIP Domain with the new NAP. If you have more then 1 registrar domain using the same registrar you can associate all of them with the NAP:

* Select a '''sip domain''' from the '''Available''' list
* Click '''"<<"''' to associate the SIP Domain with the NAP
[[Image:Associate_SIP_NAP.png]]

* For each domain, for example, example1.com, choose OPEN_NAP and NAP_PBX1, and for example2.com, choose OPEN_NAP and NAP_PBX2, and so on.

=Call Route=

You must set up call routing for your system. [[Call routing]] refers to the ability to route calls based on criteria such as origin, destination, time of day, service provider rates, and more.

==Route Configuration for Remote Phones to SIP Server==

1- Click '''Routes''' in the navigation panel

[[Image:CreateCallRoute_0.png]]

2- Click '''Create New Static Route'''

[[Image:CreateCallRoute_1.png]]

3- Create the new route:

* Enter a '''RoutesetName''' for the route
* Select '''SIP_REMOTE''', to match calls from Trunk NAP
* Select '''SIP_SERVER_NAP'''
* Click '''Create'''

[[Image:Route_Remote_to_Server.png]]

4- Verify that the '''" Route was successfully created"''' message appears and that the new route is listed in the Routeset list

[[Image:Route_Remote_to_Server_1.png]]

==Route Configuration for SIP Server to Remote Phones==

1- Click '''Routes''' in the navigation panel

[[Image:CreateCallRoute_0.png]]

2- Click '''Create New Static Route'''

[[Image:CreateCallRoute_1.png]]

3- Create the new route:

* Enter a '''RoutesetName''' for the route
* Select '''SIP_SERVER_NAP''', to match calls from Trunk NAP
* Select ''' (By registered user) '''
* Click '''Create'''

[[Image:Route_Server_to_Remote_2.png]]

4- Verify that the '''" Route was successfully created"''' message appears and that the new route is listed in the Routeset list

[[Image:Route_Server_to_Remote_3.png]]

5- Here, we are creating 3 pairs of route entry from above for 3 domains. In order for Registration/call to terminate correctly to the right NAP/PBX/Registrar, we need to set some condition to match specifically the registration/call

* for incoming call belonging to example1.com, on the route "called" field, can input /^([0-9]*)@example1.com/
* for incoming call belonging to example2.com, on the route "called" field, can input /^([0-9]*)@example2.com/
* for incoming call belonging to example2.com, on the route "called" field, can input /^([0-9]*)@example2.com/

[[Image:Multiple domains routes.png]]

=Activating the Configuration=

Changes made to the configuration of the FreeSBC units are stored in the OAM&P Configuration and Logging database. In order for changes to be used by the system, they must first be activated. This is done at the system level and accessed from the Navigation panel.

Check the following link for activating the configuration;

[[Toolpack:Activating_the_Configuration_D]]

=SBC Use Cases=
[[Toolpack:Tsbc Use Cases A|SBC Use Cases]] 
[[FreeSBC_Use_Cases|FreeSBC Use Cases]]

=SBC Tutorial Guide V3.0=
[[Tmedia Tsig Tdev Tutorial Guide v3.0|Version 3.0]]

[[Category:Revise on Major]]

File:Multiple domains routes.png

2020-10-21T08:34:45Z

William Wong:

FreeSBC: Multiple Domains/Hosted PBXs Configuration

2020-10-21T08:12:23Z

William Wong: /* Route Configuration for SIP Server to Remote Phones */

=== '''''Applies to version: v3.0, 3.1''''' ===

{{DISPLAYTITLE:FreeSBC:Multiple Domains/Hosted PBXs Configuration:Example}}
=Introduction=
FreeSBC Hosted PBX Example (see [[FreeSBC:Hosted PBX:Configuration A|FreeSBC:Hosted PBX]]) provides a step by step Hosted PBX Configuration of [[FreeSBC|FreeSBC]] systems, using the Web Portal configuration tool. What will be discussed here, with guide for example configuration, is when there are multiple domains per PBX or several PBXs with different domains. This multiple domains/PBXs configuration is basically the extension of what is configured in single domain/hosted PBX, that including the setting of different domains, followed by specific routing for different domain destinations.

=Hosted PBXs Example=

[[Image:Hosted_Multiple_PBX_topo_1.png]] 

From above, there will be several Hosted PBXs like Hosted PBX1, Hosted PBX2, Hosted PBX3, etc. And each Hosted PBX will use its own domain like example1.com for PBX1, example2.com for PBX2, and so on. This multiple domain scenario can also be applied to single Hosted PBX with different several domains that will be using the similar configuration.

With external path over the WAN, different phones belonging to different Hosted PBXs register to domain corresponding to the PBX/Registrar the phone is subscribing to.

Prerequisites and below configuration sections need to be done as listed in [[FreeSBC:Hosted PBX:Configuration A|single Hosted PBX configuration]] before proceeding:

* IP Network Configuration
* SIP Stack Configuration

=SIP NAP=

A Network Access Point or NAP represents the entry point to another network or destination peer (e.g. SIP proxy, SIP trunk, etc)

==SIP NAP Configuration for Local Area Network==

To create a new NAP:

1- Click '''NAPs''' in the navigation panel

[[Image:NAP_NavigationMenu.png]]

2- Click '''Create New NAP'''

[[Image:Create_New_NAP_SIP.png]]

3- Create the new NAP:

* Enter a '''name''' for the NAP
* Click '''Create'''

[[Image:SIP_Server_NAP_Create.png]]

4- Verify that the '''NAP was successfully created''' message appears

[[Image:Create_New_NAP_SIP2.png]]

5- Associate a SIP transport server with the new NAP:

* Select a '''SIP Transport Server''' from the '''Available''' list
* Click '''"<<"''' to associate the '''LAN_SIP_TS''' with the NAP

[[Image:SIP_Server_NAP_Create_1.png]]

6- Enter SIP Server proxy address:

[[Image:SIP_Server_NAP_Create_2.png]]

7- Associate a Port range with the new NAP:

* Select a '''port range''' from the '''Available''' list
* Click '''"<<"''' to associate '''LAN_Vlan:0''' Port range with the NAP

[[Image:SIP_Server_NAP_Create_3.png]]

9- Configure settings for the following parameter groups as required:
*Registration Parameters
*Authentication Parameters
*Network Address Translation
*SIP-I Parameters
*Advanced Parameters

*Click '''Save'''

[[Image:SIP_Server_NAP_Create_4.png]]

* Here, we need to repeat above steps and create NAP for every Hosted PBX, for example, NAP_PBX1 (instead of SIP_SERVER_NAP) for PBX1 (192.168.1.10), NAP_PBX2 for PBX2 (192.168.1.11), and NAP_PBX3 for PBX3 (192.168.1.12) and so on if more.

==Access Control List==

FreeSBC will automatically create Access Control List for each NAP you created.

[[Image:Remote_Workers_Access_Control_List_1.png]]

If you double-click one of the created ACL, you will see FreeSBC only accept the calls if source IP matches. In this sample; the calls from 192.168.1.10, 192.168.1.11, 192.168.1.12 will be accepted only.

[[Image:SIP_Access_Control_List_2.png]]

Rules for NAP_PBX1, NAP_PBX2, NAP_PBX3,.. will be created automatically in the ACL.

=SIP DOMAIN=

A SIP domain represents a grouping of devices (or users) that can communicate with one another.
You must configure SIP Registration Domain for your system. The first step in doing so is to create a SIP Domain:

==Create New SIP Registration Domain==

1- Click '''SIP Domain''' in the navigation panel

[[Image:SIP_Domain_NavigationMenu.png]]

2- Click '''Create New Domain'''

[[Image:Create_SIP_Domain.png]]

3- Create the new Domain:

* Enter a configuration '''Name''' for this domain.
* Enter a '''Domain Name''' for the SIP Registration Domain (the domain can be an FQDN or an IP address)
* Set the number '''Maximum Registered Users''' for this domain
* Set the Expires value used by SBC when the remote device doesn't supply one ('''Default Contact Expire''')
* Select '''Routing Method''' the system will use to route calls to registered users (if enabled in routing scripts).
** '''Register source''': Sends SIP Invite to the registering source IP address.
** '''Contact''': Sends SIP Invite to the 'contact' from the Register message.
* Set the '''Default Contact Expiration''', this value will be used when no ''Expires'' value is supplied by the user agent.
* Set the '''Minimum Contact Expiration''', this is the minimum ''Expires'' value that can be supplied by a user agent. Lower values will be rejected with a 423 'Interval too brief' response.
* Set the '''Maximum Contact Expiration''', this is the maximum ''Expires'' value that can be supplied by a user agent. The higher value is replaced by this parameter.
* Forwarding Parameters:
** Select the Registration '''Forwarding Mode''' to the registrar:
*** '''Contact Remapping''': Changes the user and the IP address.
*** '''Contact Passthrough''': Doesn't change anything. Enables devices to be contacted directly without going through the SBC.
** Set '''Minimum Registrar Expiration''', this is the minimum ''Expires'' value sent by the SBC to the registrar. If a user agent 'Expires' value is greater than this parameter, the SBC will do rate adaptation between the user agent and the Registrar.
** Set '''Maximum Pending Register Forward''', the maximum number of simultaneous pending register requests allowed for this domain. New REGISTER request is being refused passed this threshold.
** Set '''Maximum Simultaneous Register Forward''', the maximum number of simultaneous active register requests allowed for this domain. New REGISTER request is being refused passed this threshold.
* Click '''Create'''

[[Image:register_domain_1.png]]

4- Verify that the '''Sip registration domain was successfully created''' message appears

[[Image:register_domain_2.png]]
 

5- If your registrar has multiple domains, you need to create all the domains one by one.

6- Repeat above steps for every domain designated to every PBX, for example, example1.com for PBX1, example2.com for PBX2, and so on. This also applies to the case of multiple domains from the same PBX/Registrar as pointed out in step 5 above.

==Create New SIP Registration Registrar==

A SIP registrar represents a SIP endpoint that provides a location service.
You must configure SIP Registrar for your system. The first step in doing so is to select your SIP Domain:

1- Click on your domain in the '''SIP Domain List'''

[[Image:SIP_Domain_Select.png]]

2- Click '''Create New SIP Registration Registrar'''

[[Image:New_SIP_Registrar_Create.png]]

3- Create the new SIP Registration Registrar

* Enter a '''Name''' for the SIP Registration Registrar
* Select pre defined SIP Proxy NAP from drop-down menu
* Click '''Create'''

[[Image:SIP_Registrar_Select.png]]

4- Verify that the '''Sip registration registrar was successfully created''' message appears

[[Image:Create_New_SIP_Registrar2.png]]

5- Different Registrars need to be created, so do the above steps for SIP_Registrar_1 for example1.com and choose NAP_PBX1 (192.168.1.10), SIP_Registrar_2 for example2.com and choose NAP_PBX2 (192.168.1.11), and SIP_Registrar_3 for example3.com and choose NAP_PBX3 (192.168.1.12).

==Associate a SIP Domain with the new NAP==
Associate a SIP Domain with the new NAP. If you have more then 1 registrar domain using the same registrar you can associate all of them with the NAP:

* Select a '''sip domain''' from the '''Available''' list
* Click '''"<<"''' to associate the SIP Domain with the NAP
[[Image:Associate_SIP_NAP.png]]

* For each domain, for example, example1.com, choose OPEN_NAP and NAP_PBX1, and for example2.com, choose OPEN_NAP and NAP_PBX2, and so on.

=Call Route=

You must set up call routing for your system. [[Call routing]] refers to the ability to route calls based on criteria such as origin, destination, time of day, service provider rates, and more.

==Route Configuration for Remote Phones to SIP Server==

1- Click '''Routes''' in the navigation panel

[[Image:CreateCallRoute_0.png]]

2- Click '''Create New Static Route'''

[[Image:CreateCallRoute_1.png]]

3- Create the new route:

* Enter a '''RoutesetName''' for the route
* Select '''SIP_REMOTE''', to match calls from Trunk NAP
* Select '''SIP_SERVER_NAP'''
* Click '''Create'''

[[Image:Route_Remote_to_Server.png]]

4- Verify that the '''" Route was successfully created"''' message appears and that the new route is listed in the Routeset list

[[Image:Route_Remote_to_Server_1.png]]

==Route Configuration for SIP Server to Remote Phones==

1- Click '''Routes''' in the navigation panel

[[Image:CreateCallRoute_0.png]]

2- Click '''Create New Static Route'''

[[Image:CreateCallRoute_1.png]]

3- Create the new route:

* Enter a '''RoutesetName''' for the route
* Select '''SIP_SERVER_NAP''', to match calls from Trunk NAP
* Select ''' (By registered user) '''
* Click '''Create'''

[[Image:Route_Server_to_Remote_2.png]]

4- Verify that the '''" Route was successfully created"''' message appears and that the new route is listed in the Routeset list

[[Image:Route_Server_to_Remote_3.png]]

5- Here, we are creating 3 pairs of route entry from above for 3 domains. In order for Registration/call to terminate correctly to the right NAP/PBX/Registrar, we need to set some condition to match specifically the registration/call

* for incoming call belonging to example1.com, on the route "called" field, can input /^([0-9]*)@example1.com/
* for incoming call belonging to example2.com, on the route "called" field, can input /^([0-9]*)@example2.com/
* for incoming call belonging to example2.com, on the route "called" field, can input /^([0-9]*)@example2.com/

=Activating the Configuration=

Changes made to the configuration of the FreeSBC units are stored in the OAM&P Configuration and Logging database. In order for changes to be used by the system, they must first be activated. This is done at the system level and accessed from the Navigation panel.

Check the following link for activating the configuration;

[[Toolpack:Activating_the_Configuration_D]]

=SBC Use Cases=
[[Toolpack:Tsbc Use Cases A|SBC Use Cases]] 
[[FreeSBC_Use_Cases|FreeSBC Use Cases]]

=SBC Tutorial Guide V3.0=
[[Tmedia Tsig Tdev Tutorial Guide v3.0|Version 3.0]]

[[Category:Revise on Major]]

FreeSBC: Multiple Domains/Hosted PBXs Configuration

2020-10-21T08:09:24Z

William Wong: /* Route Configuration for SIP Server to Remote Phones */

=== '''''Applies to version: v3.0, 3.1''''' ===

{{DISPLAYTITLE:FreeSBC:Multiple Domains/Hosted PBXs Configuration:Example}}
=Introduction=
FreeSBC Hosted PBX Example (see [[FreeSBC:Hosted PBX:Configuration A|FreeSBC:Hosted PBX]]) provides a step by step Hosted PBX Configuration of [[FreeSBC|FreeSBC]] systems, using the Web Portal configuration tool. What will be discussed here, with guide for example configuration, is when there are multiple domains per PBX or several PBXs with different domains. This multiple domains/PBXs configuration is basically the extension of what is configured in single domain/hosted PBX, that including the setting of different domains, followed by specific routing for different domain destinations.

=Hosted PBXs Example=

[[Image:Hosted_Multiple_PBX_topo_1.png]] 

From above, there will be several Hosted PBXs like Hosted PBX1, Hosted PBX2, Hosted PBX3, etc. And each Hosted PBX will use its own domain like example1.com for PBX1, example2.com for PBX2, and so on. This multiple domain scenario can also be applied to single Hosted PBX with different several domains that will be using the similar configuration.

With external path over the WAN, different phones belonging to different Hosted PBXs register to domain corresponding to the PBX/Registrar the phone is subscribing to.

Prerequisites and below configuration sections need to be done as listed in [[FreeSBC:Hosted PBX:Configuration A|single Hosted PBX configuration]] before proceeding:

* IP Network Configuration
* SIP Stack Configuration

=SIP NAP=

A Network Access Point or NAP represents the entry point to another network or destination peer (e.g. SIP proxy, SIP trunk, etc)

==SIP NAP Configuration for Local Area Network==

To create a new NAP:

1- Click '''NAPs''' in the navigation panel

[[Image:NAP_NavigationMenu.png]]

2- Click '''Create New NAP'''

[[Image:Create_New_NAP_SIP.png]]

3- Create the new NAP:

* Enter a '''name''' for the NAP
* Click '''Create'''

[[Image:SIP_Server_NAP_Create.png]]

4- Verify that the '''NAP was successfully created''' message appears

[[Image:Create_New_NAP_SIP2.png]]

5- Associate a SIP transport server with the new NAP:

* Select a '''SIP Transport Server''' from the '''Available''' list
* Click '''"<<"''' to associate the '''LAN_SIP_TS''' with the NAP

[[Image:SIP_Server_NAP_Create_1.png]]

6- Enter SIP Server proxy address:

[[Image:SIP_Server_NAP_Create_2.png]]

7- Associate a Port range with the new NAP:

* Select a '''port range''' from the '''Available''' list
* Click '''"<<"''' to associate '''LAN_Vlan:0''' Port range with the NAP

[[Image:SIP_Server_NAP_Create_3.png]]

9- Configure settings for the following parameter groups as required:
*Registration Parameters
*Authentication Parameters
*Network Address Translation
*SIP-I Parameters
*Advanced Parameters

*Click '''Save'''

[[Image:SIP_Server_NAP_Create_4.png]]

* Here, we need to repeat above steps and create NAP for every Hosted PBX, for example, NAP_PBX1 (instead of SIP_SERVER_NAP) for PBX1 (192.168.1.10), NAP_PBX2 for PBX2 (192.168.1.11), and NAP_PBX3 for PBX3 (192.168.1.12) and so on if more.

==Access Control List==

FreeSBC will automatically create Access Control List for each NAP you created.

[[Image:Remote_Workers_Access_Control_List_1.png]]

If you double-click one of the created ACL, you will see FreeSBC only accept the calls if source IP matches. In this sample; the calls from 192.168.1.10, 192.168.1.11, 192.168.1.12 will be accepted only.

[[Image:SIP_Access_Control_List_2.png]]

Rules for NAP_PBX1, NAP_PBX2, NAP_PBX3,.. will be created automatically in the ACL.

=SIP DOMAIN=

A SIP domain represents a grouping of devices (or users) that can communicate with one another.
You must configure SIP Registration Domain for your system. The first step in doing so is to create a SIP Domain:

==Create New SIP Registration Domain==

1- Click '''SIP Domain''' in the navigation panel

[[Image:SIP_Domain_NavigationMenu.png]]

2- Click '''Create New Domain'''

[[Image:Create_SIP_Domain.png]]

3- Create the new Domain:

* Enter a configuration '''Name''' for this domain.
* Enter a '''Domain Name''' for the SIP Registration Domain (the domain can be an FQDN or an IP address)
* Set the number '''Maximum Registered Users''' for this domain
* Set the Expires value used by SBC when the remote device doesn't supply one ('''Default Contact Expire''')
* Select '''Routing Method''' the system will use to route calls to registered users (if enabled in routing scripts).
** '''Register source''': Sends SIP Invite to the registering source IP address.
** '''Contact''': Sends SIP Invite to the 'contact' from the Register message.
* Set the '''Default Contact Expiration''', this value will be used when no ''Expires'' value is supplied by the user agent.
* Set the '''Minimum Contact Expiration''', this is the minimum ''Expires'' value that can be supplied by a user agent. Lower values will be rejected with a 423 'Interval too brief' response.
* Set the '''Maximum Contact Expiration''', this is the maximum ''Expires'' value that can be supplied by a user agent. The higher value is replaced by this parameter.
* Forwarding Parameters:
** Select the Registration '''Forwarding Mode''' to the registrar:
*** '''Contact Remapping''': Changes the user and the IP address.
*** '''Contact Passthrough''': Doesn't change anything. Enables devices to be contacted directly without going through the SBC.
** Set '''Minimum Registrar Expiration''', this is the minimum ''Expires'' value sent by the SBC to the registrar. If a user agent 'Expires' value is greater than this parameter, the SBC will do rate adaptation between the user agent and the Registrar.
** Set '''Maximum Pending Register Forward''', the maximum number of simultaneous pending register requests allowed for this domain. New REGISTER request is being refused passed this threshold.
** Set '''Maximum Simultaneous Register Forward''', the maximum number of simultaneous active register requests allowed for this domain. New REGISTER request is being refused passed this threshold.
* Click '''Create'''

[[Image:register_domain_1.png]]

4- Verify that the '''Sip registration domain was successfully created''' message appears

[[Image:register_domain_2.png]]
 

5- If your registrar has multiple domains, you need to create all the domains one by one.

6- Repeat above steps for every domain designated to every PBX, for example, example1.com for PBX1, example2.com for PBX2, and so on. This also applies to the case of multiple domains from the same PBX/Registrar as pointed out in step 5 above.

==Create New SIP Registration Registrar==

A SIP registrar represents a SIP endpoint that provides a location service.
You must configure SIP Registrar for your system. The first step in doing so is to select your SIP Domain:

1- Click on your domain in the '''SIP Domain List'''

[[Image:SIP_Domain_Select.png]]

2- Click '''Create New SIP Registration Registrar'''

[[Image:New_SIP_Registrar_Create.png]]

3- Create the new SIP Registration Registrar

* Enter a '''Name''' for the SIP Registration Registrar
* Select pre defined SIP Proxy NAP from drop-down menu
* Click '''Create'''

[[Image:SIP_Registrar_Select.png]]

4- Verify that the '''Sip registration registrar was successfully created''' message appears

[[Image:Create_New_SIP_Registrar2.png]]

5- Different Registrars need to be created, so do the above steps for SIP_Registrar_1 for example1.com and choose NAP_PBX1 (192.168.1.10), SIP_Registrar_2 for example2.com and choose NAP_PBX2 (192.168.1.11), and SIP_Registrar_3 for example3.com and choose NAP_PBX3 (192.168.1.12).

