audiocodes mediant 1000 configuration guide · this document shows how to deploy audiocodes'...

Mediant™ 1000B

Mediant™ 800

SIP Protocol

Deployment Note Deploying AudioCodes Mediant™ 1000B / 800 with AltiGen® MAX Communications Server 7.0

December 2014

Document #: LTRT-

Deployment Note Contents

AltiGen MAX Communications Server 3 February 2014

Table of Contents 1 Introduction ......................................................................................................... 7

1.1 Intended Audience ................................................................................................. 7 1.2 About AudioCodes' Mediant Gateway .................................................................... 7 1.3 Related Documentation .......................................................................................... 7

2 Hardware .............................................................................................................. 9

2.1 Network Topology .................................................................................................. 9 2.2 Package Contents .................................................................................................. 9 2.3 Front Panel ............................................................................................................. 9 2.4 Rear Panel ........................................................................................................... 10

3 Setting up the Hardware ................................................................................... 11

3.1 Step 1: Ground the Chassis .................................................................................. 11 3.2 Step 2: Insert the T1/E1 Module ........................................................................... 11 3.3 Step 3: Connect the Gateway to a Provisioning Computer ................................... 12

4 Configuring the AudioCodes Gateway ............................................................ 13

4.1 Opening the Web Interface ................................................................................... 13 4.2 Step 1: Disable the DHCP Server ......................................................................... 14 4.3 Step 2: Set VoIP IP Parameters ........................................................................... 15 4.4 Step 3: Set Syslog Parameters ............................................................................. 16 4.5 Step 4: Configure PSTN Trace Parameters .......................................................... 17 4.6 Step 5: Set SIP Advanced Parameters ................................................................. 18 4.7 Step 6: Set TDM Bus Parameters ......................................................................... 19 4.8 Step 7: Set PSTN Trunk Parameters .................................................................... 20 4.9 Step 8: Set IPMedia Parameters .......................................................................... 21 4.10 Step 9: Set SIP General Parameters .................................................................... 22 4.11 Step 10: Set Proxy and Registration Parameters .................................................. 23 4.12 Step 11: Set VoIP Coders Parameters ................................................................. 24 4.13 Step 12: Set VoIP Trunk Group Parameters ......................................................... 25 4.14 Step 13: Set VoIP Trunk Group Settings .............................................................. 26 4.15 Step 14: Configure the IP to Trunk Group Routing ............................................... 27 4.16 Step 15: Set Digital Gateway Parameters ............................................................. 28 4.17 Step 16: Configure Outbound Trunk Routing ........................................................ 29 4.18 Step 17: Configure Inbound Trunk Routing ........................................................... 30

5 Configuring the MAX Communications Server............................................... 31

5.1 Step 1: Configure the SIP Trunk in Enterprise Manager ....................................... 31 5.2 Step 2: Configure the SIP Trunk in MaxAdministrator ........................................... 35 5.3 Troubleshooting .................................................................................................... 40

6 Verifying Interoperability .................................................................................. 42

Appendix A: Analog Configuration ....................................................................... 44

Front Panel .................................................................................................................... 44 Step A1: Add a New Rule To the Trunk Group Table .................................................... 45 Step A2: Configure Analog Gateway Parameters .......................................................... 46

AltiGen MAX Communications Server

AudioCodes Mediant 1000B and Mediant 800 4 Document #: LTRT-39265

Step A3: Configure Automatic Dialing ............................................................................ 47 Step A4: Verify Interoperability ...................................................................................... 48

Appendix B: SIP Configuration .............................................................................. 49

Step B1: Change the Default Proxy Option to No .......................................................... 49 Step B2: Enable IP-to-IP ................................................................................................ 50 Step B3: Increase Media Channels ............................................................................... 51 Step B4: Remove the Default Proxy Set ........................................................................ 52 Step B5: Configure IP Group 1 for MaxCS..................................................................... 53 Step B6: Configure IP Group 2 for the SIP Trunk........................................................... 54 Step B7: Configure Proxy Set 1 for MaxCS ................................................................... 55 Step B8: Configure Proxy Set 2 for the SIP Trunk ......................................................... 56 Step B9: Configure Coder Group 1 for MaxCS .............................................................. 57 Step B10: Configure Coder Group 2 for the SIP Trunk .................................................. 58 Step B11: Configure IP Profile 1 for MaxCS .................................................................. 59 Step B12: Configure IP Profile 2 for the SIP Trunk ........................................................ 60 Step B13: Remove Plus Sign in Destination Phone Number ......................................... 61 Step B14: Remove Plus Sign in the Source Phone Number .......................................... 62 Step B15: Configure Outbound IP Routing Rules .......................................................... 63 Step B16: Configure Inbound IP Routing Rules ............................................................. 64 Step B17: Verify Interoperability .................................................................................... 65

Appendix C: AudioCodes ini File .......................................................................... 66

AltiGen MAX Communications Server Notices


Notice This document shows how to deploy AudioCodes' Mediant 800 / Mediant 1000B with the AltiGen MAX Communications Server in your enterprise. Information contained in this document is believed to be accurate and reliable at the time of printing. However, due to ongoing product improvements and revisions, AudioCodes cannot guarantee accuracy of printed material after the Date Published nor can it accept responsibility for errors or omissions. Before consulting this document, check the corresponding Release Notes regarding feature preconditions and/or specific support in this release. In cases where there are discrepancies between this document and the Release Notes, the information in the Release Notes supersedes that in this document. Updates to this document and other documents as well as software files can be downloaded by registered customers at http://www.audiocodes.com/downloads.

© Copyright 2014 AudioCodes Ltd. All rights reserved. This document is subject to change without notice.

Date Published: June-2014

Trademarks AudioCodes, AC, AudioCoded, Ardito, CTI2, CTI², CTI Squared, HD VoIP, HD VoIP Sounds Better, InTouch, IPmedia, Mediant, MediaPack, NetCoder, Netrake, Nuera, Open Solutions Network, OSN, Stretto, TrunkPack, VMAS, VoicePacketizer, VoIPerfect, VoIPerfectHD, What’s Inside Matters, Your Gateway To VoIP and 3GX are trademarks or registered trademarks of AudioCodes Limited. All other products or trademarks are property of their respective owners. Product specifications are subject to change without notice.

WEEE EU Directive Pursuant to the WEEE EU Directive, electronic and electrical waste must not be disposed of with unsorted waste. Please contact your local recycling authority for disposal of this product.

