Programming Access4 SIP Lines on Avaya IP Office – SBC Redundancy

Refer to Access4 Sip Trunk Service Guide. Excerpt from guide as below, explaining SBC Redundancy

1.3. SBC Redundancy

We use geographically dispersed SBCs to provide a redundant and a reliable SIP service. You need to

configure your PABX in a certain way to take advantage of our redundant SBCs.

You have been provided with the addresses of two SBCs. One SBC is in VIC and the other is in QLD,

you should always use VIC SBC as the primary and only failover to the QLD SBC in case if the primary is unavailable or unreachable.

Please note: you cannot register the same pilot user at multiple SBCs simultaneously. Although the

registration will be successful it may cause unpredictable results including active call dropouts or denial of service for new calls.

There are two different ways to achieve redundancy.

First one is to create two pilot users under the same trunk and register them through both SBCs

simultaneously, while first pilot registers with the primary SBC, the second goes to the secondary. This is the most reliable design of SIP trunk solutions.

The other way to achieve redundancy is to fail over registration to the secondary SBC only on case if

the first one is unavailable. Although it is a resilient architecture that allows to minimise downtime in case of the primary SBC is unavailable, is takes much longer time to failover that the dual registration method therefore there may be a period of service unavailability before PABX reregisters through another SBC, while with the dual registration failover will happen almost instantaneously.

1.3.1. Simultaneous registration through two SBCs

Redundancy is achieved by the creation of multiple pilot users that are registered individually

through different SBCs. Inbound calls from the PABX will be accepted through any of the individually registered trunks, outbound calls to the PABX can use different routing policies. For SBC redundancy it is recommended to use the “Overflow” policy available in SASBOSS™. When the policy is set to “Overflow”, Access4 will try to deliver a call to the first pilot user registration address, then fail over to the next, there are other policies available that can be used mainly when customers have multiple clustered PABX’s to increase the resilience of the solution, those are “Ordered Load Balancing”, “Least Idle”, “Most Idle”. Each individual pilot user cannot exceed the number of channels set per each sip-trunk and the aggregated number of calls per all pilot users cannot exceed the number of channels provisioned.

Configuring Avaya IPO System

Select System, LAN1 (or LAN2 depending on which LAN the SIP will be registering via), VoIP tab

 

Select System, LAN1, VoIP tab

Enable SIP Trunks

Enable RTCP Monitoring

Set Scope RTP-RTCP

Enable Initial Keepalives

Set Periodic Timeout to 5 seconds

Select System, LAN1, Network Topology tab

Set STUN Server Address: 0.0.0.0

Set Firewall/NAT Type: One-To-One NAT

Set Public IP Address: as the public IP address on the network

NOTE: in this document we are not using the Network Topology for the SIP Trunk, however some SIP trunks might require specific STUN servers to be used

DNS

Set the Primary and Secondary DNS 

NOTE: in this example we have used google DNS 8.8.8.8 and the router IP 192.168.86.1

The DNS is used to resolve the SIP ITSP Domain Name to the correct public IP address

Licenses

Confirm the SIP trunk licenses are valid and the instances/sessions

NOTE: with IP Office Server Edition, check the Remote Server tab and set the SIP Trunk Sessions to the amount of SIP Trunk sessions that will be used on this system

The reason for this option is because in a scenario where there is a IPOSE and additional expansion systems connected to the IPOSE, the licenses are applied to the IPOSE only

The expansion systems then pull down the licenses required from the IPOSE. In each expansion system you can then set the SIP Trunk Sessions required

EXAMPLE: IPOSE has 30 SIP Trunk Channel licenses, The SIP line on the IPOSE will be using 20 channels and the SIP line on the Expansion system will be using 10 channels. The SIP Trunk Sessions would be set accordingly on each system to reserve enough licenses for each

VCM Requirements

VCM Channels are required to allow SIP calls, each SIP call uses a VCM channel

The VCM Channels on an IP500v2 are provided on a physical base card (see below)