==Associate a SIP Domain with the new NAP==
Associate a SIP Domain with the new NAP. If you have more then 1 registrar domain using the same registrar you can associate all of them with the NAP:

* Select a '''sip domain''' from the '''Available''' list
* Click '''"<<"''' to associate the SIP Domain with the NAP
[[Image:Associate_SIP_NAP.png]]

* For each domain, for example, example1.com, choose OPEN_NAP and NAP_PBX1, and for example2.com, choose OPEN_NAP and NAP_PBX2, and so on.

=Call Route=

You must set up call routing for your system. [[Call routing]] refers to the ability to route calls based on criteria such as origin, destination, time of day, service provider rates, and more.

==Route Configuration for Remote Phones to SIP Server==

1- Click '''Routes''' in the navigation panel

[[Image:CreateCallRoute_0.png]]

2- Click '''Create New Static Route'''

[[Image:CreateCallRoute_1.png]]

3- Create the new route:

* Enter a '''RoutesetName''' for the route
* Select '''SIP_REMOTE''', to match calls from Trunk NAP
* Select '''SIP_SERVER_NAP'''
* Click '''Create'''

[[Image:Route_Remote_to_Server.png]]

4- Verify that the '''" Route was successfully created"''' message appears and that the new route is listed in the Routeset list

[[Image:Route_Remote_to_Server_1.png]]

==Route Configuration for SIP Server to Remote Phones==

1- Click '''Routes''' in the navigation panel

[[Image:CreateCallRoute_0.png]]

2- Click '''Create New Static Route'''

[[Image:CreateCallRoute_1.png]]

3- Create the new route:

* Enter a '''RoutesetName''' for the route
* Select '''SIP_SERVER_NAP''', to match calls from Trunk NAP
* Select ''' (By registered user) '''
* Click '''Create'''

[[Image:Route_Server_to_Remote_2.png]]

4- Verify that the '''" Route was successfully created"''' message appears and that the new route is listed in the Routeset list

[[Image:Route_Server_to_Remote_3.png]]

5- Here, we are creating 1 route entry for each domain, in total of 3 routes in this example for incoming call from OPEN_NAP. In order for Registration/call to terminate correctly to the right NAP/PBX/Registrar, we need to set some condition to match specifically the registration/call

* for incoming call belonging to example1.com, on the route "called" field, can input /^([0-9]*)@example1.com/
* for incoming call belonging to example2.com, on the route "called" field, can input /^([0-9]*)@example2.com/
* for incoming call belonging to example2.com, on the route "called" field, can input /^([0-9]*)@example2.com/

=Activating the Configuration=

Changes made to the configuration of the FreeSBC units are stored in the OAM&P Configuration and Logging database. In order for changes to be used by the system, they must first be activated. This is done at the system level and accessed from the Navigation panel.

Check the following link for activating the configuration;

[[Toolpack:Activating_the_Configuration_D]]

=SBC Use Cases=
[[Toolpack:Tsbc Use Cases A|SBC Use Cases]] 
[[FreeSBC_Use_Cases|FreeSBC Use Cases]]

=SBC Tutorial Guide V3.0=
[[Tmedia Tsig Tdev Tutorial Guide v3.0|Version 3.0]]

[[Category:Revise on Major]]

FreeSBC: Multiple Domains/Hosted PBXs Configuration

2020-10-21T07:59:22Z

William Wong: /* SIP NAP Configuration for Local Area Network */

=== '''''Applies to version: v3.0, 3.1''''' ===

{{DISPLAYTITLE:FreeSBC:Multiple Domains/Hosted PBXs Configuration:Example}}
=Introduction=
FreeSBC Hosted PBX Example (see [[FreeSBC:Hosted PBX:Configuration A|FreeSBC:Hosted PBX]]) provides a step by step Hosted PBX Configuration of [[FreeSBC|FreeSBC]] systems, using the Web Portal configuration tool. What will be discussed here, with guide for example configuration, is when there are multiple domains per PBX or several PBXs with different domains. This multiple domains/PBXs configuration is basically the extension of what is configured in single domain/hosted PBX, that including the setting of different domains, followed by specific routing for different domain destinations.

=Hosted PBXs Example=

[[Image:Hosted_Multiple_PBX_topo_1.png]] 

From above, there will be several Hosted PBXs like Hosted PBX1, Hosted PBX2, Hosted PBX3, etc. And each Hosted PBX will use its own domain like example1.com for PBX1, example2.com for PBX2, and so on. This multiple domain scenario can also be applied to single Hosted PBX with different several domains that will be using the similar configuration.

With external path over the WAN, different phones belonging to different Hosted PBXs register to domain corresponding to the PBX/Registrar the phone is subscribing to.

Prerequisites and below configuration sections need to be done as listed in [[FreeSBC:Hosted PBX:Configuration A|single Hosted PBX configuration]] before proceeding:

* IP Network Configuration
* SIP Stack Configuration

=SIP NAP=

A Network Access Point or NAP represents the entry point to another network or destination peer (e.g. SIP proxy, SIP trunk, etc)

==SIP NAP Configuration for Local Area Network==

To create a new NAP:

1- Click '''NAPs''' in the navigation panel

[[Image:NAP_NavigationMenu.png]]

2- Click '''Create New NAP'''

[[Image:Create_New_NAP_SIP.png]]

3- Create the new NAP:

* Enter a '''name''' for the NAP
* Click '''Create'''

[[Image:SIP_Server_NAP_Create.png]]

4- Verify that the '''NAP was successfully created''' message appears

[[Image:Create_New_NAP_SIP2.png]]

5- Associate a SIP transport server with the new NAP:

* Select a '''SIP Transport Server''' from the '''Available''' list
* Click '''"<<"''' to associate the '''LAN_SIP_TS''' with the NAP

[[Image:SIP_Server_NAP_Create_1.png]]

6- Enter SIP Server proxy address:

[[Image:SIP_Server_NAP_Create_2.png]]

7- Associate a Port range with the new NAP:

* Select a '''port range''' from the '''Available''' list
* Click '''"<<"''' to associate '''LAN_Vlan:0''' Port range with the NAP

[[Image:SIP_Server_NAP_Create_3.png]]

9- Configure settings for the following parameter groups as required:
*Registration Parameters
*Authentication Parameters
*Network Address Translation
*SIP-I Parameters
*Advanced Parameters

*Click '''Save'''

[[Image:SIP_Server_NAP_Create_4.png]]

* Here, we need to repeat above steps and create NAP for every Hosted PBX, for example, NAP_PBX1 (instead of SIP_SERVER_NAP) for PBX1 (192.168.1.10), NAP_PBX2 for PBX2 (192.168.1.11), and NAP_PBX3 for PBX3 (192.168.1.12) and so on if more.

==Access Control List==

FreeSBC will automatically create Access Control List for each NAP you created.

[[Image:Remote_Workers_Access_Control_List_1.png]]

If you double-click one of the created ACL, you will see FreeSBC only accept the calls if source IP matches. In this sample; the calls from 192.168.1.10, 192.168.1.11, 192.168.1.12 will be accepted only.

[[Image:SIP_Access_Control_List_2.png]]

Rules for NAP_PBX1, NAP_PBX2, NAP_PBX3,.. will be created automatically in the ACL.

=SIP DOMAIN=

A SIP domain represents a grouping of devices (or users) that can communicate with one another.
You must configure SIP Registration Domain for your system. The first step in doing so is to create a SIP Domain:

==Create New SIP Registration Domain==

1- Click '''SIP Domain''' in the navigation panel

[[Image:SIP_Domain_NavigationMenu.png]]

2- Click '''Create New Domain'''

[[Image:Create_SIP_Domain.png]]

3- Create the new Domain:

* Enter a configuration '''Name''' for this domain.
* Enter a '''Domain Name''' for the SIP Registration Domain (the domain can be an FQDN or an IP address)
* Set the number '''Maximum Registered Users''' for this domain
* Set the Expires value used by SBC when the remote device doesn't supply one ('''Default Contact Expire''')
* Select '''Routing Method''' the system will use to route calls to registered users (if enabled in routing scripts).
** '''Register source''': Sends SIP Invite to the registering source IP address.
** '''Contact''': Sends SIP Invite to the 'contact' from the Register message.
* Set the '''Default Contact Expiration''', this value will be used when no ''Expires'' value is supplied by the user agent.
* Set the '''Minimum Contact Expiration''', this is the minimum ''Expires'' value that can be supplied by a user agent. Lower values will be rejected with a 423 'Interval too brief' response.
* Set the '''Maximum Contact Expiration''', this is the maximum ''Expires'' value that can be supplied by a user agent. The higher value is replaced by this parameter.
* Forwarding Parameters:
** Select the Registration '''Forwarding Mode''' to the registrar:
*** '''Contact Remapping''': Changes the user and the IP address.
*** '''Contact Passthrough''': Doesn't change anything. Enables devices to be contacted directly without going through the SBC.
** Set '''Minimum Registrar Expiration''', this is the minimum ''Expires'' value sent by the SBC to the registrar. If a user agent 'Expires' value is greater than this parameter, the SBC will do rate adaptation between the user agent and the Registrar.
** Set '''Maximum Pending Register Forward''', the maximum number of simultaneous pending register requests allowed for this domain. New REGISTER request is being refused passed this threshold.
** Set '''Maximum Simultaneous Register Forward''', the maximum number of simultaneous active register requests allowed for this domain. New REGISTER request is being refused passed this threshold.
* Click '''Create'''

[[Image:register_domain_1.png]]

4- Verify that the '''Sip registration domain was successfully created''' message appears

[[Image:register_domain_2.png]]
 

5- If your registrar has multiple domains, you need to create all the domains one by one.

6- Repeat above steps for every domain designated to every PBX, for example, example1.com for PBX1, example2.com for PBX2, and so on. This also applies to the case of multiple domains from the same PBX/Registrar as pointed out in step 5 above.

==Create New SIP Registration Registrar==

A SIP registrar represents a SIP endpoint that provides a location service.
You must configure SIP Registrar for your system. The first step in doing so is to select your SIP Domain:

1- Click on your domain in the '''SIP Domain List'''

[[Image:SIP_Domain_Select.png]]

2- Click '''Create New SIP Registration Registrar'''

[[Image:New_SIP_Registrar_Create.png]]

3- Create the new SIP Registration Registrar

* Enter a '''Name''' for the SIP Registration Registrar
* Select pre defined SIP Proxy NAP from drop-down menu
* Click '''Create'''

[[Image:SIP_Registrar_Select.png]]

4- Verify that the '''Sip registration registrar was successfully created''' message appears

[[Image:Create_New_SIP_Registrar2.png]]

5- Different Registrars need to be created, so do the above steps for SIP_Registrar_1 for example1.com and choose NAP_PBX1 (192.168.1.10), SIP_Registrar_2 for example2.com and choose NAP_PBX2 (192.168.1.11), and SIP_Registrar_3 for example3.com and choose NAP_PBX3 (192.168.1.12).

==Associate a SIP Domain with the new NAP==
Associate a SIP Domain with the new NAP. If you have more then 1 registrar domain using the same registrar you can associate all of them with the NAP:

* Select a '''sip domain''' from the '''Available''' list
* Click '''"<<"''' to associate the SIP Domain with the NAP
[[Image:Associate_SIP_NAP.png]]

* For each domain, for example, example1.com, choose OPEN_NAP and NAP_PBX1, and for example2.com, choose OPEN_NAP and NAP_PBX2, and so on.

=Call Route=

You must set up call routing for your system. [[Call routing]] refers to the ability to route calls based on criteria such as origin, destination, time of day, service provider rates, and more.

==Route Configuration for Remote Phones to SIP Server==

1- Click '''Routes''' in the navigation panel

[[Image:CreateCallRoute_0.png]]

2- Click '''Create New Static Route'''

[[Image:CreateCallRoute_1.png]]

3- Create the new route:

* Enter a '''RoutesetName''' for the route
* Select '''SIP_REMOTE''', to match calls from Trunk NAP
* Select '''SIP_SERVER_NAP'''
* Click '''Create'''

[[Image:Route_Remote_to_Server.png]]

4- Verify that the '''" Route was successfully created"''' message appears and that the new route is listed in the Routeset list

[[Image:Route_Remote_to_Server_1.png]]

==Route Configuration for SIP Server to Remote Phones==

1- Click '''Routes''' in the navigation panel

[[Image:CreateCallRoute_0.png]]

2- Click '''Create New Static Route'''

[[Image:CreateCallRoute_1.png]]

3- Create the new route:

* Enter a '''RoutesetName''' for the route
* Select '''SIP_SERVER_NAP''', to match calls from Trunk NAP
* Select ''' (By registered user) '''
* Click '''Create'''

[[Image:Route_Server_to_Remote_2.png]]

4- Verify that the '''" Route was successfully created"''' message appears and that the new route is listed in the Routeset list

[[Image:Route_Server_to_Remote_3.png]]

=Activating the Configuration=

Changes made to the configuration of the FreeSBC units are stored in the OAM&P Configuration and Logging database. In order for changes to be used by the system, they must first be activated. This is done at the system level and accessed from the Navigation panel.

Check the following link for activating the configuration;

[[Toolpack:Activating_the_Configuration_D]]

=SBC Use Cases=
[[Toolpack:Tsbc Use Cases A|SBC Use Cases]] 
[[FreeSBC_Use_Cases|FreeSBC Use Cases]]

=SBC Tutorial Guide V3.0=
[[Tmedia Tsig Tdev Tutorial Guide v3.0|Version 3.0]]

[[Category:Revise on Major]]

FreeSBC: Multiple Domains/Hosted PBXs Configuration

2020-10-21T07:56:27Z

William Wong: /* Associate a SIP Domain with the new NAP */

=== '''''Applies to version: v3.0, 3.1''''' ===

{{DISPLAYTITLE:FreeSBC:Multiple Domains/Hosted PBXs Configuration:Example}}
=Introduction=
FreeSBC Hosted PBX Example (see [[FreeSBC:Hosted PBX:Configuration A|FreeSBC:Hosted PBX]]) provides a step by step Hosted PBX Configuration of [[FreeSBC|FreeSBC]] systems, using the Web Portal configuration tool. What will be discussed here, with guide for example configuration, is when there are multiple domains per PBX or several PBXs with different domains. This multiple domains/PBXs configuration is basically the extension of what is configured in single domain/hosted PBX, that including the setting of different domains, followed by specific routing for different domain destinations.

=Hosted PBXs Example=

[[Image:Hosted_Multiple_PBX_topo_1.png]] 

From above, there will be several Hosted PBXs like Hosted PBX1, Hosted PBX2, Hosted PBX3, etc. And each Hosted PBX will use its own domain like example1.com for PBX1, example2.com for PBX2, and so on. This multiple domain scenario can also be applied to single Hosted PBX with different several domains that will be using the similar configuration.

With external path over the WAN, different phones belonging to different Hosted PBXs register to domain corresponding to the PBX/Registrar the phone is subscribing to.

Prerequisites and below configuration sections need to be done as listed in [[FreeSBC:Hosted PBX:Configuration A|single Hosted PBX configuration]] before proceeding:

* IP Network Configuration
* SIP Stack Configuration

=SIP NAP=

A Network Access Point or NAP represents the entry point to another network or destination peer (e.g. SIP proxy, SIP trunk, etc)

==SIP NAP Configuration for Local Area Network==

To create a new NAP:

1- Click '''NAPs''' in the navigation panel

[[Image:NAP_NavigationMenu.png]]

2- Click '''Create New NAP'''

[[Image:Create_New_NAP_SIP.png]]

3- Create the new NAP:

* Enter a '''name''' for the NAP
* Click '''Create'''

[[Image:SIP_Server_NAP_Create.png]]

4- Verify that the '''NAP was successfully created''' message appears

[[Image:Create_New_NAP_SIP2.png]]

5- Associate a SIP transport server with the new NAP:

* Select a '''SIP Transport Server''' from the '''Available''' list
* Click '''"<<"''' to associate the '''LAN_SIP_TS''' with the NAP

[[Image:SIP_Server_NAP_Create_1.png]]

6- Enter SIP Server proxy address:

[[Image:SIP_Server_NAP_Create_2.png]]

7- Associate a Port range with the new NAP:

* Select a '''port range''' from the '''Available''' list
* Click '''"<<"''' to associate '''LAN_Vlan:0''' Port range with the NAP

[[Image:SIP_Server_NAP_Create_3.png]]

9- Configure settings for the following parameter groups as required:
*Registration Parameters
*Authentication Parameters
*Network Address Translation
*SIP-I Parameters
*Advanced Parameters

*Click '''Save'''

[[Image:SIP_Server_NAP_Create_4.png]]

* Here, we need to repeat above steps and create NAP for every Hosted PBX, for example, NAP_PBX1 for PBX1 (192.168.1.10), NAP_PBX2 for PBX2 (192.168.1.11), and NAP_PBX3 for PBX3 (192.168.1.12) and so on if more.

==Access Control List==

FreeSBC will automatically create Access Control List for each NAP you created.

[[Image:Remote_Workers_Access_Control_List_1.png]]

If you double-click one of the created ACL, you will see FreeSBC only accept the calls if source IP matches. In this sample; the calls from 192.168.1.10, 192.168.1.11, 192.168.1.12 will be accepted only.

[[Image:SIP_Access_Control_List_2.png]]

Rules for NAP_PBX1, NAP_PBX2, NAP_PBX3,.. will be created automatically in the ACL.

=SIP DOMAIN=

A SIP domain represents a grouping of devices (or users) that can communicate with one another.
You must configure SIP Registration Domain for your system. The first step in doing so is to create a SIP Domain:

==Create New SIP Registration Domain==

1- Click '''SIP Domain''' in the navigation panel

[[Image:SIP_Domain_NavigationMenu.png]]

2- Click '''Create New Domain'''

[[Image:Create_SIP_Domain.png]]

3- Create the new Domain:

* Enter a configuration '''Name''' for this domain.
* Enter a '''Domain Name''' for the SIP Registration Domain (the domain can be an FQDN or an IP address)
* Set the number '''Maximum Registered Users''' for this domain
* Set the Expires value used by SBC when the remote device doesn't supply one ('''Default Contact Expire''')
* Select '''Routing Method''' the system will use to route calls to registered users (if enabled in routing scripts).
** '''Register source''': Sends SIP Invite to the registering source IP address.
** '''Contact''': Sends SIP Invite to the 'contact' from the Register message.
* Set the '''Default Contact Expiration''', this value will be used when no ''Expires'' value is supplied by the user agent.
* Set the '''Minimum Contact Expiration''', this is the minimum ''Expires'' value that can be supplied by a user agent. Lower values will be rejected with a 423 'Interval too brief' response.
* Set the '''Maximum Contact Expiration''', this is the maximum ''Expires'' value that can be supplied by a user agent. The higher value is replaced by this parameter.
* Forwarding Parameters:
** Select the Registration '''Forwarding Mode''' to the registrar:
*** '''Contact Remapping''': Changes the user and the IP address.
*** '''Contact Passthrough''': Doesn't change anything. Enables devices to be contacted directly without going through the SBC.
** Set '''Minimum Registrar Expiration''', this is the minimum ''Expires'' value sent by the SBC to the registrar. If a user agent 'Expires' value is greater than this parameter, the SBC will do rate adaptation between the user agent and the Registrar.
** Set '''Maximum Pending Register Forward''', the maximum number of simultaneous pending register requests allowed for this domain. New REGISTER request is being refused passed this threshold.
** Set '''Maximum Simultaneous Register Forward''', the maximum number of simultaneous active register requests allowed for this domain. New REGISTER request is being refused passed this threshold.
* Click '''Create'''

[[Image:register_domain_1.png]]

4- Verify that the '''Sip registration domain was successfully created''' message appears

[[Image:register_domain_2.png]]
 

5- If your registrar has multiple domains, you need to create all the domains one by one.

6- Repeat above steps for every domain designated to every PBX, for example, example1.com for PBX1, example2.com for PBX2, and so on. This also applies to the case of multiple domains from the same PBX/Registrar as pointed out in step 5 above.

==Create New SIP Registration Registrar==

A SIP registrar represents a SIP endpoint that provides a location service.
You must configure SIP Registrar for your system. The first step in doing so is to select your SIP Domain:

1- Click on your domain in the '''SIP Domain List'''

[[Image:SIP_Domain_Select.png]]

2- Click '''Create New SIP Registration Registrar'''

[[Image:New_SIP_Registrar_Create.png]]

3- Create the new SIP Registration Registrar

* Enter a '''Name''' for the SIP Registration Registrar
* Select pre defined SIP Proxy NAP from drop-down menu
* Click '''Create'''

[[Image:SIP_Registrar_Select.png]]

4- Verify that the '''Sip registration registrar was successfully created''' message appears

[[Image:Create_New_SIP_Registrar2.png]]

5- Different Registrars need to be created, so do the above steps for SIP_Registrar_1 for example1.com and choose NAP_PBX1 (192.168.1.10), SIP_Registrar_2 for example2.com and choose NAP_PBX2 (192.168.1.11), and SIP_Registrar_3 for example3.com and choose NAP_PBX3 (192.168.1.12).

==Associate a SIP Domain with the new NAP==
Associate a SIP Domain with the new NAP. If you have more then 1 registrar domain using the same registrar you can associate all of them with the NAP:

* Select a '''sip domain''' from the '''Available''' list
* Click '''"<<"''' to associate the SIP Domain with the NAP
[[Image:Associate_SIP_NAP.png]]

* For each domain, for example, example1.com, choose OPEN_NAP and NAP_PBX1, and for example2.com, choose OPEN_NAP and NAP_PBX2, and so on.

