Mitel Technical Configuration Notes

Mitel Technical Configuration Notes G12 Communications Configure MiVoice Office 250 for use with G12 Communications August 2015 SIP COE 15-494900398 Technical Configuration Notes 1 (877) 311-8750 sales@g12com.com Jackson St. #19390, Seattle, WA 98104

NOTICE The information contained in this document is believed to be accurate in all respects but is not warranted by Mitel Networks Corporation (MITEL ). The information is subject to change without notice and should not be construed in any way as a commitment by Mitel or any of its affiliates or subsidiaries. Mitel and its affiliates and subsidiaries assume no responsibility for any errors or omissions in this document. Revisions of this document or new editions of it may be issued to incorporate such changes. No part of this document can be reproduced or transmitted in any form or by any means - electronic or mechanical - for any purpose without written permission from Mitel Networks Corporation. TRADEMARKS Mitel is a trademark of Mitel Networks Corporation. Windows and Microsoft are trademarks of Microsoft Corporation. Other product names mentioned in this document may be trademarks of their respective companies and are hereby acknowledged. Mitel Technical Configuration Notes: Configure MiVoice Office for use with G12 Communications August 2015 15-4940-00398, Trademark of Mitel Networks Corporation Copyright 2015, Mitel Networks Corporation All rights reserved ii

Table of Contents OVERVIEW... 1 Interop History... 1 Interop Status... 1 Software & Hardware Setup... 1 Tested Features... 2 Device Limitations and Known Issues... 3 Network Topology... 4 CONFIGURATION NOTES... 5 MiVoice Office 250 Configuration Notes... 5 Network Requirements... 5 Assumptions for the MiVO-250 Programming... 5 Licensing and Option Selection SIP Licensing... 6 SIP Peer Trunk Groups... 8 Programming Configuration Folder... 9 Programming Trunk Group Configuration Folder... 13 IP Call Configuration Settings... 16 Programming IP and SIP Phone Sets... 18 NuPoint SIP Voice Mail Configuration... 24 Internal Voice Mailboxes... 25 MICOLLAB NUPOINT SETUP NOTES... 26 Active Configuration... 26 Active Line Groups... 26 Offline Configuration... 26 Edit Line Groups... 27 Network Element... 27 NuPoint Voice Mailbox Setup... 29 CONFIGURING MIVO-250 FOR USE WITH MICOLLAB AUDIO CONFERENCE. 30 Mitel 86xx Phones... 30 Add Mitel 86xx Phones... 30 Set Mitel 86xx phones MAC Address... 30 MiCollab Audio Conference Hunt Group... 31 Create Hunt Group for MiCollab AudioConf... 31 Program MiCollab AudioConf Hunt Group... 32 Add Hunt Group Members... 32 iii

Enable Socket... 33 MiCollab Server Manager Setup... 34 System Options... 34 SIP Server Configuration... 35 Create User... 35 MiCollab Audio Conference Setup... 36 Create New Conference... 36 Established Conference... 37 MIVOICE BORDER GATEWAY SETUP NOTES (OPTIONAL)... 38 Network Requirements... 38 Assumptions for MBG Configuration... 38 MiVoice Office 250 Setup... 38 SIP Trunk Setup... 39 iv

Overview This document provides a reference to Mitel Authorized Solutions providers for configuring the MiVoice Office (MiVO-250) 6.0 SP3 to connect to G12 Communications NetSapiens 12.23. The different devices can be configured in various configurations depending on your VoIP solution. This document covers only a basic setup with required option setup. Interop History Version Date Reason 1 August 25, 2015 Initial Interop with MiVoice Office 6.0 SP3 and G12 Communications NetSapiens 12.23 Interop Status The Interop of G12 Communications with MiVO-250 has been given a Certification status. This device will be included in the SIP CoE Reference Guide. The status the G12 Communications NetSapiens 12.23 achieved is: The most common certification which means the device/service has been tested and/or validated by the Mitel SIP CoE team. Product support will provide all necessary support related to the interop, but issues unique or specific to the 3rd party will be referred to the 3rd party as appropriate. Software & Hardware Setup The table below provides the hardware and software specifications used. Manufacturer Variant Software Version Mitel MiVoice Office 250 Mitel 5000 6.0 SP3 Mitel MiVoice Border Gateway MBG: 9.0.27.0 / MSL: 10.1.39.0 Mitel 5320, 5330 IP Sets Main Load Version: 06.03.00.12 Boot Load Version: 06.03.00.03 Aastra 6863i, 6865i SIP Sets Firmware: 4.0.0.1096 and 4.0.0.2031 G12 Communications NetSapiens 12.23 1