ATM Combo or BRI Combo base cards have 10 VCM channels

VCM32 base cards have 32 VCM channels

VCM64 base cards have 64 VCM channels

The VCM Channels on an IPOSE are provided as part of the system (no physical cards)

Open System Status and select Resources and check the VCM number of VCM channels

IP Routes

Confirm there is a default route programmed in the system to route all traffic to the gateway on the network

In the example below, our default route means all traffic is routing out via the LAN1 port to the customers data router (192.168.86.1) on the network

IP Address: 0.0.0.0 / IP Mask: 0.0.0.0 / Gateway IP Address: 192.168.86.1 / Destination: LAN1

NOTE: If the SIP is going to be connected to LAN2 (and the customer data network is connected to LAN1) then an additional IP Route is required to route the SIP traffic only, out via the LAN2 port

Access 4 use two proxy addresses (see below)

In the example below, an additional route has been created to route all traffic for the SIP IP address 45.64.148.232 and 45.64.151.232 out via the LAN2 port to the SIP router (192.168.43.254)

SIP - SIP Line

Create a new SIP Line

Note: This will be the first of two SIP trunks

Enter a Line Number (17)

Enter the ITSP Domain Name provided by the SIP carrier (or IP Address)

The two options for “In Service” and “Check OOS” are both checked by default.

When the “In Service” field is not selected, the SIP trunk is unregistered and not available to incoming and outgoing call

When the “Check OOS” field is enabled, the system will regularly check if the trunk is in service using the Session Timers methods listed below

Checking that SIP trunks are in service ensures that outgoing call routing is not delayed waiting for response on a SIP trunk that is not currently usable

For UDP and TCP trunks, OPTIONS messages are regularly sent. If no reply to an OPTIONS message is received the trunk is taken out of service.

For trunks using DNS, if the IP address is not resolved or the DNS resolution has expired, the trunk is taken out of service

SIP – SIP Transport

Enter the ITSP Proxy Address provided by the SIP carrier (or IP Address)

In Access4 setup we will setup two SIP trunks (for failover)

Set the ITSP Proxy Address of the first SIP trunk as tkreg1.a4uc.com.au

Set the Protocol as required

Set the Send Port and Listen Port

Set explicit DNS if required by SIP carrier to be different to system DNS

Separate Registrar allows the SIP registrar address to be specified if it is different from that of the SIP proxy. The address can be specified as an IP address or DNS name.

A proxy server is considered Active once the system has received a response to an INVITE, REGISTER or OPTIONS

SIP – SIP VoIP

Codecs can be selected and placed in order, depending on the codecs supported by the SIP carrier. G.711 ALAW and G.729 are commonly used in Australia

Enable Re-Invite Supported to allow change in the characteristics of the session. For example, when the target of an incoming call or a transfer does not support the codec originally negotiated on the trunk. To utilize conference calling and call forwarding over SIP, RE-INVITE Supported must be activated

Enable PRACK which supports Provisional Reliable Acknowledgement (PRACK) on SIP trunks. Enable this parameter when you want to ensure that provisional responses, such as announcement messages, have been delivered. Provisional responses provide information on the progress of the request that is in process

Fax Transport Support: T38

DTMF Support: RFC2833

SIP – SIP Credentials

Add the second SIP registration credentials provided by Access4

Enter the User Name: Primary telephone number

Enter Authentication Name: Primary telephone number

Enter Contact: Primary telephone number

Enter and Confirm a Password:

Set Expiry time: 1 Minute

Registration Required: Enabled

SIP – SIP Advanced

Association Method is set to: By Source IP Address

Call Routing Method: Request URI

Use PAI for Privacy: Enabled

Use Domain for PAI: Enabled

Send From In Clear: Enabled

NOTE: settings in here might need to be changed depending on the SIP carrier

SIP – Call Details

If users require the ability to display their own DDI or another number on the SIP trunk for outbound calls, that is not the registration number. Then two SIP URI entries are required

The First entry will have the Incoming Group set to 17 and the Outgoing Group set to 18