=Call Route=

You must set up call routing for your system. [[Call routing]] refers to the ability to route calls based on criteria such as origin, destination, time of day, service provider rates, and more.

==Route Configuration for Remote Phones to SIP Server==

1- Click '''Routes''' in the navigation panel

[[Image:CreateCallRoute_0.png]]

2- Click '''Create New Static Route'''

[[Image:CreateCallRoute_1.png]]

3- Create the new route:

* Enter a '''RoutesetName''' for the route
* Select '''SIP_REMOTE''', to match calls from Trunk NAP
* Select '''SIP_SERVER_NAP'''
* Click '''Create'''

[[Image:Route_Remote_to_Server.png]]

4- Verify that the '''" Route was successfully created"''' message appears and that the new route is listed in the Routeset list

[[Image:Route_Remote_to_Server_1.png]]

==Route Configuration for SIP Server to Remote Phones==

1- Click '''Routes''' in the navigation panel

[[Image:CreateCallRoute_0.png]]

2- Click '''Create New Static Route'''

[[Image:CreateCallRoute_1.png]]

3- Create the new route:

* Enter a '''RoutesetName''' for the route
* Select '''SIP_SERVER_NAP''', to match calls from Trunk NAP
* Select ''' (By registered user) '''
* Click '''Create'''

[[Image:Route_Server_to_Remote_2.png]]

4- Verify that the '''" Route was successfully created"''' message appears and that the new route is listed in the Routeset list

[[Image:Route_Server_to_Remote_3.png]]

=Activating the Configuration=

Changes made to the configuration of the FreeSBC units are stored in the OAM&P Configuration and Logging database. In order for changes to be used by the system, they must first be activated. This is done at the system level and accessed from the Navigation panel.

Check the following link for activating the configuration;

[[Toolpack:Activating_the_Configuration_D]]

=SBC Use Cases=
[[Toolpack:Tsbc Use Cases A|SBC Use Cases]] 
[[FreeSBC_Use_Cases|FreeSBC Use Cases]]

=SBC Tutorial Guide V3.0=
[[Tmedia Tsig Tdev Tutorial Guide v3.0|Version 3.0]]

[[Category:Revise on Major]]

FreeSBC: Multiple Domains/Hosted PBXs Configuration

2020-10-21T07:52:56Z

William Wong: /* Create New SIP Registration Registrar */

=== '''''Applies to version: v3.0, 3.1''''' ===

{{DISPLAYTITLE:FreeSBC:Multiple Domains/Hosted PBXs Configuration:Example}}
=Introduction=
FreeSBC Hosted PBX Example (see [[FreeSBC:Hosted PBX:Configuration A|FreeSBC:Hosted PBX]]) provides a step by step Hosted PBX Configuration of [[FreeSBC|FreeSBC]] systems, using the Web Portal configuration tool. What will be discussed here, with guide for example configuration, is when there are multiple domains per PBX or several PBXs with different domains. This multiple domains/PBXs configuration is basically the extension of what is configured in single domain/hosted PBX, that including the setting of different domains, followed by specific routing for different domain destinations.

=Hosted PBXs Example=

[[Image:Hosted_Multiple_PBX_topo_1.png]] 

From above, there will be several Hosted PBXs like Hosted PBX1, Hosted PBX2, Hosted PBX3, etc. And each Hosted PBX will use its own domain like example1.com for PBX1, example2.com for PBX2, and so on. This multiple domain scenario can also be applied to single Hosted PBX with different several domains that will be using the similar configuration.

With external path over the WAN, different phones belonging to different Hosted PBXs register to domain corresponding to the PBX/Registrar the phone is subscribing to.

Prerequisites and below configuration sections need to be done as listed in [[FreeSBC:Hosted PBX:Configuration A|single Hosted PBX configuration]] before proceeding:

* IP Network Configuration
* SIP Stack Configuration

=SIP NAP=

A Network Access Point or NAP represents the entry point to another network or destination peer (e.g. SIP proxy, SIP trunk, etc)

==SIP NAP Configuration for Local Area Network==

To create a new NAP:

1- Click '''NAPs''' in the navigation panel

[[Image:NAP_NavigationMenu.png]]

2- Click '''Create New NAP'''

[[Image:Create_New_NAP_SIP.png]]

3- Create the new NAP:

* Enter a '''name''' for the NAP
* Click '''Create'''

[[Image:SIP_Server_NAP_Create.png]]

4- Verify that the '''NAP was successfully created''' message appears

[[Image:Create_New_NAP_SIP2.png]]

5- Associate a SIP transport server with the new NAP:

* Select a '''SIP Transport Server''' from the '''Available''' list
* Click '''"<<"''' to associate the '''LAN_SIP_TS''' with the NAP

[[Image:SIP_Server_NAP_Create_1.png]]

6- Enter SIP Server proxy address:

[[Image:SIP_Server_NAP_Create_2.png]]

7- Associate a Port range with the new NAP:

* Select a '''port range''' from the '''Available''' list
* Click '''"<<"''' to associate '''LAN_Vlan:0''' Port range with the NAP

[[Image:SIP_Server_NAP_Create_3.png]]

9- Configure settings for the following parameter groups as required:
*Registration Parameters
*Authentication Parameters
*Network Address Translation
*SIP-I Parameters
*Advanced Parameters

*Click '''Save'''

[[Image:SIP_Server_NAP_Create_4.png]]

* Here, we need to repeat above steps and create NAP for every Hosted PBX, for example, NAP_PBX1 for PBX1 (192.168.1.10), NAP_PBX2 for PBX2 (192.168.1.11), and NAP_PBX3 for PBX3 (192.168.1.12) and so on if more.

==Access Control List==

FreeSBC will automatically create Access Control List for each NAP you created.

[[Image:Remote_Workers_Access_Control_List_1.png]]

If you double-click one of the created ACL, you will see FreeSBC only accept the calls if source IP matches. In this sample; the calls from 192.168.1.10, 192.168.1.11, 192.168.1.12 will be accepted only.

[[Image:SIP_Access_Control_List_2.png]]

Rules for NAP_PBX1, NAP_PBX2, NAP_PBX3,.. will be created automatically in the ACL.

=SIP DOMAIN=

A SIP domain represents a grouping of devices (or users) that can communicate with one another.
You must configure SIP Registration Domain for your system. The first step in doing so is to create a SIP Domain:

==Create New SIP Registration Domain==

1- Click '''SIP Domain''' in the navigation panel

[[Image:SIP_Domain_NavigationMenu.png]]

2- Click '''Create New Domain'''

[[Image:Create_SIP_Domain.png]]

3- Create the new Domain:

* Enter a configuration '''Name''' for this domain.
* Enter a '''Domain Name''' for the SIP Registration Domain (the domain can be an FQDN or an IP address)
* Set the number '''Maximum Registered Users''' for this domain
* Set the Expires value used by SBC when the remote device doesn't supply one ('''Default Contact Expire''')
* Select '''Routing Method''' the system will use to route calls to registered users (if enabled in routing scripts).
** '''Register source''': Sends SIP Invite to the registering source IP address.
** '''Contact''': Sends SIP Invite to the 'contact' from the Register message.
* Set the '''Default Contact Expiration''', this value will be used when no ''Expires'' value is supplied by the user agent.
* Set the '''Minimum Contact Expiration''', this is the minimum ''Expires'' value that can be supplied by a user agent. Lower values will be rejected with a 423 'Interval too brief' response.
* Set the '''Maximum Contact Expiration''', this is the maximum ''Expires'' value that can be supplied by a user agent. The higher value is replaced by this parameter.
* Forwarding Parameters:
** Select the Registration '''Forwarding Mode''' to the registrar:
*** '''Contact Remapping''': Changes the user and the IP address.
*** '''Contact Passthrough''': Doesn't change anything. Enables devices to be contacted directly without going through the SBC.
** Set '''Minimum Registrar Expiration''', this is the minimum ''Expires'' value sent by the SBC to the registrar. If a user agent 'Expires' value is greater than this parameter, the SBC will do rate adaptation between the user agent and the Registrar.
** Set '''Maximum Pending Register Forward''', the maximum number of simultaneous pending register requests allowed for this domain. New REGISTER request is being refused passed this threshold.
** Set '''Maximum Simultaneous Register Forward''', the maximum number of simultaneous active register requests allowed for this domain. New REGISTER request is being refused passed this threshold.
* Click '''Create'''

[[Image:register_domain_1.png]]

4- Verify that the '''Sip registration domain was successfully created''' message appears

[[Image:register_domain_2.png]]
 

5- If your registrar has multiple domains, you need to create all the domains one by one.

6- Repeat above steps for every domain designated to every PBX, for example, example1.com for PBX1, example2.com for PBX2, and so on. This also applies to the case of multiple domains from the same PBX/Registrar as pointed out in step 5 above.

==Create New SIP Registration Registrar==

A SIP registrar represents a SIP endpoint that provides a location service.
You must configure SIP Registrar for your system. The first step in doing so is to select your SIP Domain:

1- Click on your domain in the '''SIP Domain List'''

[[Image:SIP_Domain_Select.png]]

2- Click '''Create New SIP Registration Registrar'''

[[Image:New_SIP_Registrar_Create.png]]

3- Create the new SIP Registration Registrar

* Enter a '''Name''' for the SIP Registration Registrar
* Select pre defined SIP Proxy NAP from drop-down menu
* Click '''Create'''

[[Image:SIP_Registrar_Select.png]]

4- Verify that the '''Sip registration registrar was successfully created''' message appears

[[Image:Create_New_SIP_Registrar2.png]]

5- Different Registrars need to be created, so do the above steps for SIP_Registrar_1 for example1.com and choose NAP_PBX1 (192.168.1.10), SIP_Registrar_2 for example2.com and choose NAP_PBX2 (192.168.1.11), and SIP_Registrar_3 for example3.com and choose NAP_PBX3 (192.168.1.12).

==Associate a SIP Domain with the new NAP==
Associate a SIP Domain with the new NAP. If you have more then 1 registrar domain using the same registrar you can associate all of them with the NAP:

* Select a '''sip domain''' from the '''Available''' list
* Click '''"<<"''' to associate the SIP Domain with the NAP
[[Image:Associate_SIP_NAP.png]]

=Call Route=

You must set up call routing for your system. [[Call routing]] refers to the ability to route calls based on criteria such as origin, destination, time of day, service provider rates, and more.

==Route Configuration for Remote Phones to SIP Server==

1- Click '''Routes''' in the navigation panel

[[Image:CreateCallRoute_0.png]]

2- Click '''Create New Static Route'''

[[Image:CreateCallRoute_1.png]]

3- Create the new route:

* Enter a '''RoutesetName''' for the route
* Select '''SIP_REMOTE''', to match calls from Trunk NAP
* Select '''SIP_SERVER_NAP'''
* Click '''Create'''

[[Image:Route_Remote_to_Server.png]]

4- Verify that the '''" Route was successfully created"''' message appears and that the new route is listed in the Routeset list

[[Image:Route_Remote_to_Server_1.png]]

==Route Configuration for SIP Server to Remote Phones==

1- Click '''Routes''' in the navigation panel

[[Image:CreateCallRoute_0.png]]

2- Click '''Create New Static Route'''

[[Image:CreateCallRoute_1.png]]

3- Create the new route:

* Enter a '''RoutesetName''' for the route
* Select '''SIP_SERVER_NAP''', to match calls from Trunk NAP
* Select ''' (By registered user) '''
* Click '''Create'''

[[Image:Route_Server_to_Remote_2.png]]

4- Verify that the '''" Route was successfully created"''' message appears and that the new route is listed in the Routeset list

[[Image:Route_Server_to_Remote_3.png]]

=Activating the Configuration=

Changes made to the configuration of the FreeSBC units are stored in the OAM&P Configuration and Logging database. In order for changes to be used by the system, they must first be activated. This is done at the system level and accessed from the Navigation panel.

Check the following link for activating the configuration;

[[Toolpack:Activating_the_Configuration_D]]

=SBC Use Cases=
[[Toolpack:Tsbc Use Cases A|SBC Use Cases]] 
[[FreeSBC_Use_Cases|FreeSBC Use Cases]]

=SBC Tutorial Guide V3.0=
[[Tmedia Tsig Tdev Tutorial Guide v3.0|Version 3.0]]

[[Category:Revise on Major]]

FreeSBC: Multiple Domains/Hosted PBXs Configuration

2020-10-21T07:49:25Z

William Wong: /* Create New SIP Registration Registrar */

=== '''''Applies to version: v3.0, 3.1''''' ===

{{DISPLAYTITLE:FreeSBC:Multiple Domains/Hosted PBXs Configuration:Example}}
=Introduction=
FreeSBC Hosted PBX Example (see [[FreeSBC:Hosted PBX:Configuration A|FreeSBC:Hosted PBX]]) provides a step by step Hosted PBX Configuration of [[FreeSBC|FreeSBC]] systems, using the Web Portal configuration tool. What will be discussed here, with guide for example configuration, is when there are multiple domains per PBX or several PBXs with different domains. This multiple domains/PBXs configuration is basically the extension of what is configured in single domain/hosted PBX, that including the setting of different domains, followed by specific routing for different domain destinations.

=Hosted PBXs Example=

[[Image:Hosted_Multiple_PBX_topo_1.png]] 

From above, there will be several Hosted PBXs like Hosted PBX1, Hosted PBX2, Hosted PBX3, etc. And each Hosted PBX will use its own domain like example1.com for PBX1, example2.com for PBX2, and so on. This multiple domain scenario can also be applied to single Hosted PBX with different several domains that will be using the similar configuration.

With external path over the WAN, different phones belonging to different Hosted PBXs register to domain corresponding to the PBX/Registrar the phone is subscribing to.

Prerequisites and below configuration sections need to be done as listed in [[FreeSBC:Hosted PBX:Configuration A|single Hosted PBX configuration]] before proceeding:

* IP Network Configuration
* SIP Stack Configuration

=SIP NAP=

A Network Access Point or NAP represents the entry point to another network or destination peer (e.g. SIP proxy, SIP trunk, etc)

==SIP NAP Configuration for Local Area Network==

To create a new NAP:

1- Click '''NAPs''' in the navigation panel

[[Image:NAP_NavigationMenu.png]]

2- Click '''Create New NAP'''

[[Image:Create_New_NAP_SIP.png]]

3- Create the new NAP:

* Enter a '''name''' for the NAP
* Click '''Create'''

[[Image:SIP_Server_NAP_Create.png]]

4- Verify that the '''NAP was successfully created''' message appears

[[Image:Create_New_NAP_SIP2.png]]

5- Associate a SIP transport server with the new NAP:

* Select a '''SIP Transport Server''' from the '''Available''' list
* Click '''"<<"''' to associate the '''LAN_SIP_TS''' with the NAP

[[Image:SIP_Server_NAP_Create_1.png]]

6- Enter SIP Server proxy address:

[[Image:SIP_Server_NAP_Create_2.png]]

7- Associate a Port range with the new NAP:

* Select a '''port range''' from the '''Available''' list
* Click '''"<<"''' to associate '''LAN_Vlan:0''' Port range with the NAP

[[Image:SIP_Server_NAP_Create_3.png]]

9- Configure settings for the following parameter groups as required:
*Registration Parameters
*Authentication Parameters
*Network Address Translation
*SIP-I Parameters
*Advanced Parameters

*Click '''Save'''

[[Image:SIP_Server_NAP_Create_4.png]]

* Here, we need to repeat above steps and create NAP for every Hosted PBX, for example, NAP_PBX1 for PBX1 (192.168.1.10), NAP_PBX2 for PBX2 (192.168.1.11), and NAP_PBX3 for PBX3 (192.168.1.12) and so on if more.

==Access Control List==

FreeSBC will automatically create Access Control List for each NAP you created.

[[Image:Remote_Workers_Access_Control_List_1.png]]

If you double-click one of the created ACL, you will see FreeSBC only accept the calls if source IP matches. In this sample; the calls from 192.168.1.10, 192.168.1.11, 192.168.1.12 will be accepted only.

[[Image:SIP_Access_Control_List_2.png]]

Rules for NAP_PBX1, NAP_PBX2, NAP_PBX3,.. will be created automatically in the ACL.

=SIP DOMAIN=

A SIP domain represents a grouping of devices (or users) that can communicate with one another.
You must configure SIP Registration Domain for your system. The first step in doing so is to create a SIP Domain:

==Create New SIP Registration Domain==

1- Click '''SIP Domain''' in the navigation panel

[[Image:SIP_Domain_NavigationMenu.png]]

2- Click '''Create New Domain'''

[[Image:Create_SIP_Domain.png]]

3- Create the new Domain:

* Enter a configuration '''Name''' for this domain.
* Enter a '''Domain Name''' for the SIP Registration Domain (the domain can be an FQDN or an IP address)
* Set the number '''Maximum Registered Users''' for this domain
* Set the Expires value used by SBC when the remote device doesn't supply one ('''Default Contact Expire''')
* Select '''Routing Method''' the system will use to route calls to registered users (if enabled in routing scripts).
** '''Register source''': Sends SIP Invite to the registering source IP address.
** '''Contact''': Sends SIP Invite to the 'contact' from the Register message.
* Set the '''Default Contact Expiration''', this value will be used when no ''Expires'' value is supplied by the user agent.
* Set the '''Minimum Contact Expiration''', this is the minimum ''Expires'' value that can be supplied by a user agent. Lower values will be rejected with a 423 'Interval too brief' response.
* Set the '''Maximum Contact Expiration''', this is the maximum ''Expires'' value that can be supplied by a user agent. The higher value is replaced by this parameter.
* Forwarding Parameters:
** Select the Registration '''Forwarding Mode''' to the registrar:
*** '''Contact Remapping''': Changes the user and the IP address.
*** '''Contact Passthrough''': Doesn't change anything. Enables devices to be contacted directly without going through the SBC.
** Set '''Minimum Registrar Expiration''', this is the minimum ''Expires'' value sent by the SBC to the registrar. If a user agent 'Expires' value is greater than this parameter, the SBC will do rate adaptation between the user agent and the Registrar.
** Set '''Maximum Pending Register Forward''', the maximum number of simultaneous pending register requests allowed for this domain. New REGISTER request is being refused passed this threshold.
** Set '''Maximum Simultaneous Register Forward''', the maximum number of simultaneous active register requests allowed for this domain. New REGISTER request is being refused passed this threshold.
* Click '''Create'''

[[Image:register_domain_1.png]]

4- Verify that the '''Sip registration domain was successfully created''' message appears

[[Image:register_domain_2.png]]
 

5- If your registrar has multiple domains, you need to create all the domains one by one.

6- Repeat above steps for every domain designated to every PBX, for example, example1.com for PBX1, example2.com for PBX2, and so on. This also applies to the case of multiple domains from the same PBX/Registrar as pointed out in step 5 above.

==Create New SIP Registration Registrar==

A SIP registrar represents a SIP endpoint that provides a location service.
You must configure SIP Registrar for your system. The first step in doing so is to select your SIP Domain:

1- Click on your domain in the '''SIP Domain List'''

[[Image:SIP_Domain_Select.png]]

2- Click '''Create New SIP Registration Registrar'''

[[Image:New_SIP_Registrar_Create.png]]

3- Create the new SIP Registration Registrar

* Enter a '''Name''' for the SIP Registration Registrar
* Select pre defined SIP Proxy NAP from drop-down menu
* Click '''Create'''

[[Image:SIP_Registrar_Select.png]]

4- Verify that the '''Sip registration registrar was successfully created''' message appears

[[Image:Create_New_SIP_Registrar2.png]]

5- Here, different Registrars need to be created, so repeat the above steps for SIP_Registrar_1 for example1.com and choose NAP_PBX1 (192.168.1.10), SIP_Registrar_2 for example2.com and choose NAP_PBX2 (192.168.1.11), and SIP_Registrar_3 for example3.com and choose NAP_PBX3 (192.168.1.12).

==Associate a SIP Domain with the new NAP==
Associate a SIP Domain with the new NAP. If you have more then 1 registrar domain using the same registrar you can associate all of them with the NAP:

* Select a '''sip domain''' from the '''Available''' list
* Click '''"<<"''' to associate the SIP Domain with the NAP
[[Image:Associate_SIP_NAP.png]]

=Call Route=

You must set up call routing for your system. [[Call routing]] refers to the ability to route calls based on criteria such as origin, destination, time of day, service provider rates, and more.

==Route Configuration for Remote Phones to SIP Server==

1- Click '''Routes''' in the navigation panel

[[Image:CreateCallRoute_0.png]]

2- Click '''Create New Static Route'''

[[Image:CreateCallRoute_1.png]]

3- Create the new route:

* Enter a '''RoutesetName''' for the route
* Select '''SIP_REMOTE''', to match calls from Trunk NAP
* Select '''SIP_SERVER_NAP'''
* Click '''Create'''

[[Image:Route_Remote_to_Server.png]]

4- Verify that the '''" Route was successfully created"''' message appears and that the new route is listed in the Routeset list

[[Image:Route_Remote_to_Server_1.png]]

==Route Configuration for SIP Server to Remote Phones==

1- Click '''Routes''' in the navigation panel

[[Image:CreateCallRoute_0.png]]

2- Click '''Create New Static Route'''

[[Image:CreateCallRoute_1.png]]

3- Create the new route:

* Enter a '''RoutesetName''' for the route
* Select '''SIP_SERVER_NAP''', to match calls from Trunk NAP
* Select ''' (By registered user) '''
* Click '''Create'''

[[Image:Route_Server_to_Remote_2.png]]

4- Verify that the '''" Route was successfully created"''' message appears and that the new route is listed in the Routeset list

[[Image:Route_Server_to_Remote_3.png]]

=Activating the Configuration=

Changes made to the configuration of the FreeSBC units are stored in the OAM&P Configuration and Logging database. In order for changes to be used by the system, they must first be activated. This is done at the system level and accessed from the Navigation panel.