Customer Support Customer technical support and service are generally provided by AudioCodes’ Distributors, Partners, and Resellers from whom the product was purchased. For technical support for products purchased directly from AudioCodes, or for customers subscribed to AudioCodes Customer Technical Support (ACTS), contact [email protected].

AudioCodes Customer Support Center Americas: +1-732-652-1085 or +1-800-735-4588 Elsewhere: 800-735-2244 Or 972-3-976-4343 http://www.crm.audiocodes.com

Corporate website http://www.audiocodes.com

Support http://www.audiocodes.com/support

iSupport (on-line ticket management) https://crm.audiocodes.com/OA_HTML/jtflogin.jsp

Product Documentation and Software http://www.audiocodes.com/downloads

Training http://www.audiocodes.com/technical-training

Professional Services http://www.audiocodes.com/Professional-services

http://www.audiocodes.com/downloads

mailto:[email protected]

http://www.audiocodes.com/

http://www.audiocodes.com/support

https://crm.audiocodes.com/OA_HTML/jtflogin.jsp

http://www.audiocodes.com/downloads

http://www.audiocodes.com/technical-training

http://www.audiocodes.com/Professional-services



Abbreviations and Terminology Each abbreviation, unless widely used, is spelled out in full when first used.

Note: Throughout this manual, unless otherwise specified, the term device refers to the AudioCodes Mediant 800 and Mediant 1000B.

AltiGen MAX Communications Server 1. Introduction


1 Introduction This Deployment Note shows how to deploy the hardware and configure the gateway to interoperate with the AltiGen MAX Communications Server Release 7.0 Premise or Private Cloud. This Note covers how to configure T1/E1 connectivity. Appendix A gives instructions for analog configuration; Appendix B covers the configuration for SIP trunks. The Note does not cover the gateway's other functionalities. See Related Documentation, for a list of related documentation items. The instructions in this Note apply to a deployment with gateway software version 6.40A.039.008 or later.

IMPORTANT: AltiGen does not provide general configuration support for AudioCodes products. A support agreement with AudioCodes is required.

1.1 Intended Audience The Note is intended for AltiGen dealers, system administrators and network administrators who will deploy the AudioCodes Mediant 800 or 1000B gateway to interoperate with the AltiGen MAX Communication Server release 7.0.

1.2 About AudioCodes' Mediant Gateway AudioCodes' Mediant gateway provides a bridge between the AltiGen MAX Communications Server and the PSTN network. The gateway transparently manages the protocol translation and interworking between the two systems.

1.3 Related Documentation Related documentation is: Mediant 1000B Quick Guide Mediant 1000B SIP User’s Manual (Mediant 1000 MSBG User's Manual Ver. 6.4) Mediant 1000B SIP Installation Manual Release Notes Configuring Syslog Technical Note (Configuring Syslog Technical Note Ver. 6.2) Errata-Addendum for SIP CPE Documentation Restoring Factory Defaults Technical Note (Restoring Factory Defaults Configuration

Note Ver. 6.4) BootP Technical Note Customer Case Reporting Procedures SIP CPE Product Reference Manual CPE SIP Troubleshooting Guide



AudioCodes' documentation is available for download from the AudioCodes web site, http://www.audiocodes.com after registering free of charge. In the web site, click the Services tab and choose the Product Documentation & Software menu.


AltiGen MAX Communications Server 2. Hardware


2 Hardware This section itemizes the hardware components and shows how to connect the gateway to a provisioning computer.

2.1 Network Topology This illustration shows the gateway and server deployed in the network:

2.2 Package Contents You should receive one large box containing the gateway and the power cord. If the T1/E1 component wasn't preinstalled in the chassis, it will be provided in a separate box. You must provide an Ethernet cable and a T1 cable. Ensure that your Mediant 1000B Gateway package includes the following items:

• Four anti-slide bumpers for desktop installation • One 19-inch rack mounting kit (two flanges and six screws) • RS-232 serial cable adaptor for serial communication between the Mediant

1000B PSTN Gateway functionality (flat connector) and a pc (red-yellow DB-9 connector)

• Two mounting brackets for 19-inch rack mounting • One AC power cord

It's recommended to retain all AudioCodes receipts and documents. If any items are missing or damaged, contact your AudioCodes sales representative.

2.3 Front Panel This shows and explains the front panel:

1 T1 Telephony Ports

2 CRMX or CMX

Ports for CMX module, the CPU for VoIP gateway functionality, with 3 LAN and 1 WAN port. You will use one of the LAN ports. On (Green) – Ethernet link established Flashing (Green) – Data received or transmitted Off – No Ethernet link

3 Power 1 Spare power supply slot.

4 Power 2 Main power supply slot.

5 Fan Extractable fan tray.

T1/E1 SIP Trunk Gateway

PSTN MAXCS

1 2

3 4 5



2.4 Rear Panel This shows and explains the rear panel:

1 Protective earthing (grounding) screw.

2 ESD Electrostatic Discharge (ESD) socket.

3 100-240V~2A 50-60 Hz Main AC power supply outlet.

4 100-240V~2A 50-60 Hz Optional, redundant AC power supply outlet.

5 OSN3 OSN3 module (not required for AltiGen deployment; may or may not be included).

6 |O|O| Serial interface port (RJ-45).

7

Gigabit Ethernet interface port (RJ-45) - automatic detection and switching between 10Base-T, 100Base-TX, and 1000Base-T data transmissions (Auto-Negotiation), as well as auto-wire switching for crossed cables (Auto-MDI/X).

8 USB 2.0 port.

9 HDMX Hard disk drive module for OSN3 platform.

AltiGen MAX Communications Server 3. Setting up the Hardware


3 Setting up the Hardware This section shows how to set up the hardware components.

3.1 Step 1: Ground the Chassis The chassis must be permanently connected to earth (ground) using an equipment-earthing conductor.

To ground the chassis:

1. Connect an electrically earthed strap of 16 AWG wire (minimum) to the chassis' earthing screw (located on the left end of the rear panel) using the supplied washer.

2. Connect the other end of the strap to a protective earthing in accordance with the regulations enforced in the country of installation.

3.2 Step 2: Insert the T1/E1 Module If the T1/E1 module was not preinstalled, you must insert it into the chassis.

Note: Before inserting the module into the chassis, remove the clear tape which covers the module's gold edge and which protects the module’s BUS line. If you do not remove it, the module will not function and you may cause irreversible damage to the chassis.

