Avaya IPO Programming Optus SIP

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 using the Network Topology for the SIP Trunk, as Optus SIP requires to use the Network Topology

DNS

Set the Primary and Secondary DNS

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.

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.

In the example below, an additional route has been created to route all traffic for the SIP IP address 210.49.5.9 out via the LAN2 port to the SIP router (192.168.43.254)

SIP – SIP Line

Create a new SIP Line

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 message 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)

Set the Protocol as required.

Set Use Network Topology Info to LAN1 (or LAN2 if SIP is routing via LAN2)

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 new SIP registration credentials provided by SIP carrier.

Enter the User Name:

Enter Authentication Name:

Enter Contact:

Enter and Confirm a Password:

Set Expiry time: 60 Minutes

Registration Required: Enabled

NOTE: Optus provided the User Name in uppercase, but we had to use lower case to register

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 the pilot/main (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 Auto

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 to use the SIP Line 17

Example below

ARS

Delete the default Code in the ARS

Add an entry.

Code: N;

Feature: Dial

Telephone Number: N

Line Group ID: 17

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

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.