Check the following link for activating the configuration;

[[Toolpack:Activating_the_Configuration_D]]

=SBC Use Cases=
[[Toolpack:Tsbc Use Cases A|SBC Use Cases]] 
[[FreeSBC_Use_Cases|FreeSBC Use Cases]]

=SBC Tutorial Guide V3.0=
[[Tmedia Tsig Tdev Tutorial Guide v3.0|Version 3.0]]

[[Category:Revise on Major]]

FreeSBC: Multiple Domains/Hosted PBXs Configuration

2020-10-21T07:46:34Z

William Wong: /* Create New SIP Registration Registrar */

=== '''''Applies to version: v3.0, 3.1''''' ===

{{DISPLAYTITLE:FreeSBC:Multiple Domains/Hosted PBXs Configuration:Example}}
=Introduction=
FreeSBC Hosted PBX Example (see [[FreeSBC:Hosted PBX:Configuration A|FreeSBC:Hosted PBX]]) provides a step by step Hosted PBX Configuration of [[FreeSBC|FreeSBC]] systems, using the Web Portal configuration tool. What will be discussed here, with guide for example configuration, is when there are multiple domains per PBX or several PBXs with different domains. This multiple domains/PBXs configuration is basically the extension of what is configured in single domain/hosted PBX, that including the setting of different domains, followed by specific routing for different domain destinations.

=Hosted PBXs Example=

[[Image:Hosted_Multiple_PBX_topo_1.png]] 

From above, there will be several Hosted PBXs like Hosted PBX1, Hosted PBX2, Hosted PBX3, etc. And each Hosted PBX will use its own domain like example1.com for PBX1, example2.com for PBX2, and so on. This multiple domain scenario can also be applied to single Hosted PBX with different several domains that will be using the similar configuration.

With external path over the WAN, different phones belonging to different Hosted PBXs register to domain corresponding to the PBX/Registrar the phone is subscribing to.

Prerequisites and below configuration sections need to be done as listed in [[FreeSBC:Hosted PBX:Configuration A|single Hosted PBX configuration]] before proceeding:

* IP Network Configuration
* SIP Stack Configuration

=SIP NAP=

A Network Access Point or NAP represents the entry point to another network or destination peer (e.g. SIP proxy, SIP trunk, etc)

==SIP NAP Configuration for Local Area Network==

To create a new NAP:

1- Click '''NAPs''' in the navigation panel

[[Image:NAP_NavigationMenu.png]]

2- Click '''Create New NAP'''

[[Image:Create_New_NAP_SIP.png]]

3- Create the new NAP:

* Enter a '''name''' for the NAP
* Click '''Create'''

[[Image:SIP_Server_NAP_Create.png]]

4- Verify that the '''NAP was successfully created''' message appears

[[Image:Create_New_NAP_SIP2.png]]

5- Associate a SIP transport server with the new NAP:

* Select a '''SIP Transport Server''' from the '''Available''' list
* Click '''"<<"''' to associate the '''LAN_SIP_TS''' with the NAP

[[Image:SIP_Server_NAP_Create_1.png]]

6- Enter SIP Server proxy address:

[[Image:SIP_Server_NAP_Create_2.png]]

7- Associate a Port range with the new NAP:

* Select a '''port range''' from the '''Available''' list
* Click '''"<<"''' to associate '''LAN_Vlan:0''' Port range with the NAP

[[Image:SIP_Server_NAP_Create_3.png]]

9- Configure settings for the following parameter groups as required:
*Registration Parameters
*Authentication Parameters
*Network Address Translation
*SIP-I Parameters
*Advanced Parameters

*Click '''Save'''

[[Image:SIP_Server_NAP_Create_4.png]]

* Here, we need to repeat above steps and create NAP for every Hosted PBX, for example, NAP_PBX1 for PBX1 (192.168.1.10), NAP_PBX2 for PBX2 (192.168.1.11), and NAP_PBX3 for PBX3 (192.168.1.12) and so on if more.

==Access Control List==

FreeSBC will automatically create Access Control List for each NAP you created.

[[Image:Remote_Workers_Access_Control_List_1.png]]

If you double-click one of the created ACL, you will see FreeSBC only accept the calls if source IP matches. In this sample; the calls from 192.168.1.10, 192.168.1.11, 192.168.1.12 will be accepted only.

[[Image:SIP_Access_Control_List_2.png]]

Rules for NAP_PBX1, NAP_PBX2, NAP_PBX3,.. will be created automatically in the ACL.

=SIP DOMAIN=

A SIP domain represents a grouping of devices (or users) that can communicate with one another.
You must configure SIP Registration Domain for your system. The first step in doing so is to create a SIP Domain:

==Create New SIP Registration Domain==

1- Click '''SIP Domain''' in the navigation panel

[[Image:SIP_Domain_NavigationMenu.png]]

2- Click '''Create New Domain'''

[[Image:Create_SIP_Domain.png]]

3- Create the new Domain:

* Enter a configuration '''Name''' for this domain.
* Enter a '''Domain Name''' for the SIP Registration Domain (the domain can be an FQDN or an IP address)
* Set the number '''Maximum Registered Users''' for this domain
* Set the Expires value used by SBC when the remote device doesn't supply one ('''Default Contact Expire''')
* Select '''Routing Method''' the system will use to route calls to registered users (if enabled in routing scripts).
** '''Register source''': Sends SIP Invite to the registering source IP address.
** '''Contact''': Sends SIP Invite to the 'contact' from the Register message.
* Set the '''Default Contact Expiration''', this value will be used when no ''Expires'' value is supplied by the user agent.
* Set the '''Minimum Contact Expiration''', this is the minimum ''Expires'' value that can be supplied by a user agent. Lower values will be rejected with a 423 'Interval too brief' response.
* Set the '''Maximum Contact Expiration''', this is the maximum ''Expires'' value that can be supplied by a user agent. The higher value is replaced by this parameter.
* Forwarding Parameters:
** Select the Registration '''Forwarding Mode''' to the registrar:
*** '''Contact Remapping''': Changes the user and the IP address.
*** '''Contact Passthrough''': Doesn't change anything. Enables devices to be contacted directly without going through the SBC.
** Set '''Minimum Registrar Expiration''', this is the minimum ''Expires'' value sent by the SBC to the registrar. If a user agent 'Expires' value is greater than this parameter, the SBC will do rate adaptation between the user agent and the Registrar.
** Set '''Maximum Pending Register Forward''', the maximum number of simultaneous pending register requests allowed for this domain. New REGISTER request is being refused passed this threshold.
** Set '''Maximum Simultaneous Register Forward''', the maximum number of simultaneous active register requests allowed for this domain. New REGISTER request is being refused passed this threshold.
* Click '''Create'''

[[Image:register_domain_1.png]]

4- Verify that the '''Sip registration domain was successfully created''' message appears

[[Image:register_domain_2.png]]
 

5- If your registrar has multiple domains, you need to create all the domains one by one.

6- Repeat above steps for every domain designated to every PBX, for example, example1.com for PBX1, example2.com for PBX2, and so on. This also applies to the case of multiple domains from the same PBX/Registrar as pointed out in step 5 above.

==Create New SIP Registration Registrar==

A SIP registrar represents a SIP endpoint that provides a location service.
You must configure SIP Registrar for your system. The first step in doing so is to select your SIP Domain:

1- Click on your domain in the '''SIP Domain List'''

[[Image:SIP_Domain_Select.png]]

2- Click '''Create New SIP Registration Registrar'''

[[Image:New_SIP_Registrar_Create.png]]

3- Create the new SIP Registration Registrar

* Enter a '''Name''' for the SIP Registration Registrar
* Select pre defined SIP Proxy NAP from drop-down menu
* Click '''Create'''

[[Image:SIP_Registrar_Select.png]]

4- Verify that the '''Sip registration registrar was successfully created''' message appears

[[Image:Create_New_SIP_Registrar2.png]]

5- Here, different Registrars need to be created, so repeat the above steps for SIP_Registrar_1 for example1.com, SIP_Registrar_2 for example2.com, and SIP_Registrar_3 for example3.com.

==Associate a SIP Domain with the new NAP==
Associate a SIP Domain with the new NAP. If you have more then 1 registrar domain using the same registrar you can associate all of them with the NAP:

* Select a '''sip domain''' from the '''Available''' list
* Click '''"<<"''' to associate the SIP Domain with the NAP
[[Image:Associate_SIP_NAP.png]]

=Call Route=

You must set up call routing for your system. [[Call routing]] refers to the ability to route calls based on criteria such as origin, destination, time of day, service provider rates, and more.

==Route Configuration for Remote Phones to SIP Server==

1- Click '''Routes''' in the navigation panel

[[Image:CreateCallRoute_0.png]]

2- Click '''Create New Static Route'''

[[Image:CreateCallRoute_1.png]]

3- Create the new route:

* Enter a '''RoutesetName''' for the route
* Select '''SIP_REMOTE''', to match calls from Trunk NAP
* Select '''SIP_SERVER_NAP'''
* Click '''Create'''

[[Image:Route_Remote_to_Server.png]]

4- Verify that the '''" Route was successfully created"''' message appears and that the new route is listed in the Routeset list

[[Image:Route_Remote_to_Server_1.png]]

==Route Configuration for SIP Server to Remote Phones==

1- Click '''Routes''' in the navigation panel

[[Image:CreateCallRoute_0.png]]

2- Click '''Create New Static Route'''

[[Image:CreateCallRoute_1.png]]

3- Create the new route:

* Enter a '''RoutesetName''' for the route
* Select '''SIP_SERVER_NAP''', to match calls from Trunk NAP
* Select ''' (By registered user) '''
* Click '''Create'''

[[Image:Route_Server_to_Remote_2.png]]

4- Verify that the '''" Route was successfully created"''' message appears and that the new route is listed in the Routeset list

[[Image:Route_Server_to_Remote_3.png]]

=Activating the Configuration=

Changes made to the configuration of the FreeSBC units are stored in the OAM&P Configuration and Logging database. In order for changes to be used by the system, they must first be activated. This is done at the system level and accessed from the Navigation panel.

Check the following link for activating the configuration;

[[Toolpack:Activating_the_Configuration_D]]

=SBC Use Cases=
[[Toolpack:Tsbc Use Cases A|SBC Use Cases]] 
[[FreeSBC_Use_Cases|FreeSBC Use Cases]]

=SBC Tutorial Guide V3.0=
[[Tmedia Tsig Tdev Tutorial Guide v3.0|Version 3.0]]

[[Category:Revise on Major]]

FreeSBC: Multiple Domains/Hosted PBXs Configuration

2020-10-21T07:44:12Z

William Wong: /* Create New SIP Registration Registrar */

=== '''''Applies to version: v3.0, 3.1''''' ===

{{DISPLAYTITLE:FreeSBC:Multiple Domains/Hosted PBXs Configuration:Example}}
=Introduction=
FreeSBC Hosted PBX Example (see [[FreeSBC:Hosted PBX:Configuration A|FreeSBC:Hosted PBX]]) provides a step by step Hosted PBX Configuration of [[FreeSBC|FreeSBC]] systems, using the Web Portal configuration tool. What will be discussed here, with guide for example configuration, is when there are multiple domains per PBX or several PBXs with different domains. This multiple domains/PBXs configuration is basically the extension of what is configured in single domain/hosted PBX, that including the setting of different domains, followed by specific routing for different domain destinations.

=Hosted PBXs Example=

[[Image:Hosted_Multiple_PBX_topo_1.png]] 

From above, there will be several Hosted PBXs like Hosted PBX1, Hosted PBX2, Hosted PBX3, etc. And each Hosted PBX will use its own domain like example1.com for PBX1, example2.com for PBX2, and so on. This multiple domain scenario can also be applied to single Hosted PBX with different several domains that will be using the similar configuration.

With external path over the WAN, different phones belonging to different Hosted PBXs register to domain corresponding to the PBX/Registrar the phone is subscribing to.

Prerequisites and below configuration sections need to be done as listed in [[FreeSBC:Hosted PBX:Configuration A|single Hosted PBX configuration]] before proceeding:

* IP Network Configuration
* SIP Stack Configuration

=SIP NAP=

A Network Access Point or NAP represents the entry point to another network or destination peer (e.g. SIP proxy, SIP trunk, etc)

==SIP NAP Configuration for Local Area Network==

To create a new NAP:

1- Click '''NAPs''' in the navigation panel

[[Image:NAP_NavigationMenu.png]]

2- Click '''Create New NAP'''

[[Image:Create_New_NAP_SIP.png]]

3- Create the new NAP:

* Enter a '''name''' for the NAP
* Click '''Create'''

[[Image:SIP_Server_NAP_Create.png]]

4- Verify that the '''NAP was successfully created''' message appears

[[Image:Create_New_NAP_SIP2.png]]

5- Associate a SIP transport server with the new NAP:

* Select a '''SIP Transport Server''' from the '''Available''' list
* Click '''"<<"''' to associate the '''LAN_SIP_TS''' with the NAP

[[Image:SIP_Server_NAP_Create_1.png]]

6- Enter SIP Server proxy address:

[[Image:SIP_Server_NAP_Create_2.png]]

7- Associate a Port range with the new NAP:

* Select a '''port range''' from the '''Available''' list
* Click '''"<<"''' to associate '''LAN_Vlan:0''' Port range with the NAP

[[Image:SIP_Server_NAP_Create_3.png]]

9- Configure settings for the following parameter groups as required:
*Registration Parameters
*Authentication Parameters
*Network Address Translation
*SIP-I Parameters
*Advanced Parameters

*Click '''Save'''

[[Image:SIP_Server_NAP_Create_4.png]]

* Here, we need to repeat above steps and create NAP for every Hosted PBX, for example, NAP_PBX1 for PBX1 (192.168.1.10), NAP_PBX2 for PBX2 (192.168.1.11), and NAP_PBX3 for PBX3 (192.168.1.12) and so on if more.

==Access Control List==

FreeSBC will automatically create Access Control List for each NAP you created.

[[Image:Remote_Workers_Access_Control_List_1.png]]

If you double-click one of the created ACL, you will see FreeSBC only accept the calls if source IP matches. In this sample; the calls from 192.168.1.10, 192.168.1.11, 192.168.1.12 will be accepted only.

[[Image:SIP_Access_Control_List_2.png]]

Rules for NAP_PBX1, NAP_PBX2, NAP_PBX3,.. will be created automatically in the ACL.

=SIP DOMAIN=

A SIP domain represents a grouping of devices (or users) that can communicate with one another.
You must configure SIP Registration Domain for your system. The first step in doing so is to create a SIP Domain:

==Create New SIP Registration Domain==

1- Click '''SIP Domain''' in the navigation panel

[[Image:SIP_Domain_NavigationMenu.png]]

2- Click '''Create New Domain'''

[[Image:Create_SIP_Domain.png]]

3- Create the new Domain:

* Enter a configuration '''Name''' for this domain.
* Enter a '''Domain Name''' for the SIP Registration Domain (the domain can be an FQDN or an IP address)
* Set the number '''Maximum Registered Users''' for this domain
* Set the Expires value used by SBC when the remote device doesn't supply one ('''Default Contact Expire''')
* Select '''Routing Method''' the system will use to route calls to registered users (if enabled in routing scripts).
** '''Register source''': Sends SIP Invite to the registering source IP address.
** '''Contact''': Sends SIP Invite to the 'contact' from the Register message.
* Set the '''Default Contact Expiration''', this value will be used when no ''Expires'' value is supplied by the user agent.
* Set the '''Minimum Contact Expiration''', this is the minimum ''Expires'' value that can be supplied by a user agent. Lower values will be rejected with a 423 'Interval too brief' response.
* Set the '''Maximum Contact Expiration''', this is the maximum ''Expires'' value that can be supplied by a user agent. The higher value is replaced by this parameter.
* Forwarding Parameters:
** Select the Registration '''Forwarding Mode''' to the registrar:
*** '''Contact Remapping''': Changes the user and the IP address.
*** '''Contact Passthrough''': Doesn't change anything. Enables devices to be contacted directly without going through the SBC.
** Set '''Minimum Registrar Expiration''', this is the minimum ''Expires'' value sent by the SBC to the registrar. If a user agent 'Expires' value is greater than this parameter, the SBC will do rate adaptation between the user agent and the Registrar.
** Set '''Maximum Pending Register Forward''', the maximum number of simultaneous pending register requests allowed for this domain. New REGISTER request is being refused passed this threshold.
** Set '''Maximum Simultaneous Register Forward''', the maximum number of simultaneous active register requests allowed for this domain. New REGISTER request is being refused passed this threshold.
* Click '''Create'''

[[Image:register_domain_1.png]]

4- Verify that the '''Sip registration domain was successfully created''' message appears

[[Image:register_domain_2.png]]
 

5- If your registrar has multiple domains, you need to create all the domains one by one.

6- Repeat above steps for every domain designated to every PBX, for example, example1.com for PBX1, example2.com for PBX2, and so on. This also applies to the case of multiple domains from the same PBX/Registrar as pointed out in step 5 above.

==Create New SIP Registration Registrar==

A SIP registrar represents a SIP endpoint that provides a location service.
You must configure SIP Registrar for your system. The first step in doing so is to select your SIP Domain:

1- Click on your domain in the '''SIP Domain List'''

[[Image:SIP_Domain_Select.png]]

2- Click '''Create New SIP Registration Registrar'''

[[Image:New_SIP_Registrar_Create.png]]

3- Create the new SIP Registration Registrar

* Enter a '''Name''' for the SIP Registration Registrar
* Select pre defined SIP Proxy NAP from drop-down menu
* Click '''Create'''

[[Image:SIP_Registrar_Select.png]]

4- Verify that the '''Sip registration registrar was successfully created''' message appears

[[Image:Create_New_SIP_Registrar2.png]]

5- Here, different Registrars need to be created. These represent Registrar-1 for 192.168.1.10, Registrar-2 for 192.168.1.11, and Registrar-3 for 192.168.1.12

==Associate a SIP Domain with the new NAP==
Associate a SIP Domain with the new NAP. If you have more then 1 registrar domain using the same registrar you can associate all of them with the NAP:

* Select a '''sip domain''' from the '''Available''' list
* Click '''"<<"''' to associate the SIP Domain with the NAP
[[Image:Associate_SIP_NAP.png]]

=Call Route=

You must set up call routing for your system. [[Call routing]] refers to the ability to route calls based on criteria such as origin, destination, time of day, service provider rates, and more.

==Route Configuration for Remote Phones to SIP Server==

1- Click '''Routes''' in the navigation panel

[[Image:CreateCallRoute_0.png]]

2- Click '''Create New Static Route'''

[[Image:CreateCallRoute_1.png]]

3- Create the new route:

* Enter a '''RoutesetName''' for the route
* Select '''SIP_REMOTE''', to match calls from Trunk NAP
* Select '''SIP_SERVER_NAP'''
* Click '''Create'''

[[Image:Route_Remote_to_Server.png]]

4- Verify that the '''" Route was successfully created"''' message appears and that the new route is listed in the Routeset list

[[Image:Route_Remote_to_Server_1.png]]

==Route Configuration for SIP Server to Remote Phones==

1- Click '''Routes''' in the navigation panel

[[Image:CreateCallRoute_0.png]]

2- Click '''Create New Static Route'''

[[Image:CreateCallRoute_1.png]]

3- Create the new route:

* Enter a '''RoutesetName''' for the route
* Select '''SIP_SERVER_NAP''', to match calls from Trunk NAP
* Select ''' (By registered user) '''
* Click '''Create'''

[[Image:Route_Server_to_Remote_2.png]]

4- Verify that the '''" Route was successfully created"''' message appears and that the new route is listed in the Routeset list

[[Image:Route_Server_to_Remote_3.png]]

=Activating the Configuration=

Changes made to the configuration of the FreeSBC units are stored in the OAM&P Configuration and Logging database. In order for changes to be used by the system, they must first be activated. This is done at the system level and accessed from the Navigation panel.

Check the following link for activating the configuration;

[[Toolpack:Activating_the_Configuration_D]]

=SBC Use Cases=
[[Toolpack:Tsbc Use Cases A|SBC Use Cases]] 
[[FreeSBC_Use_Cases|FreeSBC Use Cases]]

=SBC Tutorial Guide V3.0=
[[Tmedia Tsig Tdev Tutorial Guide v3.0|Version 3.0]]

[[Category:Revise on Major]]

FreeSBC: Multiple Domains/Hosted PBXs Configuration

2020-10-21T07:39:58Z

William Wong: /* Create New SIP Registration Domain */

=== '''''Applies to version: v3.0, 3.1''''' ===

{{DISPLAYTITLE:FreeSBC:Multiple Domains/Hosted PBXs Configuration:Example}}
=Introduction=
FreeSBC Hosted PBX Example (see [[FreeSBC:Hosted PBX:Configuration A|FreeSBC:Hosted PBX]]) provides a step by step Hosted PBX Configuration of [[FreeSBC|FreeSBC]] systems, using the Web Portal configuration tool. What will be discussed here, with guide for example configuration, is when there are multiple domains per PBX or several PBXs with different domains. This multiple domains/PBXs configuration is basically the extension of what is configured in single domain/hosted PBX, that including the setting of different domains, followed by specific routing for different domain destinations.