Tested Features This is an overview of the features tested during the Interop test cycle and not a detailed view of the test cases. Please see the 5000 SIP Trunk Side Interoperability Test Plan APTest 530 for detailed test cases and their result. Feature Feature Description Issues Basic Call Automatic Call Distribution NuPoint Voicemail Packetization Personal Ring Groups (PRG) Video Fax Making and receiving a call through SIP service provider and their PSTN gateway, call holding, transferring, conferencing, busy calls, long calls durations, variable codec. Making calls to an ACD environment with RAD treatments, Interflow and Overflow call scenarios and DTMF detection. Terminating calls to a NuPoint voicemail boxes and DTMF detection. Forcing MIVO-250 to stream RTP packets through its E2T card at different intervals, from 20ms to 30ms. Receiving calls through G12 Communications and their PSTN gateway to a personal ring group. Also moving calls to/from the prime member and group members. Making and receiving a call through G12 Communications with video capable devices. T.38 and G711 Fax calls. - No issues found - Issues found, cannot recommend to use - Issues found Not tested/applicable 2

Device Limitations and Known Issues This is a list of issues or not supported/applicable features when G12 Communications is connected to MiVO-250. Feature Registration and Authentication Loopback Call ACD only supports RAD and no MOH Outgoing Call with Incomplete Address Outgoing and Incoming calls to Outof Service Devices PRACK Video Problem Description Not supported. No authentication was required. Recommendation: Contact G12 Communication Support for further information SIP initiated loopback trunk call was being rejected by MiVO-250 when From header contained extension number of the original caller. It only worked when From header of the INVITE of the second leg of loopback call was either empty or containing a number that was not recognized by MiVO-250. Note that MINET initiated loopback trunk calls were routed correctly as per Trunk Group configuration. Recommendation: Contact Mitel Support for further information When ACD agent is busy, caller only hears RAD and no MOH Recommendation: Contact Mitel Support for further information G12 Communication SIP Peer sends 404 Not Found instead of expected 484 Address Incomplete response to initial INVITE. Recommendation: Contact G12 Communications Support for more details MiVO-250 does not support Out of Service device. MiVO-250 does not support PRACK (MN00528344) Recommendation: Contact Mitel Support for further information MiVO-250 and G12 Communications both do not support Video 3

Network Topology This diagram shows how the testing network is configured for reference. Aastra 6865i SIP Phone Mitel SIP Interoperability Network Configuration Aastra 6863i SIP Phone PSTN MiVO-250ice Office (MiVO-250) Trunk side MBG Server G12 Communications Netsapiens Figure 1 Network Topology 4

Configuration Notes This section is a description of how the SIP Interop network was configured. These notes provide a guideline as how a device can be configured in a customer environment and how the G12 Communications NetSapiens and MiVO-250 were configured in our test environment. Disclaimer: Although Mitel has attempted to setup the interop testing facility as closely as possible to a customer premise environment, implementation setup could be different onsite. YOU MUST EXERCISE YOUR OWN DUE DILIGENCE IN REVIEWING, planning, implementing, and testing a customer configuration. MiVoice Office 250 Configuration Notes The following information shows how to program a MiVO-250 6.0 SP3 to interconnect with the G12 Communications. Network Requirements There must be adequate bandwidth to support the VoIP network. As a guide, the Ethernet bandwidth is approx 85 Kb/s per G.711 voice session and 29 Kb/s per G.729 voice session (assumes 20ms packetization). As an example, for 20 simultaneous SIP sessions, the Ethernet bandwidth consumption will be approx 1.7 Mb/s for G.711 and 0.6Mb/s. Almost all Enterprise LAN networks can support this level of traffic without any special engineering. Please refer to the MiVB Engineering guidelines on the Mitel edocs Website (http://edocs.mitel.com) for further information. For high quality voice, the network connectivity must support a voice-quality grade of service (packet loss <1%, jitter < 30ms, one-way delay < 80ms). Assumptions for the MiVO-250 Programming The SIP signaling connection uses UDP on Port 5060. 5

Licensing and Option Selection SIP Licensing Ensure that MiVO-250 is equipped with enough SIP Trunking licenses for the connection to G12 Communications. This can be verified under the Software License form. Figure 2 Software License Form 6