Remove the tape



To insert the T1/E1 module into the chassis:

1. Remove the tape from the edge of the module. 2. Use a Phillips screwdriver to remove the black metal cover plate protecting the module

slot on the front panel. 3. Insert the module into the first available slot (slots 1-5); align the module with the rails

in the slot in the following module orientation depending on whether you are inserting it in the top or bottom-row slots: • Top-row slots: Ensure that the module is orientated so that the port number labels

are located at the bottom of the module's front panel. This orientation is considered face up.

• Bottom-row slots: Ensure that the module is orientated so that the port number labels are located at the top of the module's front panel. This orientation is considered face down.

4. Push the module into the slot and press on it firmly to ensure it is fully inserted. 5. Use a flathead screwdriver to tighten the module's mounting pins.

3.3 Step 3: Connect the Gateway to a Provisioning Computer You must configure the gateway with a direct connection to a provisioning computer before you connect the gateway to the LAN. Do not connect the gateway to your corporate network until you have configured it. The gateway comes with a DHCP server that must be disabled, or else network connectivity problems may occur.

To connect the gateway to a provisioning computer:

1. On the rear panel, connect the left power socket to an electrical outlet using the supplied AC power cord. The power LED on the front panel of the power supply should be green. If the LED is off, there may be a power supply problem.

2. Use an RJ-45 Ethernet cable to connect the network port on the provisioning PC directly to the LAN port on the gateway.

3. Make sure that this PC is configured to automatically obtain IP addresses. The gateway’s embedded DHCP server (enabled by default) allocates an IP address to the PC when you connected the two devices. If this allocation does not automatically occur, manually assign the IP address 192.168.0.100.

4. Open a Web browser. In the URL field, enter the gateway default VoIP / Management LAN IP address (i.e., 192.168.0.2).

5. When the Login page opens, use the default, case-sensitive user name (Admin) and password (Admin) and click OK; the Web-based interface opens on the Home page.

AltiGen MAX Communications Server 4. Configuring the AudioCodes Gateway


4 Configuring the AudioCodes Gateway This section shows how to customize AudioCodes' software settings for interoperability with the MAX Communications Server.

4.1 Opening the Web Interface The Web interface's Home page shows components status. Click any component in the graphic representation of the gateway to display operational information or additional configuration options.

Important! In order to configure the gateway via this tool, you must have the menu set to

“Full.” If you leave the menu set to “Basic,” you will not see some of the menu choices and some options on various pages will not appear. Be especially careful to check this menu setting after you have logged out and logged back in again – make sure the application has not automatically reverted to “Basic” menu mode. In addition, if you notice that you are working in “Basic” menu mode and then choose “Full,” it is best to leave the page you are on and then re-open it. In some cases, switching to Full menu mode will not automatically refresh the current page.

Burn button: Use it to write configura-tion changes to flash memory

Navigation pane

Choose 'Full' instead of ‘Basic’

Shows real-time status of components (click a component for details on its status)

Device Actions: Use this menu to reset the gateway and load or save configuration files



4.2 Step 1: Disable the DHCP Server Disable the DHCP server to avoid conflict with another DHCP server during initial configuration.

To disable:

1. Choose Data > Data Services > DHCP Server.

2. If the WAN Ethernet service is shown as Enabled, click the pencil icon on that row to edit the settings.

3. From the menu, select Disabled and click OK.



4.3 Step 2: Set VoIP IP Parameters Set the gateway IP address information for your system. 1. Choose VoIP > Network > IP Interfaces Table. 2. Click the 0 (zero) Index button; an Edit button appears. Click it and configure network

settings corresponding to your network IP address scheme: • Set ‘IP Address’ to the AudioCodes gateway’s IP address • Set ‘Prefix length’ to the subnet mask length in CIDR notation • Set ‘Gateway’ to the default gateway’s IP address • (Optional) DNS: Enter the primary DNS server IP address (and the secondary if

relevant) • Leave the remaining options set to their default settings

3. Click Apply then Done and then Burn; the gateway reboots and starts using the new

IP address. Maintain the cabled connection between the gateway LAN port and the provisioning computer.

Note: Reset the gateway for parameters marked to take effect, e.g., 'WAN Interface Name' in the screen above.



4.4 Step 3: Set Syslog Parameters Syslog traces are important for troubleshooting. AudioCodes recommends that you set up a Syslog server.

To set up a Syslog server:

1. In the left navigation pane, click Configuration. 2. Choose System > Syslog Settings 3. Set these parameters:

• Set 'Enable Syslog' to Enable • Set 'Syslog Server IP Address' and 'Syslog Server Port' • Set 'Debug level' to 5 • Make sure all parameters in the lowermost pane are selected

4. Click Submit.



4.5 Step 4: Configure PSTN Trace Parameters Next, configure settings for the trace log.

To configure the trace log:

1. In the browser window, enter the network IP address of the AudioCodes gateway appended with this string: /AdminPage. Refer to the next figure for an example.

2. In the left pane, click ‘ini Parameters.’

3. In the ‘Parameter Name’ field, enter TRACELEVEL. 4. In the ‘Enter Value’ field, enter 1. 5. Click Apply New Value. 6. In the ‘Parameter Name’ field, enter PSTNRESERVED3. 7. In the ‘Enter Value’ field, enter 8. 8. Click Apply New Value. 9. To return to the AudioCodes program, in the left pane click Back to Main.



4.6 Step 5: Set SIP Advanced Parameters CDR (Call Data Record) information is helpful when interpreting Syslog traces. Set the parameters shown in this subsection to provide robust CDR report details. 1. Choose VoIP > SIP Definitions > Advanced Parameters 2. Set these parameters:

• Set 'Disconnect on Broken Connection' to No • Set 'CDR Server IP Address' to the same address as the Syslog server • Set 'CDR Report Level' to Start & End Connect Call

3. Click Submit.



4.7 Step 6: Set TDM Bus Parameters The PSTN gateway's TDM bus must be configured. The parameters shown below configure the signal encoding used between the AudioCodes gateway and the AltiGen MAX Communications Server.

To set TDM bus parameters:

1. Choose VoIP > TDM > TDM Bus Settings 2. Set this parameter:

• Set 'PCM Law Select' to MuLaw

3. Click Submit.



4.8 Step 7: Set PSTN Trunk Parameters This step shows how to set PSTN trunk parameters.

To set PSTN trunk parameters:

1. Choose VoIP > PSTN > Trunk Settings. 2. Set these parameters:

• Set 'Protocol Type' to T1 NI2 ISDN • Set 'Clock Master' to Recovered • For 'Line Code', select B8ZS (bipolar 8-zero substitution) for T1 trunks • Set 'Framing Method' to T1 FRAMING ESF CRC6 • Set 'ISDN Termination Side' to User side

3. Click Submit. 4. Click the Burn button to save the configuration changes to flash memory.

Note: If these parameters have previously been set, you may need to click Deactivate before you can adjust them.