Set Credentials as the one created and the Max sessions to match the SIP trunk license quantity

Set the Local URI Display and Content to Auto

Set the Contact Display and Content to Auto

Enable the P Asserted ID, Set the Contact Display and Content to Auto

The Second entry will have the Incoming Group set to 18 and the Outgoing Group set to 17

Set Credentials as the one created and the Max sessions to match the SIP trunk license quantity

Set the Local URI Display and Content to Internal Data

Set the Contact Display and Content to Internal Data

Enable the P Asserted ID, Set the Contact Display to Internal Data (Set Content to Registration Number)

If you require the original caller’s number to display on calls inbound that are then forwarded out to an external number

Enable Diversion Header, Set the Contact Display and Content to Use Internal Data

SIP - SIP Line

Create a new SIP Line

Note: This will be the second of two SIP trunks

Enter a Line Number (117)

Enter the ITSP Domain Name provided by the SIP carrier (or IP Address)

The two options for “In Service” and “Check OOS” are both checked by default.

When the “In Service” field is not selected, the SIP trunk is unregistered and not available to incoming and outgoing call

When the “Check OOS” field is enabled, the system will regularly check if the trunk is in service using the Session Timers methods listed below

Checking that SIP trunks are in service ensures that outgoing call routing is not delayed waiting for response on a SIP trunk that is not currently usable

For UDP and TCP trunks, OPTIONS messages are regularly sent. If no reply to an OPTIONS message is received the trunk is taken out of service.

For trunks using DNS, if the IP address is not resolved or the DNS resolution has expired, the trunk is taken out of service

SIP – SIP Transport

Enter the ITSP Proxy Address provided by the SIP carrier (or IP Address)

In Access4 setup we will setup two SIP trunks (for failover)

Set the ITSP Proxy Address of the first SIP trunk as tkreg2.a4uc.com.au

Set the Protocol as required

Set the Send Port and Listen Port

Set explicit DNS if required by SIP carrier to be different to system DNS

Separate Registrar allows the SIP registrar address to be specified if it is different from that of the SIP proxy. The address can be specified as an IP address or DNS name.

A proxy server is considered Active once the system has received a response to an INVITE, REGISTER or OPTIONS

SIP – SIP VoIP

Codecs can be selected and placed in order, depending on the codecs supported by the SIP carrier. G.711 ALAW and G.729 are commonly used in Australia

Enable Re-Invite Supported to allow change in the characteristics of the session. For example, when the target of an incoming call or a transfer does not support the codec originally negotiated on the trunk. To utilize conference calling and call forwarding over SIP, RE-INVITE Supported must be activated

Enable PRACK which supports Provisional Reliable Acknowledgement (PRACK) on SIP trunks. Enable this parameter when you want to ensure that provisional responses, such as announcement messages, have been delivered. Provisional responses provide information on the progress of the request that is in process

Fax Transport Support: T38

DTMF Support: RFC2833

SIP – SIP Credentials

Add the Second SIP registration credentials provided by Access4

Enter the User Name: Primary telephone number

Enter Authentication Name: Primary telephone number

Enter Contact: Primary telephone number

Enter and Confirm a Password:

Set Expiry time: 1 Minutes

Registration Required: Enabled

SIP – SIP Advanced

Association Method is set to: By Source IP Address

Call Routing Method: Request URI

Use PAI for Privacy: Enabled

Use Domain for PAI: Enabled

Send From In Clear: Enabled

NOTE: settings in here might need to be changed depending on the SIP carrier

SIP – Call Details

If users require the ability to display their own DDI or another number on the SIP trunk for outbound calls, that is not the registration number. Then two SIP URI entries are required

The First entry will have the Incoming Group set to 17 and the Outgoing Group set to 118

Set Credentials as the one created and the Max sessions to match the SIP trunk license quantity

Set the Local URI Display and Content to Auto

Set the Contact Display and Content to Auto

Enable the P Asserted ID, Set the Contact Display and Content to Auto

The Second entry will have the Incoming Group set to 118 and the Outgoing Group set to 117