To determine the number of licenses still available for use, select About Mitel Software License under Help option from the pull down menu. In below example, there are 10 SIP trunk remaining licenses available for use. Figure 3 Available Software Licenses 7

SIP Peer Trunk Groups To create a SIP Trunk Group, navigate to System > Devices and Feature Codes > SIP Peers. To add a new SIP Trunk Group, right click on the right pane and select Create SIP Trunk Group. You are automatically presented with the next available extension. Alternatively, could scroll down to select other extensions. By default one extension would be created. Click on the Description and Username fields of the created SIP Trunk Group and enter desired information. In below example, 92009 SIP Trunk group was created for G12 Communications. Figure 4 Creating SIP Trunk Group 8

Programming Configuration Folder Registration: If the SIP peer does not require registration, the fields in this folder do not need to be configured. The Enable Registration option is set to No by default and the remaining fields appear with a red X. Authentication: o Username: This field applies only if the SIP peer requires registration or call authentication. o Password: This field applies only if the SIP peer requires registration or call authentication. Keep-Alive: The Keep-Alive option keeps refreshing the NAT bindings for any Firewall/NAT in the path. It also helps in determining whether the SIP peer is reachable or not. NAT Settings: Specifies the NAT address type. The default is No NAT or SIP-Aware NAT, for systems that are using a SIP-aware firewall. If you are not using a SIP-aware firewall, you must change the setting to Non SIP-Aware NAT. Alternate IP/FQDN List: Some providers use multiple IP addresses to send SIP messages to the MiVO-250. You must add All IP addresses or FQDNs other than the primary IP/FQDN to the list for all calls to be successful. IP Address: Indicates the IP address of the SIP peer trunk group. Port Number: Indicates the port that the system listens on the system for SIP peer messages. The range is 0 65535. Fully Qualified Domain Name: Indicates the domain name of the SIP peer trunk group. Call Configuration: Clicking Call Configuration takes you to the Call Configuration folder under System > IP-Related Information > Call Configurations > $Call_Configuration_Number. Operating State: Indicates the operating state of the SIP peer. Maximum Number of Calls: Indicates the maximum number of concurrent calls that are permitted towards the SIP peer. DB Programming restricts this field based on the number of the SIP Trunks and SIP trunk licenses. This is a read-only field. It is updated as SIP peer trunks are created under Trunk Group Configuration > Trunks Use ITU-T E.164 Phone Number: If set to Yes, the MiVO-250 handles ITU-T E.164 formatted phone numbers (+1) as part of the incoming SIP INVITE messages from the SIP peer. Static Binding leave as default Yes, as it is the required value for SIP Trunk Groups Use Peer Address in From Header: set it to No. Setting it to Yes, would make the SIP initiated loopback calls not to route properly to Trunk Group configured destination. DTMF Decoding Payload: recommended to set to 101 from default 96. 9

Below is an example of configuring the SIP Trunk Group for G12 Communications sip peer. Figure 5 Configuration of SIP Trunk Group Figure 6 SIP Trunk Group > Configuration > Registrar 10

Figure 7 SIP Trunk Group > Configuration > Authentication Figure 8 SIP Trunk Group > Configuration > Keep Alive Figure 9 SIP Trunk Group > Configuration > NAT Settings 11

Figure 10 SIP Trunk Group > Configuration > Alternative IP/FQDN List Figure 11 SIP Trunk Group > Configuration > Route Sets 12

Programming Trunk Group Configuration Folder Incoming trunk calls could be routed in various ways to the destination endpoint. In our example, the Call Routing Table 13 is used to associate incoming calls to sets. Call Routing Tables Figure 12 SIP Trunk Group Configuration 13

File-Based MOH Figure 13 Call Routing Table 13 Below is where you find File-Based MOH files to provide audio for Music-On-Hold and other audio options in Trunk Group Configuration. Figure 14 File-Based MOH 14

Trunk The number of created trunks, corresponds to the Maximum Number of Calls (read-only) found under SIP Trunks Groups > $SIP_Trunk_Num > Configuration. To Create the SIP Peer Trunks, Double click Trunks Right click the right pane, and the select Create SIP Peer Trunk In Create SIP Peer Trunk Extension dialog box, select the extension number to be created. The recommended range is 94001-94999. Select the number of extensions you want to create in the number of Extensions field. If the system is set to have more than one extension, the new trunks are assigned sequentially to the next available numbers. Select OK. For the G12 Communications, 6 extensions were created. The number of SIP Peer Turnks is restricted by the number of available SIP Trunk licenses. Figure 15 Trunk Extensions 15