=Hosted PBXs Example=

[[Image:Hosted_Multiple_PBX_topo_1.png]] 

From above, there will be several Hosted PBXs like Hosted PBX1, Hosted PBX2, Hosted PBX3, etc. And each Hosted PBX will use its own domain like example1.com for PBX1, example2.com for PBX2, and so on. This multiple domain scenario can also be applied to single Hosted PBX with different several domains that will be using the similar configuration.

With external path over the WAN, different phones belonging to different Hosted PBXs register to domain corresponding to the PBX/Registrar the phone is subscribing to.

Prerequisites and below configuration sections need to be done as listed in [[FreeSBC:Hosted PBX:Configuration A|single Hosted PBX configuration]] before proceeding:

* IP Network Configuration
* SIP Stack Configuration

=SIP NAP=

A Network Access Point or NAP represents the entry point to another network or destination peer (e.g. SIP proxy, SIP trunk, etc)

==SIP NAP Configuration for Local Area Network==

To create a new NAP:

1- Click '''NAPs''' in the navigation panel

[[Image:NAP_NavigationMenu.png]]

2- Click '''Create New NAP'''

[[Image:Create_New_NAP_SIP.png]]

3- Create the new NAP:

* Enter a '''name''' for the NAP
* Click '''Create'''

[[Image:SIP_Server_NAP_Create.png]]

4- Verify that the '''NAP was successfully created''' message appears

[[Image:Create_New_NAP_SIP2.png]]

5- Associate a SIP transport server with the new NAP:

* Select a '''SIP Transport Server''' from the '''Available''' list
* Click '''"<<"''' to associate the '''LAN_SIP_TS''' with the NAP

[[Image:SIP_Server_NAP_Create_1.png]]

6- Enter SIP Server proxy address:

[[Image:SIP_Server_NAP_Create_2.png]]

7- Associate a Port range with the new NAP:

* Select a '''port range''' from the '''Available''' list
* Click '''"<<"''' to associate '''LAN_Vlan:0''' Port range with the NAP

[[Image:SIP_Server_NAP_Create_3.png]]

9- Configure settings for the following parameter groups as required:
*Registration Parameters
*Authentication Parameters
*Network Address Translation
*SIP-I Parameters
*Advanced Parameters

*Click '''Save'''

[[Image:SIP_Server_NAP_Create_4.png]]

* Here, we need to repeat above steps and create NAP for every Hosted PBX, for example, NAP_PBX1 for PBX1 (192.168.1.10), NAP_PBX2 for PBX2 (192.168.1.11), and NAP_PBX3 for PBX3 (192.168.1.12) and so on if more.

==Access Control List==

FreeSBC will automatically create Access Control List for each NAP you created.

[[Image:Remote_Workers_Access_Control_List_1.png]]

If you double-click one of the created ACL, you will see FreeSBC only accept the calls if source IP matches. In this sample; the calls from 192.168.1.10, 192.168.1.11, 192.168.1.12 will be accepted only.

[[Image:SIP_Access_Control_List_2.png]]

Rules for NAP_PBX1, NAP_PBX2, NAP_PBX3,.. will be created automatically in the ACL.

=SIP DOMAIN=

A SIP domain represents a grouping of devices (or users) that can communicate with one another.
You must configure SIP Registration Domain for your system. The first step in doing so is to create a SIP Domain:

==Create New SIP Registration Domain==

1- Click '''SIP Domain''' in the navigation panel

[[Image:SIP_Domain_NavigationMenu.png]]

2- Click '''Create New Domain'''

[[Image:Create_SIP_Domain.png]]

3- Create the new Domain:

* Enter a configuration '''Name''' for this domain.
* Enter a '''Domain Name''' for the SIP Registration Domain (the domain can be an FQDN or an IP address)
* Set the number '''Maximum Registered Users''' for this domain
* Set the Expires value used by SBC when the remote device doesn't supply one ('''Default Contact Expire''')
* Select '''Routing Method''' the system will use to route calls to registered users (if enabled in routing scripts).
** '''Register source''': Sends SIP Invite to the registering source IP address.
** '''Contact''': Sends SIP Invite to the 'contact' from the Register message.
* Set the '''Default Contact Expiration''', this value will be used when no ''Expires'' value is supplied by the user agent.
* Set the '''Minimum Contact Expiration''', this is the minimum ''Expires'' value that can be supplied by a user agent. Lower values will be rejected with a 423 'Interval too brief' response.
* Set the '''Maximum Contact Expiration''', this is the maximum ''Expires'' value that can be supplied by a user agent. The higher value is replaced by this parameter.
* Forwarding Parameters:
** Select the Registration '''Forwarding Mode''' to the registrar:
*** '''Contact Remapping''': Changes the user and the IP address.
*** '''Contact Passthrough''': Doesn't change anything. Enables devices to be contacted directly without going through the SBC.
** Set '''Minimum Registrar Expiration''', this is the minimum ''Expires'' value sent by the SBC to the registrar. If a user agent 'Expires' value is greater than this parameter, the SBC will do rate adaptation between the user agent and the Registrar.
** Set '''Maximum Pending Register Forward''', the maximum number of simultaneous pending register requests allowed for this domain. New REGISTER request is being refused passed this threshold.
** Set '''Maximum Simultaneous Register Forward''', the maximum number of simultaneous active register requests allowed for this domain. New REGISTER request is being refused passed this threshold.
* Click '''Create'''

[[Image:register_domain_1.png]]

4- Verify that the '''Sip registration domain was successfully created''' message appears

[[Image:register_domain_2.png]]
 

5- If your registrar has multiple domains, you need to create all the domains one by one.

6- Repeat above steps for every domain designated to every PBX, for example, example1.com for PBX1, example2.com for PBX2, and so on. This also applies to the case of multiple domains from the same PBX/Registrar as pointed out in step 5 above.

==Create New SIP Registration Registrar==

A SIP registrar represents a SIP endpoint that provides a location service.
You must configure SIP Registrar for your system. The first step in doing so is to select your SIP Domain:

1- Click on your domain in the '''SIP Domain List'''

[[Image:SIP_Domain_Select.png]]

2- Click '''Create New SIP Registration Registrar'''

[[Image:New_SIP_Registrar_Create.png]]

3- Create the new SIP Registration Registrar

* Enter a '''Name''' for the SIP Registration Registrar
* Select pre defined SIP Proxy NAP from drop-down menu
* Click '''Create'''

[[Image:SIP_Registrar_Select.png]]

4- Verify that the '''Sip registration registrar was successfully created''' message appears

[[Image:Create_New_SIP_Registrar2.png]]

==Associate a SIP Domain with the new NAP==
Associate a SIP Domain with the new NAP. If you have more then 1 registrar domain using the same registrar you can associate all of them with the NAP:

* Select a '''sip domain''' from the '''Available''' list
* Click '''"<<"''' to associate the SIP Domain with the NAP
[[Image:Associate_SIP_NAP.png]]

=Call Route=

You must set up call routing for your system. [[Call routing]] refers to the ability to route calls based on criteria such as origin, destination, time of day, service provider rates, and more.

==Route Configuration for Remote Phones to SIP Server==

1- Click '''Routes''' in the navigation panel

[[Image:CreateCallRoute_0.png]]

2- Click '''Create New Static Route'''

[[Image:CreateCallRoute_1.png]]

3- Create the new route:

* Enter a '''RoutesetName''' for the route
* Select '''SIP_REMOTE''', to match calls from Trunk NAP
* Select '''SIP_SERVER_NAP'''
* Click '''Create'''

[[Image:Route_Remote_to_Server.png]]

4- Verify that the '''" Route was successfully created"''' message appears and that the new route is listed in the Routeset list

[[Image:Route_Remote_to_Server_1.png]]

==Route Configuration for SIP Server to Remote Phones==

1- Click '''Routes''' in the navigation panel

[[Image:CreateCallRoute_0.png]]

2- Click '''Create New Static Route'''

[[Image:CreateCallRoute_1.png]]

3- Create the new route:

* Enter a '''RoutesetName''' for the route
* Select '''SIP_SERVER_NAP''', to match calls from Trunk NAP
* Select ''' (By registered user) '''
* Click '''Create'''

[[Image:Route_Server_to_Remote_2.png]]

4- Verify that the '''" Route was successfully created"''' message appears and that the new route is listed in the Routeset list

[[Image:Route_Server_to_Remote_3.png]]

=Activating the Configuration=

Changes made to the configuration of the FreeSBC units are stored in the OAM&P Configuration and Logging database. In order for changes to be used by the system, they must first be activated. This is done at the system level and accessed from the Navigation panel.

Check the following link for activating the configuration;

[[Toolpack:Activating_the_Configuration_D]]

=SBC Use Cases=
[[Toolpack:Tsbc Use Cases A|SBC Use Cases]] 
[[FreeSBC_Use_Cases|FreeSBC Use Cases]]

=SBC Tutorial Guide V3.0=
[[Tmedia Tsig Tdev Tutorial Guide v3.0|Version 3.0]]

[[Category:Revise on Major]]

FreeSBC: Multiple Domains/Hosted PBXs Configuration

2020-10-21T07:33:31Z

William Wong: /* Access Control List */

=== '''''Applies to version: v3.0, 3.1''''' ===

{{DISPLAYTITLE:FreeSBC:Multiple Domains/Hosted PBXs Configuration:Example}}
=Introduction=
FreeSBC Hosted PBX Example (see [[FreeSBC:Hosted PBX:Configuration A|FreeSBC:Hosted PBX]]) provides a step by step Hosted PBX Configuration of [[FreeSBC|FreeSBC]] systems, using the Web Portal configuration tool. What will be discussed here, with guide for example configuration, is when there are multiple domains per PBX or several PBXs with different domains. This multiple domains/PBXs configuration is basically the extension of what is configured in single domain/hosted PBX, that including the setting of different domains, followed by specific routing for different domain destinations.

=Hosted PBXs Example=

[[Image:Hosted_Multiple_PBX_topo_1.png]] 

From above, there will be several Hosted PBXs like Hosted PBX1, Hosted PBX2, Hosted PBX3, etc. And each Hosted PBX will use its own domain like example1.com for PBX1, example2.com for PBX2, and so on. This multiple domain scenario can also be applied to single Hosted PBX with different several domains that will be using the similar configuration.

With external path over the WAN, different phones belonging to different Hosted PBXs register to domain corresponding to the PBX/Registrar the phone is subscribing to.

Prerequisites and below configuration sections need to be done as listed in [[FreeSBC:Hosted PBX:Configuration A|single Hosted PBX configuration]] before proceeding:

* IP Network Configuration
* SIP Stack Configuration

=SIP NAP=

A Network Access Point or NAP represents the entry point to another network or destination peer (e.g. SIP proxy, SIP trunk, etc)

==SIP NAP Configuration for Local Area Network==

To create a new NAP:

1- Click '''NAPs''' in the navigation panel

[[Image:NAP_NavigationMenu.png]]

2- Click '''Create New NAP'''

[[Image:Create_New_NAP_SIP.png]]

3- Create the new NAP:

* Enter a '''name''' for the NAP
* Click '''Create'''

[[Image:SIP_Server_NAP_Create.png]]

4- Verify that the '''NAP was successfully created''' message appears

[[Image:Create_New_NAP_SIP2.png]]

5- Associate a SIP transport server with the new NAP:

* Select a '''SIP Transport Server''' from the '''Available''' list
* Click '''"<<"''' to associate the '''LAN_SIP_TS''' with the NAP

[[Image:SIP_Server_NAP_Create_1.png]]

6- Enter SIP Server proxy address:

[[Image:SIP_Server_NAP_Create_2.png]]

7- Associate a Port range with the new NAP:

* Select a '''port range''' from the '''Available''' list
* Click '''"<<"''' to associate '''LAN_Vlan:0''' Port range with the NAP

[[Image:SIP_Server_NAP_Create_3.png]]

9- Configure settings for the following parameter groups as required:
*Registration Parameters
*Authentication Parameters
*Network Address Translation
*SIP-I Parameters
*Advanced Parameters

*Click '''Save'''

[[Image:SIP_Server_NAP_Create_4.png]]

* Here, we need to repeat above steps and create NAP for every Hosted PBX, for example, NAP_PBX1 for PBX1 (192.168.1.10), NAP_PBX2 for PBX2 (192.168.1.11), and NAP_PBX3 for PBX3 (192.168.1.12) and so on if more.

==Access Control List==

FreeSBC will automatically create Access Control List for each NAP you created.

[[Image:Remote_Workers_Access_Control_List_1.png]]

If you double-click one of the created ACL, you will see FreeSBC only accept the calls if source IP matches. In this sample; the calls from 192.168.1.10, 192.168.1.11, 192.168.1.12 will be accepted only.

[[Image:SIP_Access_Control_List_2.png]]

Rules for NAP_PBX1, NAP_PBX2, NAP_PBX3,.. will be created automatically in the ACL.

=SIP DOMAIN=

A SIP domain represents a grouping of devices (or users) that can communicate with one another.
You must configure SIP Registration Domain for your system. The first step in doing so is to create a SIP Domain:

==Create New SIP Registration Domain==

1- Click '''SIP Domain''' in the navigation panel

[[Image:SIP_Domain_NavigationMenu.png]]

2- Click '''Create New Domain'''

[[Image:Create_SIP_Domain.png]]

3- Create the new Domain:

* Enter a configuration '''Name''' for this domain.
* Enter a '''Domain Name''' for the SIP Registration Domain (the domain can be an FQDN or an IP address)
* Set the number '''Maximum Registered Users''' for this domain
* Set the Expires value used by SBC when the remote device doesn't supply one ('''Default Contact Expire''')
* Select '''Routing Method''' the system will use to route calls to registered users (if enabled in routing scripts).
** '''Register source''': Sends SIP Invite to the registering source IP address.
** '''Contact''': Sends SIP Invite to the 'contact' from the Register message.
* Set the '''Default Contact Expiration''', this value will be used when no ''Expires'' value is supplied by the user agent.
* Set the '''Minimum Contact Expiration''', this is the minimum ''Expires'' value that can be supplied by a user agent. Lower values will be rejected with a 423 'Interval too brief' response.
* Set the '''Maximum Contact Expiration''', this is the maximum ''Expires'' value that can be supplied by a user agent. The higher value is replaced by this parameter.
* Forwarding Parameters:
** Select the Registration '''Forwarding Mode''' to the registrar:
*** '''Contact Remapping''': Changes the user and the IP address.
*** '''Contact Passthrough''': Doesn't change anything. Enables devices to be contacted directly without going through the SBC.
** Set '''Minimum Registrar Expiration''', this is the minimum ''Expires'' value sent by the SBC to the registrar. If a user agent 'Expires' value is greater than this parameter, the SBC will do rate adaptation between the user agent and the Registrar.
** Set '''Maximum Pending Register Forward''', the maximum number of simultaneous pending register requests allowed for this domain. New REGISTER request is being refused passed this threshold.
** Set '''Maximum Simultaneous Register Forward''', the maximum number of simultaneous active register requests allowed for this domain. New REGISTER request is being refused passed this threshold.
* Click '''Create'''

[[Image:register_domain_1.png]]

4- Verify that the '''Sip registration domain was successfully created''' message appears

[[Image:register_domain_2.png]]
 

5- If your registrar has multiple domains, you need to create all the domains one by one.

Repeat above steps for every domain designated to every PBX, for example, example1.com for PBX1, example2.com for PBX2, and so on. This also applies to the case of multiple domains from the same PBX/Registrar as pointed out in step 5 above.

==Create New SIP Registration Registrar==

A SIP registrar represents a SIP endpoint that provides a location service.
You must configure SIP Registrar for your system. The first step in doing so is to select your SIP Domain:

1- Click on your domain in the '''SIP Domain List'''

[[Image:SIP_Domain_Select.png]]

2- Click '''Create New SIP Registration Registrar'''

[[Image:New_SIP_Registrar_Create.png]]

3- Create the new SIP Registration Registrar

* Enter a '''Name''' for the SIP Registration Registrar
* Select pre defined SIP Proxy NAP from drop-down menu
* Click '''Create'''

[[Image:SIP_Registrar_Select.png]]

4- Verify that the '''Sip registration registrar was successfully created''' message appears

[[Image:Create_New_SIP_Registrar2.png]]

==Associate a SIP Domain with the new NAP==
Associate a SIP Domain with the new NAP. If you have more then 1 registrar domain using the same registrar you can associate all of them with the NAP:

* Select a '''sip domain''' from the '''Available''' list
* Click '''"<<"''' to associate the SIP Domain with the NAP
[[Image:Associate_SIP_NAP.png]]

=Call Route=

You must set up call routing for your system. [[Call routing]] refers to the ability to route calls based on criteria such as origin, destination, time of day, service provider rates, and more.

==Route Configuration for Remote Phones to SIP Server==

1- Click '''Routes''' in the navigation panel

[[Image:CreateCallRoute_0.png]]

2- Click '''Create New Static Route'''

[[Image:CreateCallRoute_1.png]]

3- Create the new route:

* Enter a '''RoutesetName''' for the route
* Select '''SIP_REMOTE''', to match calls from Trunk NAP
* Select '''SIP_SERVER_NAP'''
* Click '''Create'''

[[Image:Route_Remote_to_Server.png]]

4- Verify that the '''" Route was successfully created"''' message appears and that the new route is listed in the Routeset list

[[Image:Route_Remote_to_Server_1.png]]

==Route Configuration for SIP Server to Remote Phones==

1- Click '''Routes''' in the navigation panel

[[Image:CreateCallRoute_0.png]]

2- Click '''Create New Static Route'''

[[Image:CreateCallRoute_1.png]]

3- Create the new route:

* Enter a '''RoutesetName''' for the route
* Select '''SIP_SERVER_NAP''', to match calls from Trunk NAP
* Select ''' (By registered user) '''
* Click '''Create'''

[[Image:Route_Server_to_Remote_2.png]]

4- Verify that the '''" Route was successfully created"''' message appears and that the new route is listed in the Routeset list

[[Image:Route_Server_to_Remote_3.png]]

=Activating the Configuration=

Changes made to the configuration of the FreeSBC units are stored in the OAM&P Configuration and Logging database. In order for changes to be used by the system, they must first be activated. This is done at the system level and accessed from the Navigation panel.

Check the following link for activating the configuration;

[[Toolpack:Activating_the_Configuration_D]]

=SBC Use Cases=
[[Toolpack:Tsbc Use Cases A|SBC Use Cases]] 
[[FreeSBC_Use_Cases|FreeSBC Use Cases]]

=SBC Tutorial Guide V3.0=
[[Tmedia Tsig Tdev Tutorial Guide v3.0|Version 3.0]]

[[Category:Revise on Major]]

FreeSBC: Multiple Domains/Hosted PBXs Configuration

2020-10-21T07:32:52Z

William Wong: /* Access Control List */

=== '''''Applies to version: v3.0, 3.1''''' ===

{{DISPLAYTITLE:FreeSBC:Multiple Domains/Hosted PBXs Configuration:Example}}
=Introduction=
FreeSBC Hosted PBX Example (see [[FreeSBC:Hosted PBX:Configuration A|FreeSBC:Hosted PBX]]) provides a step by step Hosted PBX Configuration of [[FreeSBC|FreeSBC]] systems, using the Web Portal configuration tool. What will be discussed here, with guide for example configuration, is when there are multiple domains per PBX or several PBXs with different domains. This multiple domains/PBXs configuration is basically the extension of what is configured in single domain/hosted PBX, that including the setting of different domains, followed by specific routing for different domain destinations.

=Hosted PBXs Example=

[[Image:Hosted_Multiple_PBX_topo_1.png]] 

From above, there will be several Hosted PBXs like Hosted PBX1, Hosted PBX2, Hosted PBX3, etc. And each Hosted PBX will use its own domain like example1.com for PBX1, example2.com for PBX2, and so on. This multiple domain scenario can also be applied to single Hosted PBX with different several domains that will be using the similar configuration.

With external path over the WAN, different phones belonging to different Hosted PBXs register to domain corresponding to the PBX/Registrar the phone is subscribing to.

Prerequisites and below configuration sections need to be done as listed in [[FreeSBC:Hosted PBX:Configuration A|single Hosted PBX configuration]] before proceeding:

* IP Network Configuration
* SIP Stack Configuration

=SIP NAP=

A Network Access Point or NAP represents the entry point to another network or destination peer (e.g. SIP proxy, SIP trunk, etc)

==SIP NAP Configuration for Local Area Network==

To create a new NAP:

1- Click '''NAPs''' in the navigation panel

[[Image:NAP_NavigationMenu.png]]

2- Click '''Create New NAP'''

[[Image:Create_New_NAP_SIP.png]]

3- Create the new NAP:

* Enter a '''name''' for the NAP
* Click '''Create'''

[[Image:SIP_Server_NAP_Create.png]]

4- Verify that the '''NAP was successfully created''' message appears

[[Image:Create_New_NAP_SIP2.png]]

5- Associate a SIP transport server with the new NAP:

* Select a '''SIP Transport Server''' from the '''Available''' list
* Click '''"<<"''' to associate the '''LAN_SIP_TS''' with the NAP

[[Image:SIP_Server_NAP_Create_1.png]]

6- Enter SIP Server proxy address:

[[Image:SIP_Server_NAP_Create_2.png]]

7- Associate a Port range with the new NAP:

* Select a '''port range''' from the '''Available''' list
* Click '''"<<"''' to associate '''LAN_Vlan:0''' Port range with the NAP

[[Image:SIP_Server_NAP_Create_3.png]]

9- Configure settings for the following parameter groups as required:
*Registration Parameters
*Authentication Parameters
*Network Address Translation
*SIP-I Parameters
*Advanced Parameters

*Click '''Save'''

[[Image:SIP_Server_NAP_Create_4.png]]

* Here, we need to repeat above steps and create NAP for every Hosted PBX, for example, NAP_PBX1 for PBX1 (192.168.1.10), NAP_PBX2 for PBX2 (192.168.1.11), and NAP_PBX3 for PBX3 (192.168.1.12) and so on if more.