Set Credentials as the one created and the Max sessions to match the SIP trunk license quantity

Set the Local URI Display and Content to Internal Data

Set the Contact Display and Content to Internal Data

Enable the P Asserted ID, Set the Contact Display to Internal Data (Set Content to Registration Number)

If you require the original caller’s number to display on calls inbound that are then forwarded out to an external number

Enable Diversion Header, Set the Contact Display and Content to Use Internal Data

User

Each user on the system has a SIP tab that becomes available once a SIP line has been added, when using internal data in the SIP trunk programming the details in the SIP tab will be used for outbound calls

In the SIP Name and Contact, set the number to be displayed when calling out

Group

Each user on the system has a SIP tab that becomes available once a SIP line has been added, when using internal data in the SIP trunk programming the details in the SIP tab will be used for outbound calls

In the SIP Name and Contact, set the number to be displayed when calling out

Voicemail

In System and Voicemail tab, the SIP settings should also be programmed with the SIP number details

Incoming Call Route

Add a new entry

Set the Line Group ID as 17

Leave the Incoming Number blank

Select Destinations tab and set the destination as required (extension, Group or Auto Attendant)

This blank entry will route any incoming call that doesn’t match any of the other Incoming Numbers

Add a new entry

Set the Line Group ID as 17

Set the Incoming Number as the full number

Select Destinations tab and set the destination as required (extension, Group or Auto Attendant)

If using Voicemail Pro, to route a call to a Module (Auto Attendant in Voicemail Pro) then the destination will be entered as VM:AutoAttendant (replace AutoAttendant with the name of the module)

NOTE: If the destination is an AutoAttendant then it is recommended to set a Fallback Extension as a group or user, in case the voicemail is down the call can route to the fallback destination

Add any additional DID/DDI number into the Incoming Call Route

To add a range of numbers, select Tools, MSN Configuration and configure as required

Shortcodes

Check to confirm how the calls are made outbound by searching in the system Shortcodes for the Shortcode with Feature Dial

As default the IP Office system doesn’t require any prefix such as 0 to use an external line

There should be an entry

Code: ?

Feature: Dial

Telephone Number: . (dot)

Line Group ID: 50 Main

When a number is dialed, and the system checks the number and find no internal user or group matching the number dialed. The number is sent to the Line Group ID 

The default is 50: Main (which is an ARS)

A Dail Emergency Shortcode should also be created

Example below

ARS

Create a new ARS

Enter the Route Name as Access4 failover

Set the Dial Delay to 3 seconds

Select Add, to create a new code entry

Set the below

Code: N;

Feature: Dial

Telephone Number: N

Line Group ID: 117

Select the Main ARS

Delete the existing Code ? . Dial

Select Add, to create a new code entry

Set the below

Code: N;

Feature: Dial

Telephone Number: N

Line Group ID: 17

Set the Alternate Route Wait Time to 5 Seconds

Set the Alternate Route as the Access4r failover ARS created

If the SIP trunk does not recognise local numbers when dialed then additional codes will need to be added to prefix local numbers with the area code

In the example, any local number beginning with 8 (8 digits long) will be prefixed with 03

Code: 8XXXXXXX

Feature: Dial

Telephone Number: 03. (dot after the 03)

Line Group ID: 17

These entries would need to be repeated in the second ARS

Using the Line Group ID: 117

Testing Calls

View the SIP trunk in System Status

Confirm the SIP is registered and the public IP address is resolving correctly

Make inbound and outbound calls and watch in System Status to confirm working correctly

Use Monitor and System Status to troubleshoot if the SIP is not working

Scenario 1A - Requirement to display different CLI to the Internal Data (User SIP tab CLI) Using Short Code/Button Programmed

In the SIP Line, add a new SIP URI entry

This entry will have the Incoming Group set to 19 and the Outgoing Group set to 19

Set Credentials as the one created and the Max sessions to match the SIP trunk license quantity