4.9 Step 8: Set IPMedia Parameters This step shows how to set IPMedia parameters.

To set IPMedia parameters:

1. Choose VoIP > Media > IPMedia Settings. 2. Set these (optional) parameters:

• Set 'IPMedia Detectors' to Enable • Set 'Enable AGC' to Enable

3. Click Submit. 4. Click Burn.



4.10 Step 9: Set SIP General Parameters This step shows how to configure the gateway SIP parameters.

To configure the gateway SIP parameters:

1. Choose VoIP > SIP Definitions > General Parameters. 2. Set the following parameters:

• Set ‘NAT IP Address’ to the MaxCS server's public IP address • Set 'Enable Early Media' to Enable • (Recommended) Set 'Session Expires Time' to 300 (or your preferred duration) • Make sure that 'SIP Transport Type' is set to UDP

3. Click Submit.



4.11 Step 10: Set Proxy and Registration Parameters This step shows how to configure the address (IP address or FQDN) of the MAX CS SIP trunk that communicates with the gateway. The PSTN gateway forwards all calls from the PSTN to the SIP trunk using this address. The address is configured in the PSTN gateway as a proxy server, i.e., the SIP trunk acts as a proxy server (without registration) for the PSTN gateway.

To set proxy and registration parameters:

1. Choose VoIP > SIP Definitions > Proxy & Registration. 2. Set 'Use Default Proxy' to Yes. 3. Click the arrow below the 'Use Default Proxy' parameter to open the Proxy Set Table.

4. On row 1, set 'Proxy Address' to the MaxCS server's IP address and port 5060. 5. Set 'Transport Type' to UDP.




4.12 Step 11: Set VoIP Coders Parameters This step shows how to set the VoIP coder parameters. The PSTN gateway communicates with the SIP trunk using either the G.711 A-law or G.711 µ-law (Mu-Law) voice coder. In addition, silence suppression can be enabled per coder.

To configure the VoIP coder parameters:

1. Choose VoIP > Coders and Profiles > Coders. 2. For 'Coder Name', select the codec type. 3. For each coder name, choose whether you want Silence Suppression. Silence

suppression saves network bandwidth (not critical for LAN) but may cause small artifacts when voice reestablishes. AudioCodes recommends setting 'Silence Suppression' to Disabled.




4.13 Step 12: Set VoIP Trunk Group Parameters This step shows how to set the Trunk Group parameters. To enable trunks, you need to assign them to trunk groups. Trunk groups are local entities that the SIP trunk will use for call routing.

To set the VoIP Trunk Group parameters:

1. Choose VoIP > GW and IP to IP > Trunk Group > Trunk Group. 2. For each trunk group, indicate these parameters:

• Set 'Module’ to the module number and type (e.g., PRI) on which the trunks are located

• Set 'From Trunk' (first available) and 'To Trunk' (last) to the physical trunk range • For 'Channels,' specify the B-channels (e.g., 1-24) that you want to enable • Set 'Phone number' to the number for the first channel; the rest will be assigned

sequentially • Set 'Trunk group ID' to the ID group




4.14 Step 13: Set VoIP Trunk Group Settings This step shows how to set the VoIP Trunk Group Settings.

To set the VoIP Trunk Group Settings:

1. Choose VoIP > GW and IP to IP > Trunk Group> Trunk Group Settings. 2. For each trunk group, set 'Channel Select Mode' to Descending and click Submit.

Note: Descending assumes that the PSTN uses Ascending mode. Set the channel mode that is the opposite of the PSTN mode to avoid possible glare conditions (when the same b-channels are allocated for both incoming and outgoing calls).

See the AudioCodes User's Manual for a full explanation of all supported channel select modes.



4.15 Step 14: Configure the IP to Trunk Group Routing After enabling the trunks and assigning them to trunk groups, you must configure how the PSTN gateway selects trunk channels belonging to a group for receiving calls.

To configure the IP to Trunk Group routing:

1. Choose VoIP > GW and IP to IP > Routing > IP to Trunk Group Routing. 2. Set 'Source IP Address' to the IP address of the MaxCS Server. 3. Click Submit.

Do not put an asterisk in the “Source IP Address” field. If you do, anyone can use the Mediant 1000 gateway to make PSTN calls.



4.16 Step 15: Set Digital Gateway Parameters Here's how to set the Digital Gateway Parameters.

To set the Digital Gateway Parameters:

1. Choose VoIP > GW and IP to IP > Digital Gateway > Digital Gateway Parameters. 2. Set 'B-Channel Negotiation' to Preferred.

3. Click Submit.



4.17 Step 16: Configure Outbound Trunk Routing This section shows how to configure outbound call routing. The Outbound IP Routing table lets you configure up to 180 Tel-to-IP or outbound IP call routing rules. You will create a rule to indicate that all outbound calls should route through MaxCS.

To configure outbound trunk routing:

1. Choose VOIP > GW and IP to IP > Routing > Tel to IP Routing. 2. For routing index 1 (the first row), specify that all outgoing calls are included in this

rule. Do this by entering the asterisk character (*) in the first five fields, as shown in the next figure.

3. For ‘Dest. IP Address’ enter the IP address of the MaxCS server, to indicate that all outbound calls route through MaxCS.

4. Click Submit.



4.18 Step 17: Configure Inbound Trunk Routing This section shows how to configure inbound call routing. The Inbound IP Routing table lets you configure up to 20 inbound call routing rules. You will create a rule to indicate that all inbound calls should be routed through MaxCS.

To configure inbound routing:

1. Choose VOIP > GW and IP to IP > Routing > IP to Trunk Group Routing. 2. In the first row, specify that all incoming calls are included in this rule. Do this by

entering the asterisk character (*) in the first four fields, as shown in the next figure. 3. In the same row, for ‘Source IP Address,’ enter the IP address of the MaxCS server, to

indicate that all incoming calls route through MaxCS. 4. Click Submit.

5. On the toolbar, from the Device Actions menu, choose Reset; the reset takes

approximately four minutes to perform; while it occurs, unplug the Ethernet cable from the provisioning PC and plug it into your local network; when the gateway restarts, it will have the new IP address.