IP Call Configuration Settings Call configurations defines the settings that IP endpoints and gateways use when connected to calls. Multiple devices can be connected to a specific call configuration. By default, all IP devices are placed in Call Configuration1, which is programmable. You do not need to add SIP endpoints to Call Configurations since these devices negotiate call configurations before establishing a connection. You can program up to 25 different Call Configurations. Call Configuration 1 was used for our testing. NOTE: G.711 FAX passthrough mode is setup as per below. However, for T.38 FAX mode, the respective highlighted settings in blue must be set. Figure 16 Call Configuration 1 Settings 16

Call Configuration Extensions List of phones associated with Call Configuration 1. Figure 17 Call Configuration Extensions Call Configuration SIP Trunk Association List of SIP Trunk groups associated with Call Configuration 1. Figure 18 Call Configuration SIP Trunk Association 17

Programming IP and SIP Phone Sets Created IP Phones have extensions 1000, and 1082. The extension 1082 is associated with NuPoint voice mailbox. SIP Phones have extensions 1055 and 1056. Figure 19 Basic Test Phone Extensions 18

IP Phone 1000 Basic Configuration Associated Extensions Figure 20 IP Phone 1000 Basic Configuration Below is list of extensions associated to IP Phone extension 1000. Figure 21 Associated Extensions to IP Phone 1000 19

Forwarding Path Below find the added Forwarding Paths to the list. Path 10 is added to the list that forwards the call to Embedded Voicemail with application extension of 3. Figure 22 Forwarding Path 10 to Embedded Voicemail Figure 23 Forwarding Path 10 Configuration Figure 24 Forwarding Point for System Forwarding Path 10 20

IP Settings This folder displays below information for each IP device. Network Configuration Figure 25 IP Setting for IP device MAC Address of IP Phone is found under network configuration associated with IP settings for IP devices. Figure 26 Network Configuration Mailboxes Mailboxes associated with IP device with extension 1000. Figure 27 Mailboxes 21

IP Phone 1082 Basic Configuration Following section outlines programming changes for extension 1082, so that call would be forwarded to NuPoint Voicemail. Figure 28 Forwarding Path 9 to NuPoint Voicemail Figure 29 Forwarding Path 9 Configuration Figure 30 Forwarding Point for System Forwarding Path 9 22

NuPoint Mailbox When Mailboxes are properly associated with NuPoint MAS mailbox, Associated field would appear as Yes. Figure 31 Associated NuPoint Mailboxes to IP Phone Extension 23

NuPoint SIP Voice Mail Configuration MiVoice Office 250 is connected to NuPoint Voicemail as an application in virtaul Mitel Application Suite (vmas) server 192.168.101.178:5058 and Call Configuration 4. Note: port 5058 is required for NuPoint. Figure 32 NuPoint SIP Voice Mail Figure 33 SIP Voice Mail Pilot 24

Figure 34 SIP Voice Mail Mailboxes Internal Voice Mailboxes Figure 35 Internal Voice Mailboxes 25

MiCollab NuPoint Setup Notes In our setup, NuPoint Web console is an application blade in Mitel MiCollab. Active Configuration Active Line Groups To view the list of active NuPoint Line Groups, select Active Configuration > Line Groups. In example below, Line Group 5000-50 VM with IP address of 192.168.101.50 was created for use with MiVO-250. Notice that 5000-50 mwi application is DTMF to PBX Dialer. Figure 36 View Active NuPoint Line Groups Offline Configuration To add, edit or delete elements of active configuration, the changes must be performed in Offline mode. Select Edit Offline Configuration to enter the Offline configuration mode. To accept the changes select Commit and Exit. This would modify offline configuration. Select Activate Offline Configuration to push changes to Active configuration. 26

Edit Line Groups In Offline configuration mode, select Line Groups which has options to Add, Edit or Delete line groups. Network Element Figure 37 Edit NuPoint Line Groups The network element 5000-50 with SIP-Gateway type and IP address 192.168.101.50 was created for use with MiVO-250. To view list of created network elements, must be in Offline Configuration mode by selecting Edit Offline Configuration. Figure 38 View Network Elements 27

To edit an already created network element, double click on the NE name to enter edit mode. Figure 39 Edit Network Elements 28