==Access Control List==

FreeSBC will automatically create Access Control List for each NAP you created.

[[Image:Remote_Workers_Access_Control_List_1.png]]

If you double-click one of the created ACL, you will see FreeSBC only accept the calls if source IP matches. In this sample; the calls from 192.168.1.10, 192.168.1.11, 192.168.1.12 will accepted only.

[[Image:SIP_Access_Control_List_2.png]]

Rules for NAP_PBX1, NAP_PBX2, NAP_PBX3,.. will be created automatically in the ACL.

=SIP DOMAIN=

A SIP domain represents a grouping of devices (or users) that can communicate with one another.
You must configure SIP Registration Domain for your system. The first step in doing so is to create a SIP Domain:

==Create New SIP Registration Domain==

1- Click '''SIP Domain''' in the navigation panel

[[Image:SIP_Domain_NavigationMenu.png]]

2- Click '''Create New Domain'''

[[Image:Create_SIP_Domain.png]]

3- Create the new Domain:

* Enter a configuration '''Name''' for this domain.
* Enter a '''Domain Name''' for the SIP Registration Domain (the domain can be an FQDN or an IP address)
* Set the number '''Maximum Registered Users''' for this domain
* Set the Expires value used by SBC when the remote device doesn't supply one ('''Default Contact Expire''')
* Select '''Routing Method''' the system will use to route calls to registered users (if enabled in routing scripts).
** '''Register source''': Sends SIP Invite to the registering source IP address.
** '''Contact''': Sends SIP Invite to the 'contact' from the Register message.
* Set the '''Default Contact Expiration''', this value will be used when no ''Expires'' value is supplied by the user agent.
* Set the '''Minimum Contact Expiration''', this is the minimum ''Expires'' value that can be supplied by a user agent. Lower values will be rejected with a 423 'Interval too brief' response.
* Set the '''Maximum Contact Expiration''', this is the maximum ''Expires'' value that can be supplied by a user agent. The higher value is replaced by this parameter.
* Forwarding Parameters:
** Select the Registration '''Forwarding Mode''' to the registrar:
*** '''Contact Remapping''': Changes the user and the IP address.
*** '''Contact Passthrough''': Doesn't change anything. Enables devices to be contacted directly without going through the SBC.
** Set '''Minimum Registrar Expiration''', this is the minimum ''Expires'' value sent by the SBC to the registrar. If a user agent 'Expires' value is greater than this parameter, the SBC will do rate adaptation between the user agent and the Registrar.
** Set '''Maximum Pending Register Forward''', the maximum number of simultaneous pending register requests allowed for this domain. New REGISTER request is being refused passed this threshold.
** Set '''Maximum Simultaneous Register Forward''', the maximum number of simultaneous active register requests allowed for this domain. New REGISTER request is being refused passed this threshold.
* Click '''Create'''

[[Image:register_domain_1.png]]

4- Verify that the '''Sip registration domain was successfully created''' message appears

[[Image:register_domain_2.png]]
 

5- If your registrar has multiple domains, you need to create all the domains one by one.

Repeat above steps for every domain designated to every PBX, for example, example1.com for PBX1, example2.com for PBX2, and so on. This also applies to the case of multiple domains from the same PBX/Registrar as pointed out in step 5 above.

==Create New SIP Registration Registrar==

A SIP registrar represents a SIP endpoint that provides a location service.
You must configure SIP Registrar for your system. The first step in doing so is to select your SIP Domain:

1- Click on your domain in the '''SIP Domain List'''

[[Image:SIP_Domain_Select.png]]

2- Click '''Create New SIP Registration Registrar'''

[[Image:New_SIP_Registrar_Create.png]]

3- Create the new SIP Registration Registrar

* Enter a '''Name''' for the SIP Registration Registrar
* Select pre defined SIP Proxy NAP from drop-down menu
* Click '''Create'''

[[Image:SIP_Registrar_Select.png]]

4- Verify that the '''Sip registration registrar was successfully created''' message appears

[[Image:Create_New_SIP_Registrar2.png]]

==Associate a SIP Domain with the new NAP==
Associate a SIP Domain with the new NAP. If you have more then 1 registrar domain using the same registrar you can associate all of them with the NAP:

* Select a '''sip domain''' from the '''Available''' list
* Click '''"<<"''' to associate the SIP Domain with the NAP
[[Image:Associate_SIP_NAP.png]]

=Call Route=

You must set up call routing for your system. [[Call routing]] refers to the ability to route calls based on criteria such as origin, destination, time of day, service provider rates, and more.

==Route Configuration for Remote Phones to SIP Server==

1- Click '''Routes''' in the navigation panel

[[Image:CreateCallRoute_0.png]]

2- Click '''Create New Static Route'''

[[Image:CreateCallRoute_1.png]]

3- Create the new route:

* Enter a '''RoutesetName''' for the route
* Select '''SIP_REMOTE''', to match calls from Trunk NAP
* Select '''SIP_SERVER_NAP'''
* Click '''Create'''

[[Image:Route_Remote_to_Server.png]]

4- Verify that the '''" Route was successfully created"''' message appears and that the new route is listed in the Routeset list

[[Image:Route_Remote_to_Server_1.png]]

==Route Configuration for SIP Server to Remote Phones==

1- Click '''Routes''' in the navigation panel

[[Image:CreateCallRoute_0.png]]

2- Click '''Create New Static Route'''

[[Image:CreateCallRoute_1.png]]

3- Create the new route:

* Enter a '''RoutesetName''' for the route
* Select '''SIP_SERVER_NAP''', to match calls from Trunk NAP
* Select ''' (By registered user) '''
* Click '''Create'''

[[Image:Route_Server_to_Remote_2.png]]

4- Verify that the '''" Route was successfully created"''' message appears and that the new route is listed in the Routeset list

[[Image:Route_Server_to_Remote_3.png]]

=Activating the Configuration=

Changes made to the configuration of the FreeSBC units are stored in the OAM&P Configuration and Logging database. In order for changes to be used by the system, they must first be activated. This is done at the system level and accessed from the Navigation panel.

Check the following link for activating the configuration;

[[Toolpack:Activating_the_Configuration_D]]

=SBC Use Cases=
[[Toolpack:Tsbc Use Cases A|SBC Use Cases]] 
[[FreeSBC_Use_Cases|FreeSBC Use Cases]]

=SBC Tutorial Guide V3.0=
[[Tmedia Tsig Tdev Tutorial Guide v3.0|Version 3.0]]

[[Category:Revise on Major]]

FreeSBC: Multiple Domains/Hosted PBXs Configuration

2020-10-21T07:32:02Z

William Wong: /* SIP NAP Configuration for Local Area Network */

=== '''''Applies to version: v3.0, 3.1''''' ===

{{DISPLAYTITLE:FreeSBC:Multiple Domains/Hosted PBXs Configuration:Example}}
=Introduction=
FreeSBC Hosted PBX Example (see [[FreeSBC:Hosted PBX:Configuration A|FreeSBC:Hosted PBX]]) provides a step by step Hosted PBX Configuration of [[FreeSBC|FreeSBC]] systems, using the Web Portal configuration tool. What will be discussed here, with guide for example configuration, is when there are multiple domains per PBX or several PBXs with different domains. This multiple domains/PBXs configuration is basically the extension of what is configured in single domain/hosted PBX, that including the setting of different domains, followed by specific routing for different domain destinations.

=Hosted PBXs Example=

[[Image:Hosted_Multiple_PBX_topo_1.png]] 

From above, there will be several Hosted PBXs like Hosted PBX1, Hosted PBX2, Hosted PBX3, etc. And each Hosted PBX will use its own domain like example1.com for PBX1, example2.com for PBX2, and so on. This multiple domain scenario can also be applied to single Hosted PBX with different several domains that will be using the similar configuration.

With external path over the WAN, different phones belonging to different Hosted PBXs register to domain corresponding to the PBX/Registrar the phone is subscribing to.

Prerequisites and below configuration sections need to be done as listed in [[FreeSBC:Hosted PBX:Configuration A|single Hosted PBX configuration]] before proceeding:

* IP Network Configuration
* SIP Stack Configuration

=SIP NAP=

A Network Access Point or NAP represents the entry point to another network or destination peer (e.g. SIP proxy, SIP trunk, etc)

==SIP NAP Configuration for Local Area Network==

To create a new NAP:

1- Click '''NAPs''' in the navigation panel

[[Image:NAP_NavigationMenu.png]]

2- Click '''Create New NAP'''

[[Image:Create_New_NAP_SIP.png]]

3- Create the new NAP:

* Enter a '''name''' for the NAP
* Click '''Create'''

[[Image:SIP_Server_NAP_Create.png]]

4- Verify that the '''NAP was successfully created''' message appears

[[Image:Create_New_NAP_SIP2.png]]

5- Associate a SIP transport server with the new NAP:

* Select a '''SIP Transport Server''' from the '''Available''' list
* Click '''"<<"''' to associate the '''LAN_SIP_TS''' with the NAP

[[Image:SIP_Server_NAP_Create_1.png]]

6- Enter SIP Server proxy address:

[[Image:SIP_Server_NAP_Create_2.png]]

7- Associate a Port range with the new NAP:

* Select a '''port range''' from the '''Available''' list
* Click '''"<<"''' to associate '''LAN_Vlan:0''' Port range with the NAP

[[Image:SIP_Server_NAP_Create_3.png]]

9- Configure settings for the following parameter groups as required:
*Registration Parameters
*Authentication Parameters
*Network Address Translation
*SIP-I Parameters
*Advanced Parameters

*Click '''Save'''

[[Image:SIP_Server_NAP_Create_4.png]]

* Here, we need to repeat above steps and create NAP for every Hosted PBX, for example, NAP_PBX1 for PBX1 (192.168.1.10), NAP_PBX2 for PBX2 (192.168.1.11), and NAP_PBX3 for PBX3 (192.168.1.12) and so on if more.

==Access Control List==

FreeSBC will automatically create Access Control List for each NAP you created.

[[Image:Remote_Workers_Access_Control_List_1.png]]

If you double-click one of the created ACL, you will see FreeSBC only accept the calls if source IP matches. In this sample; the calls from 192.168.1.10 will accepted only.

[[Image:SIP_Access_Control_List_2.png]]

Rules for NAP_PBX1, NAP_PBX2, NAP_PBX3,.. will be created automatically in the ACL.

=SIP DOMAIN=

A SIP domain represents a grouping of devices (or users) that can communicate with one another.
You must configure SIP Registration Domain for your system. The first step in doing so is to create a SIP Domain:

==Create New SIP Registration Domain==

1- Click '''SIP Domain''' in the navigation panel

[[Image:SIP_Domain_NavigationMenu.png]]

2- Click '''Create New Domain'''

[[Image:Create_SIP_Domain.png]]

3- Create the new Domain:

* Enter a configuration '''Name''' for this domain.
* Enter a '''Domain Name''' for the SIP Registration Domain (the domain can be an FQDN or an IP address)
* Set the number '''Maximum Registered Users''' for this domain
* Set the Expires value used by SBC when the remote device doesn't supply one ('''Default Contact Expire''')
* Select '''Routing Method''' the system will use to route calls to registered users (if enabled in routing scripts).
** '''Register source''': Sends SIP Invite to the registering source IP address.
** '''Contact''': Sends SIP Invite to the 'contact' from the Register message.
* Set the '''Default Contact Expiration''', this value will be used when no ''Expires'' value is supplied by the user agent.
* Set the '''Minimum Contact Expiration''', this is the minimum ''Expires'' value that can be supplied by a user agent. Lower values will be rejected with a 423 'Interval too brief' response.
* Set the '''Maximum Contact Expiration''', this is the maximum ''Expires'' value that can be supplied by a user agent. The higher value is replaced by this parameter.
* Forwarding Parameters:
** Select the Registration '''Forwarding Mode''' to the registrar:
*** '''Contact Remapping''': Changes the user and the IP address.
*** '''Contact Passthrough''': Doesn't change anything. Enables devices to be contacted directly without going through the SBC.
** Set '''Minimum Registrar Expiration''', this is the minimum ''Expires'' value sent by the SBC to the registrar. If a user agent 'Expires' value is greater than this parameter, the SBC will do rate adaptation between the user agent and the Registrar.
** Set '''Maximum Pending Register Forward''', the maximum number of simultaneous pending register requests allowed for this domain. New REGISTER request is being refused passed this threshold.
** Set '''Maximum Simultaneous Register Forward''', the maximum number of simultaneous active register requests allowed for this domain. New REGISTER request is being refused passed this threshold.
* Click '''Create'''

[[Image:register_domain_1.png]]

4- Verify that the '''Sip registration domain was successfully created''' message appears

[[Image:register_domain_2.png]]
 

5- If your registrar has multiple domains, you need to create all the domains one by one.

Repeat above steps for every domain designated to every PBX, for example, example1.com for PBX1, example2.com for PBX2, and so on. This also applies to the case of multiple domains from the same PBX/Registrar as pointed out in step 5 above.

==Create New SIP Registration Registrar==

A SIP registrar represents a SIP endpoint that provides a location service.
You must configure SIP Registrar for your system. The first step in doing so is to select your SIP Domain:

1- Click on your domain in the '''SIP Domain List'''

[[Image:SIP_Domain_Select.png]]

2- Click '''Create New SIP Registration Registrar'''

[[Image:New_SIP_Registrar_Create.png]]

3- Create the new SIP Registration Registrar

* Enter a '''Name''' for the SIP Registration Registrar
* Select pre defined SIP Proxy NAP from drop-down menu
* Click '''Create'''

[[Image:SIP_Registrar_Select.png]]

4- Verify that the '''Sip registration registrar was successfully created''' message appears

[[Image:Create_New_SIP_Registrar2.png]]

==Associate a SIP Domain with the new NAP==
Associate a SIP Domain with the new NAP. If you have more then 1 registrar domain using the same registrar you can associate all of them with the NAP:

* Select a '''sip domain''' from the '''Available''' list
* Click '''"<<"''' to associate the SIP Domain with the NAP
[[Image:Associate_SIP_NAP.png]]

=Call Route=

You must set up call routing for your system. [[Call routing]] refers to the ability to route calls based on criteria such as origin, destination, time of day, service provider rates, and more.

==Route Configuration for Remote Phones to SIP Server==

1- Click '''Routes''' in the navigation panel

[[Image:CreateCallRoute_0.png]]

2- Click '''Create New Static Route'''

[[Image:CreateCallRoute_1.png]]

3- Create the new route:

* Enter a '''RoutesetName''' for the route
* Select '''SIP_REMOTE''', to match calls from Trunk NAP
* Select '''SIP_SERVER_NAP'''
* Click '''Create'''

[[Image:Route_Remote_to_Server.png]]

4- Verify that the '''" Route was successfully created"''' message appears and that the new route is listed in the Routeset list

[[Image:Route_Remote_to_Server_1.png]]

==Route Configuration for SIP Server to Remote Phones==

1- Click '''Routes''' in the navigation panel

[[Image:CreateCallRoute_0.png]]

2- Click '''Create New Static Route'''

[[Image:CreateCallRoute_1.png]]

3- Create the new route:

* Enter a '''RoutesetName''' for the route
* Select '''SIP_SERVER_NAP''', to match calls from Trunk NAP
* Select ''' (By registered user) '''
* Click '''Create'''

[[Image:Route_Server_to_Remote_2.png]]

4- Verify that the '''" Route was successfully created"''' message appears and that the new route is listed in the Routeset list

[[Image:Route_Server_to_Remote_3.png]]

=Activating the Configuration=

Changes made to the configuration of the FreeSBC units are stored in the OAM&P Configuration and Logging database. In order for changes to be used by the system, they must first be activated. This is done at the system level and accessed from the Navigation panel.

Check the following link for activating the configuration;

[[Toolpack:Activating_the_Configuration_D]]

=SBC Use Cases=
[[Toolpack:Tsbc Use Cases A|SBC Use Cases]] 
[[FreeSBC_Use_Cases|FreeSBC Use Cases]]

=SBC Tutorial Guide V3.0=
[[Tmedia Tsig Tdev Tutorial Guide v3.0|Version 3.0]]

[[Category:Revise on Major]]

FreeSBC: Multiple Domains/Hosted PBXs Configuration

2020-10-21T07:30:39Z

William Wong: /* SIP NAP Configuration for Local Area Network */

=== '''''Applies to version: v3.0, 3.1''''' ===

{{DISPLAYTITLE:FreeSBC:Multiple Domains/Hosted PBXs Configuration:Example}}
=Introduction=
FreeSBC Hosted PBX Example (see [[FreeSBC:Hosted PBX:Configuration A|FreeSBC:Hosted PBX]]) provides a step by step Hosted PBX Configuration of [[FreeSBC|FreeSBC]] systems, using the Web Portal configuration tool. What will be discussed here, with guide for example configuration, is when there are multiple domains per PBX or several PBXs with different domains. This multiple domains/PBXs configuration is basically the extension of what is configured in single domain/hosted PBX, that including the setting of different domains, followed by specific routing for different domain destinations.

=Hosted PBXs Example=

[[Image:Hosted_Multiple_PBX_topo_1.png]] 

From above, there will be several Hosted PBXs like Hosted PBX1, Hosted PBX2, Hosted PBX3, etc. And each Hosted PBX will use its own domain like example1.com for PBX1, example2.com for PBX2, and so on. This multiple domain scenario can also be applied to single Hosted PBX with different several domains that will be using the similar configuration.

With external path over the WAN, different phones belonging to different Hosted PBXs register to domain corresponding to the PBX/Registrar the phone is subscribing to.

Prerequisites and below configuration sections need to be done as listed in [[FreeSBC:Hosted PBX:Configuration A|single Hosted PBX configuration]] before proceeding:

* IP Network Configuration
* SIP Stack Configuration

=SIP NAP=

A Network Access Point or NAP represents the entry point to another network or destination peer (e.g. SIP proxy, SIP trunk, etc)

==SIP NAP Configuration for Local Area Network==

To create a new NAP:

1- Click '''NAPs''' in the navigation panel

[[Image:NAP_NavigationMenu.png]]

2- Click '''Create New NAP'''

[[Image:Create_New_NAP_SIP.png]]

3- Create the new NAP:

* Enter a '''name''' for the NAP
* Click '''Create'''

[[Image:SIP_Server_NAP_Create.png]]

4- Verify that the '''NAP was successfully created''' message appears

[[Image:Create_New_NAP_SIP2.png]]

5- Associate a SIP transport server with the new NAP:

* Select a '''SIP Transport Server''' from the '''Available''' list
* Click '''"<<"''' to associate the '''LAN_SIP_TS''' with the NAP

[[Image:SIP_Server_NAP_Create_1.png]]

6- Enter SIP Server proxy address:

[[Image:SIP_Server_NAP_Create_2.png]]

7- Associate a Port range with the new NAP:

* Select a '''port range''' from the '''Available''' list
* Click '''"<<"''' to associate '''LAN_Vlan:0''' Port range with the NAP

[[Image:SIP_Server_NAP_Create_3.png]]

9- Configure settings for the following parameter groups as required:
*Registration Parameters
*Authentication Parameters
*Network Address Translation
*SIP-I Parameters
*Advanced Parameters

*Click '''Save'''

[[Image:SIP_Server_NAP_Create_4.png]]

* Here, we need to repeat above steps and create NAP for every Hosted PBX, for example, NAP_PBX1 for PBX1 (192.168.1.10), NAP_PBX2 for PBX2 (192.168.1.11), and NAP_PBX3 for PBX3 (192.168.1.12)and so on if more.

==Access Control List==

FreeSBC will automatically create Access Control List for each NAP you created.

[[Image:Remote_Workers_Access_Control_List_1.png]]

If you double-click one of the created ACL, you will see FreeSBC only accept the calls if source IP matches. In this sample; the calls from 192.168.1.10 will accepted only.

[[Image:SIP_Access_Control_List_2.png]]

Rules for NAP_PBX1, NAP_PBX2, NAP_PBX3,.. will be created automatically in the ACL.