6. After the gateway restarts, plug your T1 cable into the one of the T1 ports on the front panel.

AltiGen MAX Communications Server 5. Configuring the MAX Communications Server


5 Configuring the MAX Communications Server This section shows how to configure the MAX Communications Server for interoperability with AudioCodes' gateway. Note: When AudioCodes is used as a PSTN trunking gateway, select your codec

carefully based upon your anticipated number of simultaneous calls and your internet/WAN bandwidth. Calls (both incoming and outgoing) will be routed from the premise AudioCodes Mediant 1000 to the Private Cloud MaxCS server’s SIP trunks, then will follow the MaxCS configured call routing rules. Since the call originates from the local PRI and most likely will terminate to a local IP extension, this doubles the amount of bandwidth used when compared with using AltiGen SIP trunks.

From an administrator’s point of view, a SIP trunk is similar to a Triton analog trunk. After it's up and running, you can assign a trunk access code to it. You can also mix SIP trunks with analog trunks, T1 trunks or PRI trunks in the Out Call Routing table. SIP trunks can also receive caller name and caller ID.

5.1 Step 1: Configure the SIP Trunk in Enterprise Manager This section shows the steps to take to configure the SIP trunk within the Enterprise Manager.

To configure the SIP trunk in Enterprise Manager:

1. In MaxAdministrator, open Enterprise Manager by choosing VoIP > Enterprise Network Management.

2. In Enterprise Manager, click the Codec button at the top. 3. In the lower corner, click Add to add a new codec profile.



4. Name this new codec profile AudioCodes SIP and click OK.

5. Select the correct codec from the drop-down list. AudioCodes supports both g.711

μLaw and g.729 codecs. If the MAX and the Mediant gateway reside on the same site, AudioCodes recommends that you use g.711. If you select g.729, make sure that your system has sufficient VoIP codec resources.

6. Set 'DTMF Delivery' to RFC 2833. 7. Set 'SIP Early Media' to Enable. 8. Click Apply.



9. Click the Servers button. 10. Click the IP Codec tab.

11. In the IP Codec pane, click Add.



12. Enter the gateway IP address for both the 'From' and 'To' IP addresses. 13. In the 'Codec' drop-down list, select AudioCodes SIP.

14. Click OK.



5.2 Step 2: Configure the SIP Trunk in MaxAdministrator This section shows the steps to take to configure and enable the SIP trunk within MaxAdministrator.

To configure the SIP trunk in MaxAdministrator:

1. In MaxAdministrator, double-click the SIPSP board in the Boards panel.

2. In the Board Configuration window, double-click SIP trunk.

3. In the SIP Signaling Channel Configuration window, click SIP Trunk Configuration.



4. In the SIP Trunk Configuration window, click SIP Trunk Profile.

5. In the SIP Trunk Profile window, click Add to add a new profile.



6. Name this new profile AudioCodes and click OK. 7. Set the 'SIP Protocol Field' for the SIP Calling Number to P-Asserted-Identity. 8. Set the options for the carrier’s ability to accept transmitted caller ID based on the

restrictions provided by the PRI carrier. In this example, 'Carrier can only accept Calling Number with minimum' is set to 10 digits.

9. In the 'Use this as Calling Number if the Carrier cannot accept configured numbers' box, enter your main AudioCodes number.

10. Select Send Caller Name to send the extension’s name in the outbound call. Note that many remote systems may not receive or display this information.

11. Make sure that the 'Enable Standard Record-Route Header' check box is NOT checked.

12. Select To Header as the 'Incoming DID Number Field'. Click OK. 13. In the SIP Trunk Configuration window, select the first SIP Trunk channel to configure

and click Edit.



14. Enter this information:

• For 'SIP Server IP Address' enter the gateway IP address. • For 'User Name', enter AudioCodes • Enter AltiGen as the password • Enter a Domain name • Set the SIP Register Period to 0 • For the 'SIP Trunk Profile', select AudioCodes • Set both the 'SIP Source Port' and the 'SIP Destination Port' to 5060 • Check the 'Enable Channel' check box

15. Click OK to save the changes. 16. Use the Copy To… feature to copy the configuration for this channel to any other

channels that are needed. 17. Click OK again to close the SIP Trunk Configuration window; a pop-up warns you to

restart switching; this does not apply to changes made to SIP Trunks and can be safely ignored. Click OK to proceed.



18. Click OK to close the Board Configuration window.



5.3 Troubleshooting If the SIP gateway requires a domain name in addition to the SIP Server IP address, make sure the domain can be pinged and resolved to the address specified in SIP Server IP Address in the MAX Communications Server command console. If it cannot be pinged, you can add the DNS to the MAX Communications Server's DNS server or you can add the DNS mapping to the host file in C:\WINDOWS\system32\drivers\ folder and then use the IPCONFIG /flushdns command to refresh the DNS mapping. When the DNS is added correctly, you should be able to ping the domain without rebooting the MAX Communications Server.



Reader's Notes



6 Verifying Interoperability Verify that the gateway is correctly configured for your environment.

To verify that the gateway is correctly configured for your environment:

1. Log in to the gateway using your browser, changing your PC’s IP address to one that is able to communicate with the gateway’s new IP address, if required.

2. Click the Home button; the LAN icon on the Home page (under CRMX) should be green, as well as the trunk.

3. Under 'General Information', confirm that the IP address, subnet mask, and default gateway parameters are correct.

4. Place test calls - using hold and transfer features as well - to ensure that calls are handled correctly.

This completes the configuration process for T1 connectivity. For instructions on configuring analog connectivity, please turn to Appendix A on page 44. For instructions on configuring SIP Trunk connectivity, turn to Appendix B on page 49.

AltiGen MAX Communications Server 6. Verifying Interoperability


Reader's Notes



Appendix A: Analog Configuration In order to configure analog support, you must first complete the T1/E1 steps provided earlier in this document.

Note: For ground-start signaling, the FXO G module must be is installed (instead of the regular FXO module) in the Mediant 1000B chassis. The Mediant 800B does not support ground-start signaling. Refer to the AudioCodes Hardware Installation Manual for your specific product.

Front Panel There are specific ports for analog connections.

1 T1 The T1 port.

2 FXO The FXO module includes up to four analog RJ-11 ports.

3 FXS The FXS module includes up to four analog RJ-11 ports (not shown in this figure).

1 2 3



Step A1: Add a New Rule To the Trunk Group Table To configure analog routing, you will add another rule to the Trunk Group table.

Before you begin configuring the gateway for analog routing, you must connect it to a provisioning computer and log into the AudioCodes web page. See page 12 for instructions.