Set the Local URI Display and Content to the specific CLI to display on outbound calls

Set the Contact Display and Content to the specific CLI to display on outbound calls

Enable the P Asserted ID, Set the Contact Display to the specific CLI to display on outbound calls (Set Content to the Registration Number)

Create a new ARS

Duplicate the Main ARS and update the named (example Display CLI)

Update the Line Group ID in this new ARS to use 19

Create a new System Short Code (this will display #7 on the phone/softphone screen when used)

Example Code #7N, with Feature Dial, Telephone Number N, Line Group ID (select the new ARS)

Any extension on the system can now dial #7 as a prefix, followed by the number they want to call, it will route this call out via the new ARS and display the CLI configured (not the CLI in the SIP User tab)

Create a new Button (this will display #7 on the phone/softphone screen when used)

In the User Button Programming

Create new button with the Label Display CLI, Action Dial, Action Data #7N

The User can now press this button which will prefix the #7, they can then dial the number they want to call, it will route this call out via the new ARS and display the CLI configured (not the CLI in the SIP User tab)

Scenario 1B – Requirement to Withhold CLI Using Short Code/Button Programmed

In the SIP Line, add a new SIP URI entry

This entry will have the Incoming Group set to 21 and the Outgoing Group set to 21

Set Credentials as the one created and the Max sessions to match the SIP trunk license quantity

Set the Local URI Display to anonymous (Set the Content to the Registration Number)

Set the Contact Display and Content to anonymous

Enable the P Asserted ID, Set the Contact Display to anonymous (Set Content to the Registration Number)

Note the Field Meanings are all set to Explicit

Create a new ARS

Duplicate the Main ARS and update the named (example Withhold CLI)

Update the Line Group ID in this new ARS to use 21

Create a new System Short Code (this will display #6 on the phone/softphone screen when used)

Example Code #7N, with Feature Dial, Telephone Number N, Line Group ID (select the new ARS)

Any extension on the system can now dial #6 as a prefix, followed by the number they want to call, it will route this call out via the new ARS and Withhold the CLI

Create a new Button (this will display #6 on the phone/softphone screen when used)

In the User Button Programming

Create new button with the Label Display CLI, Action Dial, Action Data #6N

The User can now press this button which will prefix the #6, they can then dial the number they want to call, it will route this call out via the new ARS and Withhold the CLI

Scenario 2A - Requirement to display different CLI to the Internal Data (User SIP tab CLI) Using Line Key

In the SIP Line, Enable SIP Line Appearance

Add a new SIP Line Appearance

Set the Incoming Group to 0 and the Outgoing Group to 17

Set Credentials as the one created; Set the Max sessions set to 1

Set the Local URI Display and Content to the specific CLI to display on outbound calls

Set the Contact Display and Content to the specific CLI to display on outbound calls

Set the Contact Display and Content to the specific CLI to display on outbound calls

Button Programming

In the User Button Programming

Create new button with the Label Line, Action Line Appearance, Action Data (select the Line appearance ID number assigned (example above 705) set to No Ring

The User can now press this button, they can then dial the number they want to call, it will route this call and display the CLI configured (not the CLI in the SIP User tab)

Scenario 2B - Requirement to Withhold CLI Using Line Key

In the SIP Line, Enable SIP Line Appearance

Add a new SIP Line Appearance

This entry will have the Incoming Group set to 0 and the Outgoing Group set to 17

Set Credentials as the one created; Set the Max sessions set to 1

Set the Local URI Display to anonymous (Set the Content to the Registration Number)

Set the Contact Display and Content to anonymous

Enable the P Asserted ID, Set the Contact Display to anonymous (Set Content to the Registration Number)

Note the Field Meanings are all set to Explicit

Button Programming

In the User Button Programming

Create new button with the Label Line, Action Line Appearance, Action Data (select the Line appearance ID number assigned (example above 705) set to No Ring

The User can now press this button, they can then dial the number they want to call, it will route this call and display the CLI configured (not the CLI in the SIP User tab)