=SIP DOMAIN=

A SIP domain represents a grouping of devices (or users) that can communicate with one another.
You must configure SIP Registration Domain for your system. The first step in doing so is to create a SIP Domain:

==Create New SIP Registration Domain==

1- Click '''SIP Domain''' in the navigation panel

[[Image:SIP_Domain_NavigationMenu.png]]

2- Click '''Create New Domain'''

[[Image:Create_SIP_Domain.png]]

3- Create the new Domain:

* Enter a configuration '''Name''' for this domain.
* Enter a '''Domain Name''' for the SIP Registration Domain (the domain can be an FQDN or an IP address)
* Set the number '''Maximum Registered Users''' for this domain
* Set the Expires value used by SBC when the remote device doesn't supply one ('''Default Contact Expire''')
* Select '''Routing Method''' the system will use to route calls to registered users (if enabled in routing scripts).
** '''Register source''': Sends SIP Invite to the registering source IP address.
** '''Contact''': Sends SIP Invite to the 'contact' from the Register message.
* Set the '''Default Contact Expiration''', this value will be used when no ''Expires'' value is supplied by the user agent.
* Set the '''Minimum Contact Expiration''', this is the minimum ''Expires'' value that can be supplied by a user agent. Lower values will be rejected with a 423 'Interval too brief' response.
* Set the '''Maximum Contact Expiration''', this is the maximum ''Expires'' value that can be supplied by a user agent. The higher value is replaced by this parameter.
* Forwarding Parameters:
** Select the Registration '''Forwarding Mode''' to the registrar:
*** '''Contact Remapping''': Changes the user and the IP address.
*** '''Contact Passthrough''': Doesn't change anything. Enables devices to be contacted directly without going through the SBC.
** Set '''Minimum Registrar Expiration''', this is the minimum ''Expires'' value sent by the SBC to the registrar. If a user agent 'Expires' value is greater than this parameter, the SBC will do rate adaptation between the user agent and the Registrar.
** Set '''Maximum Pending Register Forward''', the maximum number of simultaneous pending register requests allowed for this domain. New REGISTER request is being refused passed this threshold.
** Set '''Maximum Simultaneous Register Forward''', the maximum number of simultaneous active register requests allowed for this domain. New REGISTER request is being refused passed this threshold.
* Click '''Create'''

[[Image:register_domain_1.png]]

4- Verify that the '''Sip registration domain was successfully created''' message appears

[[Image:register_domain_2.png]]
 

5- If your registrar has multiple domains, you need to create all the domains one by one.

Repeat above steps for every domain designated to every PBX, for example, example1.com for PBX1, example2.com for PBX2, and so on. This also applies to the case of multiple domains from the same PBX/Registrar as pointed out in step 5 above.

==Create New SIP Registration Registrar==

A SIP registrar represents a SIP endpoint that provides a location service.
You must configure SIP Registrar for your system. The first step in doing so is to select your SIP Domain:

1- Click on your domain in the '''SIP Domain List'''

[[Image:SIP_Domain_Select.png]]

2- Click '''Create New SIP Registration Registrar'''

[[Image:New_SIP_Registrar_Create.png]]

3- Create the new SIP Registration Registrar

* Enter a '''Name''' for the SIP Registration Registrar
* Select pre defined SIP Proxy NAP from drop-down menu
* Click '''Create'''

[[Image:SIP_Registrar_Select.png]]

4- Verify that the '''Sip registration registrar was successfully created''' message appears

[[Image:Create_New_SIP_Registrar2.png]]

==Associate a SIP Domain with the new NAP==
Associate a SIP Domain with the new NAP. If you have more then 1 registrar domain using the same registrar you can associate all of them with the NAP:

* Select a '''sip domain''' from the '''Available''' list
* Click '''"<<"''' to associate the SIP Domain with the NAP
[[Image:Associate_SIP_NAP.png]]

=Call Route=

You must set up call routing for your system. [[Call routing]] refers to the ability to route calls based on criteria such as origin, destination, time of day, service provider rates, and more.

==Route Configuration for Remote Phones to SIP Server==

1- Click '''Routes''' in the navigation panel

[[Image:CreateCallRoute_0.png]]

2- Click '''Create New Static Route'''

[[Image:CreateCallRoute_1.png]]

3- Create the new route:

* Enter a '''RoutesetName''' for the route
* Select '''SIP_REMOTE''', to match calls from Trunk NAP
* Select '''SIP_SERVER_NAP'''
* Click '''Create'''

[[Image:Route_Remote_to_Server.png]]

4- Verify that the '''" Route was successfully created"''' message appears and that the new route is listed in the Routeset list

[[Image:Route_Remote_to_Server_1.png]]

==Route Configuration for SIP Server to Remote Phones==

1- Click '''Routes''' in the navigation panel

[[Image:CreateCallRoute_0.png]]

2- Click '''Create New Static Route'''

[[Image:CreateCallRoute_1.png]]

3- Create the new route:

* Enter a '''RoutesetName''' for the route
* Select '''SIP_SERVER_NAP''', to match calls from Trunk NAP
* Select ''' (By registered user) '''
* Click '''Create'''

[[Image:Route_Server_to_Remote_2.png]]

4- Verify that the '''" Route was successfully created"''' message appears and that the new route is listed in the Routeset list

[[Image:Route_Server_to_Remote_3.png]]

=Activating the Configuration=

Changes made to the configuration of the FreeSBC units are stored in the OAM&P Configuration and Logging database. In order for changes to be used by the system, they must first be activated. This is done at the system level and accessed from the Navigation panel.

Check the following link for activating the configuration;

[[Toolpack:Activating_the_Configuration_D]]

=SBC Use Cases=
[[Toolpack:Tsbc Use Cases A|SBC Use Cases]] 
[[FreeSBC_Use_Cases|FreeSBC Use Cases]]

=SBC Tutorial Guide V3.0=
[[Tmedia Tsig Tdev Tutorial Guide v3.0|Version 3.0]]

[[Category:Revise on Major]]

FreeSBC: Multiple Domains/Hosted PBXs Configuration

2020-10-21T07:29:04Z

William Wong: /* SIP NAP Configuration for Local Area Network */

=== '''''Applies to version: v3.0, 3.1''''' ===

{{DISPLAYTITLE:FreeSBC:Multiple Domains/Hosted PBXs Configuration:Example}}
=Introduction=
FreeSBC Hosted PBX Example (see [[FreeSBC:Hosted PBX:Configuration A|FreeSBC:Hosted PBX]]) provides a step by step Hosted PBX Configuration of [[FreeSBC|FreeSBC]] systems, using the Web Portal configuration tool. What will be discussed here, with guide for example configuration, is when there are multiple domains per PBX or several PBXs with different domains. This multiple domains/PBXs configuration is basically the extension of what is configured in single domain/hosted PBX, that including the setting of different domains, followed by specific routing for different domain destinations.

=Hosted PBXs Example=

[[Image:Hosted_Multiple_PBX_topo_1.png]] 

From above, there will be several Hosted PBXs like Hosted PBX1, Hosted PBX2, Hosted PBX3, etc. And each Hosted PBX will use its own domain like example1.com for PBX1, example2.com for PBX2, and so on. This multiple domain scenario can also be applied to single Hosted PBX with different several domains that will be using the similar configuration.

With external path over the WAN, different phones belonging to different Hosted PBXs register to domain corresponding to the PBX/Registrar the phone is subscribing to.

Prerequisites and below configuration sections need to be done as listed in [[FreeSBC:Hosted PBX:Configuration A|single Hosted PBX configuration]] before proceeding:

* IP Network Configuration
* SIP Stack Configuration

=SIP NAP=

A Network Access Point or NAP represents the entry point to another network or destination peer (e.g. SIP proxy, SIP trunk, etc)

==SIP NAP Configuration for Local Area Network==

To create a new NAP:

1- Click '''NAPs''' in the navigation panel

[[Image:NAP_NavigationMenu.png]]

2- Click '''Create New NAP'''

[[Image:Create_New_NAP_SIP.png]]

3- Create the new NAP:

* Enter a '''name''' for the NAP
* Click '''Create'''

[[Image:SIP_Server_NAP_Create.png]]

4- Verify that the '''NAP was successfully created''' message appears

[[Image:Create_New_NAP_SIP2.png]]

5- Associate a SIP transport server with the new NAP:

* Select a '''SIP Transport Server''' from the '''Available''' list
* Click '''"<<"''' to associate the '''LAN_SIP_TS''' with the NAP

[[Image:SIP_Server_NAP_Create_1.png]]

6- Enter SIP Server proxy address:

[[Image:SIP_Server_NAP_Create_2.png]]

7- Associate a Port range with the new NAP:

* Select a '''port range''' from the '''Available''' list
* Click '''"<<"''' to associate '''LAN_Vlan:0''' Port range with the NAP

[[Image:SIP_Server_NAP_Create_3.png]]

9- Configure settings for the following parameter groups as required:
*Registration Parameters
*Authentication Parameters
*Network Address Translation
*SIP-I Parameters
*Advanced Parameters

*Click '''Save'''

[[Image:SIP_Server_NAP_Create_4.png]]

* Here, we need to repeat above steps and create NAP for every Hosted PBX, for example, NAP_PBX1 for PBX1, NAP_PBX2 for PBX2, and so on.

==Access Control List==

FreeSBC will automatically create Access Control List for each NAP you created.

[[Image:Remote_Workers_Access_Control_List_1.png]]

If you double-click one of the created ACL, you will see FreeSBC only accept the calls if source IP matches. In this sample; the calls from 192.168.1.10 will accepted only.

[[Image:SIP_Access_Control_List_2.png]]

Rules for NAP_PBX1, NAP_PBX2, NAP_PBX3,.. will be created automatically in the ACL.

=SIP DOMAIN=

A SIP domain represents a grouping of devices (or users) that can communicate with one another.
You must configure SIP Registration Domain for your system. The first step in doing so is to create a SIP Domain:

==Create New SIP Registration Domain==

1- Click '''SIP Domain''' in the navigation panel

[[Image:SIP_Domain_NavigationMenu.png]]

2- Click '''Create New Domain'''

[[Image:Create_SIP_Domain.png]]

3- Create the new Domain:

* Enter a configuration '''Name''' for this domain.
* Enter a '''Domain Name''' for the SIP Registration Domain (the domain can be an FQDN or an IP address)
* Set the number '''Maximum Registered Users''' for this domain
* Set the Expires value used by SBC when the remote device doesn't supply one ('''Default Contact Expire''')
* Select '''Routing Method''' the system will use to route calls to registered users (if enabled in routing scripts).
** '''Register source''': Sends SIP Invite to the registering source IP address.
** '''Contact''': Sends SIP Invite to the 'contact' from the Register message.
* Set the '''Default Contact Expiration''', this value will be used when no ''Expires'' value is supplied by the user agent.
* Set the '''Minimum Contact Expiration''', this is the minimum ''Expires'' value that can be supplied by a user agent. Lower values will be rejected with a 423 'Interval too brief' response.
* Set the '''Maximum Contact Expiration''', this is the maximum ''Expires'' value that can be supplied by a user agent. The higher value is replaced by this parameter.
* Forwarding Parameters:
** Select the Registration '''Forwarding Mode''' to the registrar:
*** '''Contact Remapping''': Changes the user and the IP address.
*** '''Contact Passthrough''': Doesn't change anything. Enables devices to be contacted directly without going through the SBC.
** Set '''Minimum Registrar Expiration''', this is the minimum ''Expires'' value sent by the SBC to the registrar. If a user agent 'Expires' value is greater than this parameter, the SBC will do rate adaptation between the user agent and the Registrar.
** Set '''Maximum Pending Register Forward''', the maximum number of simultaneous pending register requests allowed for this domain. New REGISTER request is being refused passed this threshold.
** Set '''Maximum Simultaneous Register Forward''', the maximum number of simultaneous active register requests allowed for this domain. New REGISTER request is being refused passed this threshold.
* Click '''Create'''

[[Image:register_domain_1.png]]

4- Verify that the '''Sip registration domain was successfully created''' message appears

[[Image:register_domain_2.png]]
 

5- If your registrar has multiple domains, you need to create all the domains one by one.

Repeat above steps for every domain designated to every PBX, for example, example1.com for PBX1, example2.com for PBX2, and so on. This also applies to the case of multiple domains from the same PBX/Registrar as pointed out in step 5 above.

==Create New SIP Registration Registrar==

A SIP registrar represents a SIP endpoint that provides a location service.
You must configure SIP Registrar for your system. The first step in doing so is to select your SIP Domain:

1- Click on your domain in the '''SIP Domain List'''

[[Image:SIP_Domain_Select.png]]

2- Click '''Create New SIP Registration Registrar'''

[[Image:New_SIP_Registrar_Create.png]]

3- Create the new SIP Registration Registrar

* Enter a '''Name''' for the SIP Registration Registrar
* Select pre defined SIP Proxy NAP from drop-down menu
* Click '''Create'''

[[Image:SIP_Registrar_Select.png]]

4- Verify that the '''Sip registration registrar was successfully created''' message appears

[[Image:Create_New_SIP_Registrar2.png]]

==Associate a SIP Domain with the new NAP==
Associate a SIP Domain with the new NAP. If you have more then 1 registrar domain using the same registrar you can associate all of them with the NAP:

* Select a '''sip domain''' from the '''Available''' list
* Click '''"<<"''' to associate the SIP Domain with the NAP
[[Image:Associate_SIP_NAP.png]]

=Call Route=

You must set up call routing for your system. [[Call routing]] refers to the ability to route calls based on criteria such as origin, destination, time of day, service provider rates, and more.

==Route Configuration for Remote Phones to SIP Server==

1- Click '''Routes''' in the navigation panel

[[Image:CreateCallRoute_0.png]]

2- Click '''Create New Static Route'''

[[Image:CreateCallRoute_1.png]]

3- Create the new route:

* Enter a '''RoutesetName''' for the route
* Select '''SIP_REMOTE''', to match calls from Trunk NAP
* Select '''SIP_SERVER_NAP'''
* Click '''Create'''

[[Image:Route_Remote_to_Server.png]]

4- Verify that the '''" Route was successfully created"''' message appears and that the new route is listed in the Routeset list

[[Image:Route_Remote_to_Server_1.png]]

==Route Configuration for SIP Server to Remote Phones==

1- Click '''Routes''' in the navigation panel

[[Image:CreateCallRoute_0.png]]

2- Click '''Create New Static Route'''

[[Image:CreateCallRoute_1.png]]

3- Create the new route:

* Enter a '''RoutesetName''' for the route
* Select '''SIP_SERVER_NAP''', to match calls from Trunk NAP
* Select ''' (By registered user) '''
* Click '''Create'''

[[Image:Route_Server_to_Remote_2.png]]

4- Verify that the '''" Route was successfully created"''' message appears and that the new route is listed in the Routeset list

[[Image:Route_Server_to_Remote_3.png]]

=Activating the Configuration=

Changes made to the configuration of the FreeSBC units are stored in the OAM&P Configuration and Logging database. In order for changes to be used by the system, they must first be activated. This is done at the system level and accessed from the Navigation panel.

Check the following link for activating the configuration;

[[Toolpack:Activating_the_Configuration_D]]

=SBC Use Cases=
[[Toolpack:Tsbc Use Cases A|SBC Use Cases]] 
[[FreeSBC_Use_Cases|FreeSBC Use Cases]]

=SBC Tutorial Guide V3.0=
[[Tmedia Tsig Tdev Tutorial Guide v3.0|Version 3.0]]

[[Category:Revise on Major]]

FreeSBC: Multiple Domains/Hosted PBXs Configuration

2020-10-21T07:27:45Z

William Wong: /* Hosted PBXs Example */

=== '''''Applies to version: v3.0, 3.1''''' ===

{{DISPLAYTITLE:FreeSBC:Multiple Domains/Hosted PBXs Configuration:Example}}
=Introduction=
FreeSBC Hosted PBX Example (see [[FreeSBC:Hosted PBX:Configuration A|FreeSBC:Hosted PBX]]) provides a step by step Hosted PBX Configuration of [[FreeSBC|FreeSBC]] systems, using the Web Portal configuration tool. What will be discussed here, with guide for example configuration, is when there are multiple domains per PBX or several PBXs with different domains. This multiple domains/PBXs configuration is basically the extension of what is configured in single domain/hosted PBX, that including the setting of different domains, followed by specific routing for different domain destinations.

=Hosted PBXs Example=

[[Image:Hosted_Multiple_PBX_topo_1.png]] 

From above, there will be several Hosted PBXs like Hosted PBX1, Hosted PBX2, Hosted PBX3, etc. And each Hosted PBX will use its own domain like example1.com for PBX1, example2.com for PBX2, and so on. This multiple domain scenario can also be applied to single Hosted PBX with different several domains that will be using the similar configuration.

With external path over the WAN, different phones belonging to different Hosted PBXs register to domain corresponding to the PBX/Registrar the phone is subscribing to.

Prerequisites and below configuration sections need to be done as listed in [[FreeSBC:Hosted PBX:Configuration A|single Hosted PBX configuration]] before proceeding:

* IP Network Configuration
* SIP Stack Configuration

=SIP NAP=

A Network Access Point or NAP represents the entry point to another network or destination peer (e.g. SIP proxy, SIP trunk, etc)

==SIP NAP Configuration for Local Area Network==

To create a new NAP:

1- Click '''NAPs''' in the navigation panel

[[Image:NAP_NavigationMenu.png]]

2- Click '''Create New NAP'''

[[Image:Create_New_NAP_SIP.png]]

3- Create the new NAP:

* Enter a '''name''' for the NAP
* Click '''Create'''

[[Image:SIP_Server_NAP_Create.png]]

4- Verify that the '''NAP was successfully created''' message appears

[[Image:Create_New_NAP_SIP2.png]]

5- Associate a SIP transport server with the new NAP:

* Select a '''SIP Transport Server''' from the '''Available''' list
* Click '''"<<"''' to associate the '''LAN_SIP_TS''' with the NAP

[[Image:SIP_Server_NAP_Create_1.png]]

6- Enter SIP Server proxy address:

[[Image:SIP_Server_NAP_Create_2.png]]

7- Associate a Port range with the new NAP:

* Select a '''port range''' from the '''Available''' list
* Click '''"<<"''' to associate '''LAN_Vlan:0''' Port range with the NAP

[[Image:SIP_Server_NAP_Create_3.png]]

9- Configure settings for the following parameter groups as required:
*Registration Parameters
*Authentication Parameters
*Network Address Translation
*SIP-I Parameters
*Advanced Parameters

*Click '''Save'''

[[Image:SIP_Server_NAP_Create_4.png]]

Here, we need to repeat above steps and create NAP for every Hosted PBX, for example, NAP_PBX1 for PBX1, NAP_PBX2 for PBX2, and so on.

==Access Control List==

FreeSBC will automatically create Access Control List for each NAP you created.

[[Image:Remote_Workers_Access_Control_List_1.png]]

If you double-click one of the created ACL, you will see FreeSBC only accept the calls if source IP matches. In this sample; the calls from 192.168.1.10 will accepted only.

[[Image:SIP_Access_Control_List_2.png]]

Rules for NAP_PBX1, NAP_PBX2, NAP_PBX3,.. will be created automatically in the ACL.

=SIP DOMAIN=

A SIP domain represents a grouping of devices (or users) that can communicate with one another.
You must configure SIP Registration Domain for your system. The first step in doing so is to create a SIP Domain:

==Create New SIP Registration Domain==

1- Click '''SIP Domain''' in the navigation panel

[[Image:SIP_Domain_NavigationMenu.png]]

2- Click '''Create New Domain'''

[[Image:Create_SIP_Domain.png]]

3- Create the new Domain:

* Enter a configuration '''Name''' for this domain.
* Enter a '''Domain Name''' for the SIP Registration Domain (the domain can be an FQDN or an IP address)
* Set the number '''Maximum Registered Users''' for this domain
* Set the Expires value used by SBC when the remote device doesn't supply one ('''Default Contact Expire''')
* Select '''Routing Method''' the system will use to route calls to registered users (if enabled in routing scripts).
** '''Register source''': Sends SIP Invite to the registering source IP address.
** '''Contact''': Sends SIP Invite to the 'contact' from the Register message.
* Set the '''Default Contact Expiration''', this value will be used when no ''Expires'' value is supplied by the user agent.
* Set the '''Minimum Contact Expiration''', this is the minimum ''Expires'' value that can be supplied by a user agent. Lower values will be rejected with a 423 'Interval too brief' response.
* Set the '''Maximum Contact Expiration''', this is the maximum ''Expires'' value that can be supplied by a user agent. The higher value is replaced by this parameter.
* Forwarding Parameters:
** Select the Registration '''Forwarding Mode''' to the registrar:
*** '''Contact Remapping''': Changes the user and the IP address.
*** '''Contact Passthrough''': Doesn't change anything. Enables devices to be contacted directly without going through the SBC.
** Set '''Minimum Registrar Expiration''', this is the minimum ''Expires'' value sent by the SBC to the registrar. If a user agent 'Expires' value is greater than this parameter, the SBC will do rate adaptation between the user agent and the Registrar.
** Set '''Maximum Pending Register Forward''', the maximum number of simultaneous pending register requests allowed for this domain. New REGISTER request is being refused passed this threshold.
** Set '''Maximum Simultaneous Register Forward''', the maximum number of simultaneous active register requests allowed for this domain. New REGISTER request is being refused passed this threshold.
* Click '''Create'''

[[Image:register_domain_1.png]]

4- Verify that the '''Sip registration domain was successfully created''' message appears

[[Image:register_domain_2.png]]
 

5- If your registrar has multiple domains, you need to create all the domains one by one.

Repeat above steps for every domain designated to every PBX, for example, example1.com for PBX1, example2.com for PBX2, and so on. This also applies to the case of multiple domains from the same PBX/Registrar as pointed out in step 5 above.

==Create New SIP Registration Registrar==

A SIP registrar represents a SIP endpoint that provides a location service.
You must configure SIP Registrar for your system. The first step in doing so is to select your SIP Domain:

1- Click on your domain in the '''SIP Domain List'''

[[Image:SIP_Domain_Select.png]]

2- Click '''Create New SIP Registration Registrar'''

[[Image:New_SIP_Registrar_Create.png]]

3- Create the new SIP Registration Registrar

* Enter a '''Name''' for the SIP Registration Registrar
* Select pre defined SIP Proxy NAP from drop-down menu
* Click '''Create'''

[[Image:SIP_Registrar_Select.png]]

4- Verify that the '''Sip registration registrar was successfully created''' message appears

[[Image:Create_New_SIP_Registrar2.png]]

==Associate a SIP Domain with the new NAP==
Associate a SIP Domain with the new NAP. If you have more then 1 registrar domain using the same registrar you can associate all of them with the NAP:

* Select a '''sip domain''' from the '''Available''' list
* Click '''"<<"''' to associate the SIP Domain with the NAP
[[Image:Associate_SIP_NAP.png]]

=Call Route=

You must set up call routing for your system. [[Call routing]] refers to the ability to route calls based on criteria such as origin, destination, time of day, service provider rates, and more.