To configure the trunk group table settings:

1. Choose VoIP > GW and IP to IP > Trunk Group > Trunk Group. 2. The first row, Group Index 1, contains the rule you created earlier, on page 25. Set up

a new rule on the second row: • Set 'Module type' to Module 2 FXO • Set 'Channel' to the B-channels (e.g., 1-24) that you want to enable (in the next

figure, channels 1-4 were selected) • Set 'Phone number' to the number for the first channel; the rest will be assigned

sequentially • Set 'Trunk group ID' to the ID for the group




Step A2: Configure Analog Gateway Parameters The FXO Settings page allows you to configure the device's specific FXO parameters.

To configure analog gateway parameters:

1. Choose VoIP > GW and IP to IP > Analog Gateway > FXO Settings. 2. Set ‘Dialing Mode’ to One Stage. 3. Click Submit.



Step A3: Configure Automatic Dialing The Automatic Dialing page allows you to define a telephone number that is automatically dialed when an FXS or FXO port goes off-hook.

To configure the automatic dialing number:

1. Choose VoIP > GW and IP to IP > Analog Gateway > Automatic Dialing. 2. For the Module 2 Port 1 FXO row, enter the phone number you want the system to

dial when the port goes off-hook. Some firms choose to use the Auto Attendant number for this entry.

3. Click Submit.

4. On the toolbar, from the Device Actions menu, choose Reset; the reset takes approximately four minutes to perform; while it occurs, unplug the Ethernet cable from the provisioning computer and plug it into your local network; when the gateway restarts, it will have the new IP address.

5. After the gateway restarts, plug your cable into the one of the analog ports on the front panel.



Step A4: Verify Interoperability Verify that the gateway is correctly configured for your environment.

To verify that the gateway is correctly configured for analog call routing:

1. Log in to the gateway using your browser, changing your PC’s IP address to one that is able to communicate with the gateway’s new IP address, if required.

2. Click the Home button; the FXO icon on the Home page should be grey, meaning idle. While a call is connected, the icon should appear green.

3. Under 'General Information', confirm that the IP address, subnet mask, and default gateway parameters are correct.




Appendix B: SIP Configuration This section describes how to configure AudioCodes for use with a SIP trunk.

The instructions in this section assume that you have already configured the server for T1/E1.

Step B1: Change the Default Proxy Option to No This step shows how to set the default proxy setting to No. You initially set this to Yes through the steps on page 23.

To change the default proxy setting to No:

1. Choose VoIP > SIP Definitions > Proxy & Registration. 2. Set ‘Use Default Proxy’ to No. 3. Click Submit.



Step B2: Enable IP-to-IP This section shows how to enable IP-to-IP routing.

You must have the appropriate license to enable this. If you do not see the choice Applications Enabling on the VoIP menu, you must obtain the license before you proceed.

To enable IP to IP routing:

1. Choose VoIP > Applications Enabling > Applications Enabling. 2. Set ‘IP to IP Application’ to Enable. 3. Click Submit.

4. Click Burn. 5. Reboot the server before you continue to the next step.



Step B3: Increase Media Channels This step shows how to increase the number of Media channels.

Media channels are used for transcoding in an IP-to-IP configuration.

Two media channels are used for each call. The Mediant 1000B gateway should be equipped with an MPM module, which provides the necessary resources. Each MPM module provides 40 media channels. Up to three MPM modules can be installed, allowing up to 60 simultaneous IP-to-IP calls.

Refer to the AudioCodes documentation for instructions on installing the MPM module.

To increase the number of available Media channels:

1. Choose VOIP > IP Media > IP Media Settings. 2. Set ‘Number of Media Channels’ to an appropriate quantity (for example, 20). 3. Click Submit.



Step B4: Remove the Default Proxy Set This step shows how to remove the default proxy set that you set up earlier, on page 23.

To remove the default Proxy Set:

1. Choose VOIP > Control Network > Proxy Sets Table. 2. Choose Proxy Set ID 0 (if it does not automatically display). 3. Remove the IP Address and port from row 1. 4. Click Submit.



Step B5: Configure IP Group 1 for MaxCS This step shows how to configure an IP group to route calls from MaxCS to the SIP trunk.

To configure an IP Group for routing MaxCS calls to the SIP trunk:

1. Choose VOIP > Control Network > IP Group Table. 2. Choose Index 1 and configure these settings:

• Set ‘Type’ to SERVER • For ‘Description,’ enter AltiGen MaxCS • Set ‘Proxy Set ID’ to 1 • Set IP Profile ID to 1 • Make sure that ‘Routing Mode’ is set to Routing Table • Set ‘Serving IP Group ID’ to -1

3. Click Submit.



Step B6: Configure IP Group 2 for the SIP Trunk This step shows how to configure an IP group to route calls from the SIP trunk to Lync.

To configure an IP Group for routing calls to Lync:

1. Choose VOIP > Control Network > IP Group Table. 2. Choose Index 2 and configure these settings:

• Set ‘Type’ to SERVER • For ‘Description,’ enter SIP Trunk • Set ‘Proxy Set ID’ to 2 • Set IP Profile ID to 2 • Make sure that ‘Routing Mode’ is set to Routing Table • Set ‘Serving IP Group ID’ to -1

3. Click Submit.



Step B7: Configure Proxy Set 1 for MaxCS This step shows how to create a proxy set for MaxCS.

To create a new proxy set:

1. Choose VOIP > Control Network > Proxy Sets Table. 2. Configure settings for Proxy set 1:

• Set ‘Proxy Address’ to the IP Address and port of the MaxCS Server • Set ‘Transport Type’ to UDP

3. Click Submit.



Step B8: Configure Proxy Set 2 for the SIP Trunk This step shows how to create a proxy set for the SIP Trunk.

To create a new proxy set:

1. Choose VOIP > Control Network > Proxy Sets Table. 2. Switch to Proxy Set 2 and configure these settings:

• Set ‘Proxy Address’ to the IP Address and port of the SIP Trunk • Set ‘Transport Type’ to UDP

3. Click Submit.



Step B9: Configure Coder Group 1 for MaxCS This step shows how to create a coder group set for MaxCS.

To create a new coder group:

1. Choose VOIP > Coders and Profiles > Coders Group Settings. 2. Choose Coder Group 1. 3. Set the Coder Names as shown in the following figure. 4. Click Submit.



Step B10: Configure Coder Group 2 for the SIP Trunk This step shows how to create another coder group, this time for the SIP Trunk.


1. Choose VOIP > Coders and Profiles > Coders Group Settings. 2. Choose Coder Group 2. 3. Set the Coder Names as shown in the following figure.

