TECHNICAL NOTE Author: Adam Wells Date: June 6th, 2018

Alloy Computer Products Pty Ltd ABN 41 006 507 473

4/585 Blackburn Road Notting Hill 3168

Victoria, Australia Telephone: 03 8562 9000 Facsimile: 03 8562 9099

HOW TO CONFIGURE ALLOYVOICE SIP TRUNKS ON

GRANDSTREAM UCM 6XXX SERIES

1. Introduction

This Technical note will go through information on how to setup AlloyVoice on Grandstream UCM PBX, as well as general information on the SIP Protocol.

SIP trunks are a VoIP service that can be provided from an ITSP (Internet Telephony Service Provider) to extend telephony features beyond IP-PBX local area. SIP trunks can carry voice calls, video calls, multimedia conferences, and other SIP-based, real-time communications services.

TECHNICAL NOTE Author: Adam Wells Date: June 6th, 2018

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Contents

1. Introduction ................................................................................................................................................... 1

2. Requirements ................................................................................................................................................ 3

3. Grandstream UCM Architecture Examples ................................................................................................... 3

3.1 SIP ALG ..................................................................................................................................................... 4

3.2 Bandwidth Requirements ........................................................................................................................ 4

4. Connecting to the Grandstream UCM6XXX Series PBX. ................................................................................ 5

5. Logging into the Web Management Console ................................................................................................ 6

Configure SIP Trunk ....................................................................................................................................... 8

6.1. Add New SIP Trunks ................................................................................................................................ 8

Configure Outbound Rules ........................................................................................................................ 9

Configure Inbound Routes ....................................................................................................................... 10

Direct Outward Dialing ............................................................................................................................ 11

Applying the Configuration Changes ...................................................................................................... 13

SIP Trunk Status ...................................................................................................................................... 14

TECHNICAL NOTE Author: Adam Wells Date: June 6th, 2018

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2. Requirements

Before configuring the Grandstream UCM PBX, the following minimum requirements should be considered.

Prerequisites

- Internet/Network Services - It is also highly recommended that you have a static IP address. If your external IP changes

intermittently, inbound calls will fail. - A firewall/router/NAT device that supports static port mapping. - Open the following ports to allow Grandstream UCM to communicate with the Alloy Voice SIP

Trunk : o Port 5060 (UDP/TCP) for SIP communications o Port 10000-20000 (UDP) for RTP/RTCP communications.

3. Grandstream UCM Architecture Examples

Network Scenario - On-Premises Deployment

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3.1 SIP ALG

To maximize your chances of success, make sure you choose a device that does not implement a SIP Helper or SIP ALG (Application Layer Gateway), or choose a device on which SIP ALG can be disabled.

3.2 Bandwidth Requirements

VoIP is in real time, so it does place a demand on your Internet connection. Example: Call Using G711 Codec Each RTP packet contains 20ms of audio (typical). Each 20ms of audio requires 160 bytes. Each second of audio will require 50 packets, each containing an audio payload of 160 bytes. Before being transmitted over the network the IP packet will contain: 160 bytes for the audio payload 12 bytes for the RTP header 8 bytes for the UDP header 20 bytes for the IP header = for a total of 200 bytes per packet If the transmission medium is Ethernet, the IP packet is encapsulated in an Ethernet Frame which adds an 18-byte header, for a total of 218 bytes per frame * 50 packets per second * 8 bits per byte. This equates to 87,200 bits per second or 87.2 kbps (RTP Call Only), to include message overhead, add +5%.

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4. Connecting to the Grandstream UCM6XXX Series PBX.

Please follow the following steps to connect the Grandstream UCM6XXX Series Phone System.

Connect the WAN Port of the Grandstream UCM6XXX Series Phone System to the Local Area Network.

o Note: For Initial Set-up be sure to utilise the WAN port to your network to receive an IP Address via DHCP. Alternately you can connect to the LAN port and receive DHCP from the UCM, however you will not be able to register a SIP trunk via the LAN side.

Power-on the Grandstream UCM6XXX Series Phone System.

From the LCD and navigation keys, press "OK" button and "DOWN" button to view the IP address on the Grandstream UCM6XXX Series Phone System (e.g. UCM6102).

Connect the administration console computer to the same local area network to which the

Grandstream UCM6XXX Series Phone System WAN port is connected to.

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5. Logging into the Web Management Console The Grandstream UCM6XXX Series Phone System provides a web console to monitor and configure the Grandstream UCM6XXX Series Phone System parameters.

Please follow the following steps to login to the Grandstream UCM6XXX.

- Open a web browser such as Google Chrome - Enter in the IP Address of the UCM, EG 192.168.10.250. https://192.168.10.250 this will redirect

you to port 8089 by default. - Enter in the username and password. If you have not logged in or changed the password before

this will be located underneath the UCM on a label.

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After successfully logging in, you will be presented with the Grandstream Dashboards which provides an overview of the system such as resources, extensions, memory etc.