==Route Configuration for Remote Phones to SIP Server==

1- Click '''Routes''' in the navigation panel

[[Image:CreateCallRoute_0.png]]

2- Click '''Create New Static Route'''

[[Image:CreateCallRoute_1.png]]

3- Create the new route:

* Enter a '''RoutesetName''' for the route
* Select '''SIP_REMOTE''', to match calls from Trunk NAP
* Select '''SIP_SERVER_NAP'''
* Click '''Create'''

[[Image:Route_Remote_to_Server.png]]

4- Verify that the '''" Route was successfully created"''' message appears and that the new route is listed in the Routeset list

[[Image:Route_Remote_to_Server_1.png]]

==Route Configuration for SIP Server to Remote Phones==

1- Click '''Routes''' in the navigation panel

[[Image:CreateCallRoute_0.png]]

2- Click '''Create New Static Route'''

[[Image:CreateCallRoute_1.png]]

3- Create the new route:

* Enter a '''RoutesetName''' for the route
* Select '''SIP_SERVER_NAP''', to match calls from Trunk NAP
* Select ''' (By registered user) '''
* Click '''Create'''

[[Image:Route_Server_to_Remote_2.png]]

4- Verify that the '''" Route was successfully created"''' message appears and that the new route is listed in the Routeset list

[[Image:Route_Server_to_Remote_3.png]]

=Activating the Configuration=

Changes made to the configuration of the FreeSBC units are stored in the OAM&P Configuration and Logging database. In order for changes to be used by the system, they must first be activated. This is done at the system level and accessed from the Navigation panel.

Check the following link for activating the configuration;

[[Toolpack:Activating_the_Configuration_D]]

=SBC Use Cases=
[[Toolpack:Tsbc Use Cases A|SBC Use Cases]] 
[[FreeSBC_Use_Cases|FreeSBC Use Cases]]

=SBC Tutorial Guide V3.0=
[[Tmedia Tsig Tdev Tutorial Guide v3.0|Version 3.0]]

[[Category:Revise on Major]]

FreeSBC: Multiple Domains/Hosted PBXs Configuration

2020-10-21T07:27:10Z

William Wong:

=== '''''Applies to version: v3.0, 3.1''''' ===

{{DISPLAYTITLE:FreeSBC:Multiple Domains/Hosted PBXs Configuration:Example}}
=Introduction=
FreeSBC Hosted PBX Example (see [[FreeSBC:Hosted PBX:Configuration A|FreeSBC:Hosted PBX]]) provides a step by step Hosted PBX Configuration of [[FreeSBC|FreeSBC]] systems, using the Web Portal configuration tool. What will be discussed here, with guide for example configuration, is when there are multiple domains per PBX or several PBXs with different domains. This multiple domains/PBXs configuration is basically the extension of what is configured in single domain/hosted PBX, that including the setting of different domains, followed by specific routing for different domain destinations.

=Hosted PBXs Example=

[[Image:Hosted_Multiple_PBX_topo_1.png]] 

From above, there will be several Hosted PBXs like Hosted PBX1, Hosted PBX2, Hosted PBX3, etc. And each Hosted PBX will use its own domain like example1.com for PBX1, example2.com for PBX2, and so on. This multiple domain scenario can also be applied to single Hosted PBX with different several domains that will be using the similar configuration.

With external path over the WAN, different phones belonging to different Hosted PBXs register to domain corresponding to the PBX/Registrar the phone is subscribing to.

Prerequisites and below configuration sections need to be done as listed in [[FreeSBC:Hosted PBX:Configuration A|single Hosted PBX configuration]] before proceeding:

* IP Network Configuration
* SIP Stack Configuration

=SIP NAP=

A Network Access Point or NAP represents the entry point to another network or destination peer (e.g. SIP proxy, SIP trunk, etc)

==SIP NAP Configuration for Local Area Network==

To create a new NAP:

1- Click '''NAPs''' in the navigation panel

[[Image:NAP_NavigationMenu.png]]

2- Click '''Create New NAP'''

[[Image:Create_New_NAP_SIP.png]]

3- Create the new NAP:

* Enter a '''name''' for the NAP
* Click '''Create'''

[[Image:SIP_Server_NAP_Create.png]]

4- Verify that the '''NAP was successfully created''' message appears

[[Image:Create_New_NAP_SIP2.png]]

5- Associate a SIP transport server with the new NAP:

* Select a '''SIP Transport Server''' from the '''Available''' list
* Click '''"<<"''' to associate the '''LAN_SIP_TS''' with the NAP

[[Image:SIP_Server_NAP_Create_1.png]]

6- Enter SIP Server proxy address:

[[Image:SIP_Server_NAP_Create_2.png]]

7- Associate a Port range with the new NAP:

* Select a '''port range''' from the '''Available''' list
* Click '''"<<"''' to associate '''LAN_Vlan:0''' Port range with the NAP

[[Image:SIP_Server_NAP_Create_3.png]]

9- Configure settings for the following parameter groups as required:
*Registration Parameters
*Authentication Parameters
*Network Address Translation
*SIP-I Parameters
*Advanced Parameters

*Click '''Save'''

[[Image:SIP_Server_NAP_Create_4.png]]

Here, we need to repeat above steps and create NAP for every Hosted PBX, for example, NAP_PBX1 for PBX1, NAP_PBX2 for PBX2, and so on.

==Access Control List==

FreeSBC will automatically create Access Control List for each NAP you created.

[[Image:Remote_Workers_Access_Control_List_1.png]]

If you double-click one of the created ACL, you will see FreeSBC only accept the calls if source IP matches. In this sample; the calls from 192.168.1.10 will accepted only.

[[Image:SIP_Access_Control_List_2.png]]

Rules for NAP_PBX1, NAP_PBX2, NAP_PBX3,.. will be created automatically in the ACL.

=SIP DOMAIN=

A SIP domain represents a grouping of devices (or users) that can communicate with one another.
You must configure SIP Registration Domain for your system. The first step in doing so is to create a SIP Domain:

==Create New SIP Registration Domain==

1- Click '''SIP Domain''' in the navigation panel

[[Image:SIP_Domain_NavigationMenu.png]]

2- Click '''Create New Domain'''

[[Image:Create_SIP_Domain.png]]

3- Create the new Domain:

* Enter a configuration '''Name''' for this domain.
* Enter a '''Domain Name''' for the SIP Registration Domain (the domain can be an FQDN or an IP address)
* Set the number '''Maximum Registered Users''' for this domain
* Set the Expires value used by SBC when the remote device doesn't supply one ('''Default Contact Expire''')
* Select '''Routing Method''' the system will use to route calls to registered users (if enabled in routing scripts).
** '''Register source''': Sends SIP Invite to the registering source IP address.
** '''Contact''': Sends SIP Invite to the 'contact' from the Register message.
* Set the '''Default Contact Expiration''', this value will be used when no ''Expires'' value is supplied by the user agent.
* Set the '''Minimum Contact Expiration''', this is the minimum ''Expires'' value that can be supplied by a user agent. Lower values will be rejected with a 423 'Interval too brief' response.
* Set the '''Maximum Contact Expiration''', this is the maximum ''Expires'' value that can be supplied by a user agent. The higher value is replaced by this parameter.
* Forwarding Parameters:
** Select the Registration '''Forwarding Mode''' to the registrar:
*** '''Contact Remapping''': Changes the user and the IP address.
*** '''Contact Passthrough''': Doesn't change anything. Enables devices to be contacted directly without going through the SBC.
** Set '''Minimum Registrar Expiration''', this is the minimum ''Expires'' value sent by the SBC to the registrar. If a user agent 'Expires' value is greater than this parameter, the SBC will do rate adaptation between the user agent and the Registrar.
** Set '''Maximum Pending Register Forward''', the maximum number of simultaneous pending register requests allowed for this domain. New REGISTER request is being refused passed this threshold.
** Set '''Maximum Simultaneous Register Forward''', the maximum number of simultaneous active register requests allowed for this domain. New REGISTER request is being refused passed this threshold.
* Click '''Create'''

[[Image:register_domain_1.png]]

4- Verify that the '''Sip registration domain was successfully created''' message appears

[[Image:register_domain_2.png]]
 

5- If your registrar has multiple domains, you need to create all the domains one by one.

Repeat above steps for every domain designated to every PBX, for example, example1.com for PBX1, example2.com for PBX2, and so on. This also applies to the case of multiple domains from the same PBX/Registrar as pointed out in step 5 above.

==Create New SIP Registration Registrar==

A SIP registrar represents a SIP endpoint that provides a location service.
You must configure SIP Registrar for your system. The first step in doing so is to select your SIP Domain:

1- Click on your domain in the '''SIP Domain List'''

[[Image:SIP_Domain_Select.png]]

2- Click '''Create New SIP Registration Registrar'''

[[Image:New_SIP_Registrar_Create.png]]

3- Create the new SIP Registration Registrar

* Enter a '''Name''' for the SIP Registration Registrar
* Select pre defined SIP Proxy NAP from drop-down menu
* Click '''Create'''

[[Image:SIP_Registrar_Select.png]]

4- Verify that the '''Sip registration registrar was successfully created''' message appears

[[Image:Create_New_SIP_Registrar2.png]]

==Associate a SIP Domain with the new NAP==
Associate a SIP Domain with the new NAP. If you have more then 1 registrar domain using the same registrar you can associate all of them with the NAP:

* Select a '''sip domain''' from the '''Available''' list
* Click '''"<<"''' to associate the SIP Domain with the NAP
[[Image:Associate_SIP_NAP.png]]

=Call Route=

You must set up call routing for your system. [[Call routing]] refers to the ability to route calls based on criteria such as origin, destination, time of day, service provider rates, and more.

==Route Configuration for Remote Phones to SIP Server==

1- Click '''Routes''' in the navigation panel

[[Image:CreateCallRoute_0.png]]

2- Click '''Create New Static Route'''

[[Image:CreateCallRoute_1.png]]

3- Create the new route:

* Enter a '''RoutesetName''' for the route
* Select '''SIP_REMOTE''', to match calls from Trunk NAP
* Select '''SIP_SERVER_NAP'''
* Click '''Create'''

[[Image:Route_Remote_to_Server.png]]

4- Verify that the '''" Route was successfully created"''' message appears and that the new route is listed in the Routeset list

[[Image:Route_Remote_to_Server_1.png]]

==Route Configuration for SIP Server to Remote Phones==

1- Click '''Routes''' in the navigation panel

[[Image:CreateCallRoute_0.png]]

2- Click '''Create New Static Route'''

[[Image:CreateCallRoute_1.png]]

3- Create the new route:

* Enter a '''RoutesetName''' for the route
* Select '''SIP_SERVER_NAP''', to match calls from Trunk NAP
* Select ''' (By registered user) '''
* Click '''Create'''

[[Image:Route_Server_to_Remote_2.png]]

4- Verify that the '''" Route was successfully created"''' message appears and that the new route is listed in the Routeset list

[[Image:Route_Server_to_Remote_3.png]]

=Activating the Configuration=

Changes made to the configuration of the FreeSBC units are stored in the OAM&P Configuration and Logging database. In order for changes to be used by the system, they must first be activated. This is done at the system level and accessed from the Navigation panel.

Check the following link for activating the configuration;

[[Toolpack:Activating_the_Configuration_D]]

=SBC Use Cases=
[[Toolpack:Tsbc Use Cases A|SBC Use Cases]] 
[[FreeSBC_Use_Cases|FreeSBC Use Cases]]

=SBC Tutorial Guide V3.0=
[[Tmedia Tsig Tdev Tutorial Guide v3.0|Version 3.0]]

[[Category:Revise on Major]]

FreeSBC: Multiple Domains/Hosted PBXs Configuration

2020-10-21T07:26:44Z

William Wong: /* SIP NAP */

=== '''''Applies to version: v3.0''''' ===

{{DISPLAYTITLE:FreeSBC:Multiple Domains/Hosted PBXs Configuration:Example}}
=Introduction=
FreeSBC Hosted PBX Example (see [[FreeSBC:Hosted PBX:Configuration A|FreeSBC:Hosted PBX]]) provides a step by step Hosted PBX Configuration of [[FreeSBC|FreeSBC]] systems, using the Web Portal configuration tool. What will be discussed here, with guide for example configuration, is when there are multiple domains per PBX or several PBXs with different domains. This multiple domains/PBXs configuration is basically the extension of what is configured in single domain/hosted PBX, that including the setting of different domains, followed by specific routing for different domain destinations.

=Hosted PBXs Example=

[[Image:Hosted_Multiple_PBX_topo_1.png]] 

From above, there will be several Hosted PBXs like Hosted PBX1, Hosted PBX2, Hosted PBX3, etc. And each Hosted PBX will use its own domain like example1.com for PBX1, example2.com for PBX2, and so on. This multiple domain scenario can also be applied to single Hosted PBX with different several domains that will be using the similar configuration.

With external path over the WAN, different phones belonging to different Hosted PBXs register to domain corresponding to the PBX/Registrar the phone is subscribing to.

Prerequisites and below configuration sections need to be done as listed in [[FreeSBC:Hosted PBX:Configuration A|single Hosted PBX configuration]] before proceeding:

* IP Network Configuration
* SIP Stack Configuration

=SIP NAP=

A Network Access Point or NAP represents the entry point to another network or destination peer (e.g. SIP proxy, SIP trunk, etc)

==SIP NAP Configuration for Local Area Network==

To create a new NAP:

1- Click '''NAPs''' in the navigation panel

[[Image:NAP_NavigationMenu.png]]

2- Click '''Create New NAP'''

[[Image:Create_New_NAP_SIP.png]]

3- Create the new NAP:

* Enter a '''name''' for the NAP
* Click '''Create'''

[[Image:SIP_Server_NAP_Create.png]]

4- Verify that the '''NAP was successfully created''' message appears

[[Image:Create_New_NAP_SIP2.png]]

5- Associate a SIP transport server with the new NAP:

* Select a '''SIP Transport Server''' from the '''Available''' list
* Click '''"<<"''' to associate the '''LAN_SIP_TS''' with the NAP

[[Image:SIP_Server_NAP_Create_1.png]]

6- Enter SIP Server proxy address:

[[Image:SIP_Server_NAP_Create_2.png]]

7- Associate a Port range with the new NAP:

* Select a '''port range''' from the '''Available''' list
* Click '''"<<"''' to associate '''LAN_Vlan:0''' Port range with the NAP

[[Image:SIP_Server_NAP_Create_3.png]]

9- Configure settings for the following parameter groups as required:
*Registration Parameters
*Authentication Parameters
*Network Address Translation
*SIP-I Parameters
*Advanced Parameters

*Click '''Save'''

[[Image:SIP_Server_NAP_Create_4.png]]

Here, we need to repeat above steps and create NAP for every Hosted PBX, for example, NAP_PBX1 for PBX1, NAP_PBX2 for PBX2, and so on.

==Access Control List==

FreeSBC will automatically create Access Control List for each NAP you created.

[[Image:Remote_Workers_Access_Control_List_1.png]]

If you double-click one of the created ACL, you will see FreeSBC only accept the calls if source IP matches. In this sample; the calls from 192.168.1.10 will accepted only.

[[Image:SIP_Access_Control_List_2.png]]

Rules for NAP_PBX1, NAP_PBX2, NAP_PBX3,.. will be created automatically in the ACL.

=SIP DOMAIN=

A SIP domain represents a grouping of devices (or users) that can communicate with one another.
You must configure SIP Registration Domain for your system. The first step in doing so is to create a SIP Domain:

==Create New SIP Registration Domain==

1- Click '''SIP Domain''' in the navigation panel

[[Image:SIP_Domain_NavigationMenu.png]]

2- Click '''Create New Domain'''

[[Image:Create_SIP_Domain.png]]

3- Create the new Domain:

* Enter a configuration '''Name''' for this domain.
* Enter a '''Domain Name''' for the SIP Registration Domain (the domain can be an FQDN or an IP address)
* Set the number '''Maximum Registered Users''' for this domain
* Set the Expires value used by SBC when the remote device doesn't supply one ('''Default Contact Expire''')
* Select '''Routing Method''' the system will use to route calls to registered users (if enabled in routing scripts).
** '''Register source''': Sends SIP Invite to the registering source IP address.
** '''Contact''': Sends SIP Invite to the 'contact' from the Register message.
* Set the '''Default Contact Expiration''', this value will be used when no ''Expires'' value is supplied by the user agent.
* Set the '''Minimum Contact Expiration''', this is the minimum ''Expires'' value that can be supplied by a user agent. Lower values will be rejected with a 423 'Interval too brief' response.
* Set the '''Maximum Contact Expiration''', this is the maximum ''Expires'' value that can be supplied by a user agent. The higher value is replaced by this parameter.
* Forwarding Parameters:
** Select the Registration '''Forwarding Mode''' to the registrar:
*** '''Contact Remapping''': Changes the user and the IP address.
*** '''Contact Passthrough''': Doesn't change anything. Enables devices to be contacted directly without going through the SBC.
** Set '''Minimum Registrar Expiration''', this is the minimum ''Expires'' value sent by the SBC to the registrar. If a user agent 'Expires' value is greater than this parameter, the SBC will do rate adaptation between the user agent and the Registrar.
** Set '''Maximum Pending Register Forward''', the maximum number of simultaneous pending register requests allowed for this domain. New REGISTER request is being refused passed this threshold.
** Set '''Maximum Simultaneous Register Forward''', the maximum number of simultaneous active register requests allowed for this domain. New REGISTER request is being refused passed this threshold.
* Click '''Create'''

[[Image:register_domain_1.png]]

4- Verify that the '''Sip registration domain was successfully created''' message appears

[[Image:register_domain_2.png]]
 

5- If your registrar has multiple domains, you need to create all the domains one by one.

Repeat above steps for every domain designated to every PBX, for example, example1.com for PBX1, example2.com for PBX2, and so on. This also applies to the case of multiple domains from the same PBX/Registrar as pointed out in step 5 above.

==Create New SIP Registration Registrar==

A SIP registrar represents a SIP endpoint that provides a location service.
You must configure SIP Registrar for your system. The first step in doing so is to select your SIP Domain:

1- Click on your domain in the '''SIP Domain List'''

[[Image:SIP_Domain_Select.png]]

2- Click '''Create New SIP Registration Registrar'''

[[Image:New_SIP_Registrar_Create.png]]

3- Create the new SIP Registration Registrar

* Enter a '''Name''' for the SIP Registration Registrar
* Select pre defined SIP Proxy NAP from drop-down menu
* Click '''Create'''

[[Image:SIP_Registrar_Select.png]]

4- Verify that the '''Sip registration registrar was successfully created''' message appears

[[Image:Create_New_SIP_Registrar2.png]]

==Associate a SIP Domain with the new NAP==
Associate a SIP Domain with the new NAP. If you have more then 1 registrar domain using the same registrar you can associate all of them with the NAP:

* Select a '''sip domain''' from the '''Available''' list
* Click '''"<<"''' to associate the SIP Domain with the NAP
[[Image:Associate_SIP_NAP.png]]

=Call Route=

You must set up call routing for your system. [[Call routing]] refers to the ability to route calls based on criteria such as origin, destination, time of day, service provider rates, and more.

==Route Configuration for Remote Phones to SIP Server==

1- Click '''Routes''' in the navigation panel

[[Image:CreateCallRoute_0.png]]

2- Click '''Create New Static Route'''

[[Image:CreateCallRoute_1.png]]

3- Create the new route:

* Enter a '''RoutesetName''' for the route
* Select '''SIP_REMOTE''', to match calls from Trunk NAP
* Select '''SIP_SERVER_NAP'''
* Click '''Create'''

[[Image:Route_Remote_to_Server.png]]

4- Verify that the '''" Route was successfully created"''' message appears and that the new route is listed in the Routeset list

[[Image:Route_Remote_to_Server_1.png]]

==Route Configuration for SIP Server to Remote Phones==

1- Click '''Routes''' in the navigation panel

[[Image:CreateCallRoute_0.png]]

2- Click '''Create New Static Route'''

[[Image:CreateCallRoute_1.png]]

3- Create the new route:

* Enter a '''RoutesetName''' for the route
* Select '''SIP_SERVER_NAP''', to match calls from Trunk NAP
* Select ''' (By registered user) '''
* Click '''Create'''

[[Image:Route_Server_to_Remote_2.png]]

4- Verify that the '''" Route was successfully created"''' message appears and that the new route is listed in the Routeset list

[[Image:Route_Server_to_Remote_3.png]]

=Activating the Configuration=

Changes made to the configuration of the FreeSBC units are stored in the OAM&P Configuration and Logging database. In order for changes to be used by the system, they must first be activated. This is done at the system level and accessed from the Navigation panel.

Check the following link for activating the configuration;

[[Toolpack:Activating_the_Configuration_D]]

=SBC Use Cases=
[[Toolpack:Tsbc Use Cases A|SBC Use Cases]] 
[[FreeSBC_Use_Cases|FreeSBC Use Cases]]

=SBC Tutorial Guide V3.0=
[[Tmedia Tsig Tdev Tutorial Guide v3.0|Version 3.0]]

[[Category:Revise on Major]]