Note: Make sure that you place the information for G.729 on the first row.

4. Click Submit.



Step B11: Configure IP Profile 1 for MaxCS This step shows how to create an IP Profile for MaxCS, configuring the profile to use Coder group 1.


1. Choose VOIP > Coders and Profiles > IP Profile Settings. 2. For Profile 1, set these parameters:

• Profile Name: MaxCS • Coder Group: Coder Group 1

3. Click Submit.



Step B12: Configure IP Profile 2 for the SIP Trunk This step shows how to create an IP Profile for the SIP trunk, configuring the profile to use Coder group 2.


1. Choose VOIP > Coders and Profiles > IP Profile Settings. 2. Select Profile 2 and set these parameters:

• Profile Name: SIP Trunk • Coder Group: Coder Group 2

3. Click Submit.



Step B13: Remove Plus Sign in Destination Phone Number This step shows how to remove the plus sign (+) in the destination phone number.

To remove the plus sign:

1. Choose VOIP > GW and IP to IP > Manipulations > Dest Number IP->Tel. 2. Enter 1 and click Add. 3. In the row for Index 1, set these parameters:

• In the ‘Destination Prefix’ field, enter a plus sign (+) • In the ‘Stripped Digits from Left’ field, enter 1

4. Click Apply.



Step B14: Remove Plus Sign in the Source Phone Number This step shows how to remove the plus sign (+) in the source phone number.

To remove the plus sign:

1. Choose VOIP > GW and IP to IP > Manipulations > Source Number IP->Tel. 2. Enter 1 and click Add. 3. In the row for Index 1, set these parameters:

• In the ‘Source Prefix’ field, enter a plus sign (+) • In the ‘Stripped Digits from Left’ field, enter 1

4. Click Apply.



Step B15: Configure Outbound IP Routing Rules This section shows how to configure routing rules for outbound IP-to-IP calls. You will make several entries in this routing table.

To configure SIP Trunk routing:

1. Choose VOIP > GW and IP to IP > Routing > Tel to IP Routing. 2. For Index 1 (first row), indicate that outbound calls should go to IP Group 1 (the

MaxCS server): • Set ‘Src. IP Group ID’ to -1 (all) • Set ‘Src Trunk Group ID’ to 1 • Set’ Dest. IP Group ID’ to 1

3. For Index 2, indicate that calls from the SIP Trunk (Proxy Set 2) should go to the MaxCS server: • Set ‘Src. IP Group ID’ to 2 • For ‘Src Trunk Group ID’ enter an asterisk * • Set ‘Dest. IP Group ID’ to 1

4. For Index 3, indicate that calls from MaxCS (Proxy Set 1) should go to the SIP Trunk: • Set ‘Src. IP Group ID’ to 1 • For ‘Src Trunk Group ID’ enter an asterisk * • Set’ Dest. IP Group ID’ to 2

5. Click Submit.



Step B16: Configure Inbound IP Routing Rules This section shows how to configure inbound routing rules.

To configure inbound SIP Trunk routing:

1. Choose VOIP > GW and IP to IP > Routing > IP to Trunk Group Routing. 2. For Index 1, indicate that calls from the MaxCS server should route to IP Group 1:

• Set ‘Source IP Address’ to the IP address of MaxCS Server • Set ‘Source IP Group’ to 1

3. For Index 2, indicate that calls from the SIP Trunk should route to IP Group 2: • Set ‘Source IP Address’ to the IP address of the SIP Trunk • Set ‘Source IP Group’ to 2


6. On the toolbar, from the Device Actions menu, choose Reset; the reset process takes

approximately four minutes to complete.



Step B17: Verify Interoperability Verify that the gateway is correctly configured for your environment.

To verify that the gateway is correctly configured for SIP call routing:

1. One the reset completes, log into the gateway.




Appendix C: AudioCodes ini File Following is an example of an AudioCodes ini file for AltiGen MaxCS. Your .ini file will differ slightly from this example.

;************** ;** Ini File ** ;************** ;Board: Mediant 1000 - MSBG ;Serial Number: 3248912 ;Slot Number: 1 ;Software Version: 6.40A.039.008 ;DSP Software Version: 620AE3 => 640.03 ;Board IP Address: 10.30.6.55 ;Board Subnet Mask: 255.255.224.0 ;Board Default Gateway: 10.30.0.1 ;Ram size: 512M Flash size: 64M ;Num of DSP Cores: 18 Num DSP Channels: 88 ;Profile: NONE ;Key features:;Board Type: Mediant 1000 - MSBG ;DSP Voice features: IpmDetector ;Coders: G723 G729 GSM-FR G727 ILBC ;DATA features: Routing FireWall&VPN WAN Advanced-Routing ;E1Trunks=4 ;T1Trunks=4 ;Channel Type: RTP DspCh=240 IPMediaDspCh=240 ;PSTN Protocols: ISDN IUA=4 CAS ;Security: IPSEC MediaEncryption StrongEncryption EncryptControlProtocol ;IP Media: Conf VXML VoicePromptAnnounc(H248.9) ExtVoicePrompt=1MB ;Control Protocols: MSFT MGCP MEGACO SIP SBC=25 ;Default features:;Coders: G711 G726; ;--------- Mediant-1000 HW components --------- ; ; Slot # : Module type : # of ports : # of DSPs ;---------------------------------------------- ; 1 : FALC56 : 2 : 3 ; 2 : DAA_O : 4 : 1 ; 3 : FXS : 4 : 1 ; 4 : Empty ; 5 : Empty ; 6 : IPMedia : 0 : 2 ;---------------------------------------------- [SYSTEM Params] SyslogServerIP = 10.30.6.162 EnableSyslog = 1 ENABLEPARAMETERSMONITORING = 1 ActivityListToLog = 'pvc', 'afl', 'dr', 'fb', 'swu', 'ard', 'naa', 'spc', 'll' PM_VEDSPUtil = '1,79,88,15' [BSP Params] PCMLawSelect = 3 TDMBusClockSource = 4 [Analog Params]