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Configure SIP Trunk

In this section we discuss the configuration of the Alloy Voice SIP VoIP Trunk through the Grandstream UCM web console.

6.1. Add New SIP Trunks

To add a new SIP trunk on the Grandstream UCM, follow the steps below.

- Click the Extension / Trunk menu from the left hand side - Select the VoIP Trunks sub menu - Under VoIP Trunks, Click the +Create New SIP Trunk button

Enter in the following details under the Create New SIP Trunk

- Type: Register SIP Trunk Provider Name: AlloyVoice

- Host Name: alloyvoice.com.au - Keep Original CID: Enable this option - Keep Trunk CID: Leave this option unticked - NAT: Leave this option unticked - Allow outgoing calls if registration fails: Leave this option unticked - Username: Enter in the username of the AlloyVoice Trunk - Password: Enter the password of the AlloyVoice Trunk - Auth ID: Enter in the username of the AlloyVoice Trunk - Click Save

This will bring you back to the VoIP Trunks section. Select Edit on the AlloyVoice Trunk

Under the Basic Settings Tab;

- From Domain: alloyvoice.com.au - From User: Enter the Username for the trunk here eg 0312345678

Under the Advanced Settings Tab;

- Codec Preference: Select Codec Preference to PCMA - Send PAI Header: Tick Send PAI Header - Outbound Proxy Support: Tick Enable Outbound Proxy Support - Outbound Proxy: enter sbc02.alloyvoice.com.au - The Maximum Number of Call Lines: Enter in the number of concurrent calls allowed on

your AlloyVoice Plan

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Configure Outbound Rules

In this section we discuss the configuration of enabling outgoing calls for extensions on the Grandstream UCM to route through the AlloyVoice SIP Trunk.

Add Outbound Rules

To add an outbound rule, follow the steps below.

- Select the Extension / Trunk menu on the left hand side - Select Outbound Routes - Select +Add

Enter in the following details

- Calling Rule Name: Enter the Calling rule name – EG AlloyVoice - Pattern: Enter pattern _X. (X – Any digit between 0-9, “.” Wildcard, Match one or more characters)

Definition Of Pattern Options: o All patterns are prefixed with the "_". o Special characters: o X: Any Digit from 0-9. o Z: Any Digit from 1-9. o N: Any Digit from 2-9. o ".": Wildcard. Match one or more characters. o "!": Wildcard. Match zero or more characters immediately.

- Use Trunk: Select the configured trunk that was setup in the previous step. EG AlloyVoice - Privilege level: Enter the privilege level that you wish to allow. If you set this to International it will

allow international calls to be routed via the AlloyVoice Trunk. - Click Save

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Configure Inbound Routes

In this section we will go through adding inbound routes and DID’s associated AlloyVoice Trunk.

- Select the Extension / Trunk menu on the left hand side - Select Inbound Routes - Select +Add

Please follow the following steps to configure the Inbound Route settings. The configuration discusses configuring the example DID number (e.g. 0312345678) to the end-user extension (e.g. 1002). Please refer to your Grandstream Administration guide for further instructions on inbound route options.

- Calling Rule Name: Enter the Calling rule name – EG Main Inbound Number - Pattern: Enter pattern _0312345678 - Definition Of Pattern Options:

o All patterns are prefixed with the "_". o Special characters: o X: Any Digit from 0-9. o Z: Any Digit from 1-9. o N: Any Digit from 2-9. o ".": Wildcard. Match one or more characters. o "!": Wildcard. Match zero or more characters immediately.

- Default Destination (Type): Select the type of default destination Type - E.G extension - Default Destination: Select the destination extension E.G 1002 - Click Save

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Direct Outward Dialing

In this section we discuss the basic configuration of Direct Outward Dialing (DOD) which enables an internal end-user/extension to present their DID number when making and outgoing call.

Example Configuration:

The configuration example is for the DID number (e.g. 0312345678) which will be configured as the DOD number assigned to the end-user/extension (e.g. 1002).

When end-user/extension (e.g. 1002) makes and outgoing it will present the number (e.g. 0312345678)

Add Direct Outward Dialing:

Please follow the following steps to configure a DID to an end-user/extension which is presented when the appropriate end/extension makes and outgoing calls.

- Select the Extension / Trunk menu on the left hand side - Select VoIP Trunks - Click the DOD icon on the right hand side of the AlloyVoice Trunk

Enter in the following details

- DOD Number: The telephone number you wish to display on the outbound call - Available: The Extension list you can choose from,

Select the extension you wish to use this DID when dialing out by pressing the -> Symbol - Selected: The extension you have selected to display this number on outbound dialing - Click Save

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Applying the Configuration Changes

After you have made any changes on the Grandstream UCM you must select Apply Changes at the top of the Web UI. If you do not click this the changes will not take effect.

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SIP Trunk Status

To view the status of the SIP Trunk, this is displayed on the dashboard of the Grandstream Web Interface under the Trunks section. You can also view Extension information here under the PBX Status.