NuPoint Voice Mailbox Setup NuPoint 1055 and 1082 associated voice mailbox was created for use with MiVO-250 corresponding extensions. Figure 40 NuPoint Mailboxes in Mitel MiCollab 29

Configuring MiVO-250 for use with MiCollab Audio Conference Mitel 86xx Phones This requires DB programming on MiVO-250. Add Mitel 86xx Phones Add 3 MiVO-250 Mitel 86xx phones 1042, 1043, and 1044. Currently, Mitel Minet phones are not supported. Figure 41 List of Mitel 86xx Phones Set Mitel 86xx phones MAC Address Ensure that Mitel 86xx phone extensions 1042, 1043, and 1044 are integrated as part of their respective MAC addresses as per following: FF:FF:FF:FF:10:42, FF:FF:FF:FF:10:43, and FF:FF:FF:FF:10:44. 30

Figure 42 MAC Address of Mitel 86xx Phone MiCollab Audio Conference Hunt Group Create Hunt Group for MiCollab AudioConf Click on right pane and select create Hunt Group. In our example, MiCollab-AudioConf Hunt Group 2004 was created. Figure 43 MiCollab AudioConf Hunt Group 31

Program MiCollab AudioConf Hunt Group Select 2004 Hunt group and program as outlined below Figure 44 MiCollab AudioConf Hunt Group Configuration Add Hunt Group Members Figure 45 MiCollab AudioConf Hunt Group Members 32

Enable Socket Enable special 86xx protocol OAI socket as outlined below. Figure 46 Enable OAI socket 33

MiCollab Server Manager Setup In our setup, Audio, Web and Video Conferencing is an application blade in Mitel MiCollab. System Options When in Audio, Web and Video Conferencing application, System Options is found under Configuration heading. Set the Platform field to Mitel 5000, and Save. Select Configuration > Start/Stop AWV, and perform a stop and start for changes to take place. Figure 47 System Options in AWC application 34

SIP Server Configuration Create User Figure 48 SIP Server Configuration In order to be able to login to../portal have to provision a user while still../server-manager. Select Provisioning > Add User Username: set to Corp email Password: set to Corp email password ID: set to any number Registered Phone: set to a registered IP or SIP device, for example: 1048 Select Create User for new user to be added. Figure 49 Add User to AWC MiCollab 35

MiCollab Audio Conference Setup Create New Conference On browser of your choice, enter <vmas IP address>/portal url to login MiCollab End User portal. Use Username and password of the user created in../server_manager (previous section). In Instant Conference section, dial number or extension of a participant to create a conference. Figure 50 AWC Home page 36

Established Conference When conference is established, Access Code for leader and participants is generated. Use these access codes for incoming calls to join the conference. Use Add a new participant to send an outgoing conference invitation. Figure 51 AWC Established Conference 37

MiVoice Border Gateway Setup Notes (Optional) The following steps show how to program the MiVoice Border Gateway (MBG) server to allow connections between the MiVO-250 and SIP Peer. Network Requirements Please refer to the Multi-Protocol Border Gateway Engineering guidelines for further information. Assumptions for MBG Configuration MiVO-250 configuration completed as per instructions in previous section. The SIP signaling connection between the MiVO-250 and MBG server uses UDP on Port 5060. MBG server installed and configured for SIP clients support. MiVoice Office 250 Setup To program a MiVO-250 into MiVoice Border Gateway (MBG), Login to Server Manager of MBG Select MiVoice Border Gateway under Applications Select Service Configuration tab Click on ICPs Select Add ICP Enter a name for MiVO-250; example: 5000_50 Enter the IP address of MiVO-250 Select the Type of MiVO-250, example: MiVoice Office Figure 52 Configuration - ICP Setup 38

SIP Trunk Setup To program a G12 Communications SIP trunking into MiVoice Border Gateway (MBG), Under the Service Configuration tab of MBG, click on SIP Trunking Select + and Enter the SIP trunk s details as shown. Name: Enter the trunk name; for example: G12_Comm Remote trunk endpoint address: Enter the public IP address or FQDN of the provider s switch or gateway. This address should be provided to you by the provider. Local/Remote RTP framesize (ms): Enter the packetization rate you want to set on this trunk. PRACK: Use Disabled, as MiVO-250 does not support PRACK Routing rule one: Allows routing of any digits to the selected MiVO-250 The rest of the settings are optional and could be configured if required Click Save Figure 53 Services - SIP Trunking setup 39

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