PolarityReversalType = 1 MinFlashHookTime = 100 [ControlProtocols Params] AdminStateLockControl = 0 [MGCP Params] [MEGACO Params] EP_Num_0 = 0 EP_Num_1 = 1 EP_Num_2 = 1 EP_Num_3 = 0 EP_Num_4 = 0 [PSTN Params] TraceLevel_0 = 1 TraceLevel_1 = 0 ProtocolType = 10 FramingMethod = D PSTNReserved3 = 8 [SS7 Params] [Voice Engine Params] RFC2833TxPayloadType = 101 EnableAGC = 1 EnableDSPIPMDetectors = 1 CallProgressTonesFilename = 'usa_tones_13.dat' [WEB Params] LogoWidth = '145' HTTPSCipherString = 'RC4:EXP' [SIP Params] MAXDIGITS = 20 MEDIACHANNELS = 20 PLAYRBTONE2TEL = 3 ISTWOSTAGEDIAL = 0 BCHANNELNEGOTIATION = 0 SECURECALLSFROMIP = 2 CDRREPORTLEVEL = 4 CHANNELSELECTMODE = 4 GWDEBUGLEVEL = 5 ENABLEEARLYMEDIA = 1 SIPSESSIONEXPIRES = 300 DISCONNECTONBROKENCONNECTION = 0 CDRSYSLOGSERVERIP = 10.30.6.162 REDUNDANTROUTINGMODE = 2 ENABLEIP2IPAPPLICATION = 1



[SCTP Params] [VXML Params] [IPsec Params] [Audio Staging Params] [SNMP Params] [ PREFIX ] FORMAT PREFIX_Index = PREFIX_DestinationPrefix, PREFIX_DestAddress, PREFIX_SourcePrefix, PREFIX_ProfileId, PREFIX_MeteringCode, PREFIX_DestPort, PREFIX_SrcIPGroupID, PREFIX_DestHostPrefix, PREFIX_DestIPGroupID, PREFIX_SrcHostPrefix, PREFIX_TransportType, PREFIX_SrcTrunkGroupID, PREFIX_DestSRD, PREFIX_CostGroup, PREFIX_ForkingGroup; PREFIX 0 = *, , *, 0, 255, 0, -1, *, 1, *, -1, 1, -1, , -1; PREFIX 1 = *, , *, 0, 255, 0, 2, *, 1, *, -1, -1, -1, , -1; PREFIX 2 = *, , *, 0, 255, 0, 1, *, 2, *, -1, -1, -1, , -1; [ \PREFIX ] [ CoderName ] FORMAT CoderName_Index = CoderName_Type, CoderName_PacketInterval, CoderName_rate, CoderName_PayloadType, CoderName_Sce; CoderName 0 = g711Alaw64k, 20, 255, -1, 0; CoderName 1 = g711Ulaw64k, 20, 255, -1, 0; CoderName 2 = g729, 20, 0, -1, 0; [ \CoderName ] [ TrunkGroup ] FORMAT TrunkGroup_Index = TrunkGroup_TrunkGroupNum, TrunkGroup_FirstTrunkId, TrunkGroup_FirstBChannel, TrunkGroup_LastBChannel, TrunkGroup_FirstPhoneNumber, TrunkGroup_ProfileId, TrunkGroup_LastTrunkId, TrunkGroup_Module; TrunkGroup 0 = 1, 0, 1, 24, 1000, 0, 1, 1; TrunkGroup 1 = 1, 255, 1, 4, 2000, 0, 255, 2; [ \TrunkGroup ] [ NumberMapIp2Tel ] FORMAT NumberMapIp2Tel_Index = NumberMapIp2Tel_DestinationPrefix, NumberMapIp2Tel_SourcePrefix, NumberMapIp2Tel_SourceAddress, NumberMapIp2Tel_NumberType, NumberMapIp2Tel_NumberPlan, NumberMapIp2Tel_RemoveFromLeft, NumberMapIp2Tel_RemoveFromRight, NumberMapIp2Tel_LeaveFromRight, NumberMapIp2Tel_Prefix2Add, NumberMapIp2Tel_Suffix2Add, NumberMapIp2Tel_IsPresentationRestricted, NumberMapIp2Tel_SrcTrunkGroupID, NumberMapIp2Tel_SrcIPGroupID; NumberMapIp2Tel 1 = +, *, *, 255, 255, 1, 0, 255, , , 255, -1, -1; [ \NumberMapIp2Tel ]



[ SourceNumberMapIp2Tel ] FORMAT SourceNumberMapIp2Tel_Index = SourceNumberMapIp2Tel_DestinationPrefix, SourceNumberMapIp2Tel_SourcePrefix, SourceNumberMapIp2Tel_SourceAddress, SourceNumberMapIp2Tel_NumberType, SourceNumberMapIp2Tel_NumberPlan, SourceNumberMapIp2Tel_RemoveFromLeft, SourceNumberMapIp2Tel_RemoveFromRight, SourceNumberMapIp2Tel_LeaveFromRight, SourceNumberMapIp2Tel_Prefix2Add, SourceNumberMapIp2Tel_Suffix2Add, SourceNumberMapIp2Tel_IsPresentationRestricted, SourceNumberMapIp2Tel_SrcTrunkGroupID, SourceNumberMapIp2Tel_SrcIPGroupID; SourceNumberMapIp2Tel 1 = *, +, *, 255, 255, 1, 0, 255, , , 255, -1, -1; [ \SourceNumberMapIp2Tel ] [ PstnPrefix ] FORMAT PstnPrefix_Index = PstnPrefix_DestPrefix, PstnPrefix_TrunkGroupId, PstnPrefix_SourcePrefix, PstnPrefix_SourceAddress, PstnPrefix_ProfileId, PstnPrefix_SrcIPGroupID, PstnPrefix_DestHostPrefix, PstnPrefix_SrcHostPrefix, PstnPrefix_SrcSRDID, PstnPrefix_TrunkId; PstnPrefix 0 = *, -1, *, 10.30.5.210, 0, 1, *, *, , -1; PstnPrefix 1 = *, -1, *, 208.79.53.214, 0, 2, *, *, , -1; [ \PstnPrefix ] [ ProxyIp ] FORMAT ProxyIp_Index = ProxyIp_IpAddress, ProxyIp_TransportType, ProxyIp_ProxySetId; ProxyIp 0 = 10.30.6.185:5060, 0, 1; ProxyIp 1 = 208.79.53.214:5060, 0, 2; [ \ProxyIp ] [ TxDtmfOption ] FORMAT TxDtmfOption_Index = TxDtmfOption_Type; TxDtmfOption 0 = 4; [ \TxDtmfOption ] [ TrunkGroupSettings ] FORMAT TrunkGroupSettings_Index = TrunkGroupSettings_TrunkGroupId, TrunkGroupSettings_ChannelSelectMode, TrunkGroupSettings_RegistrationMode, TrunkGroupSettings_GatewayName, TrunkGroupSettings_ContactUser, TrunkGroupSettings_ServingIPGroup, TrunkGroupSettings_MWIInterrogationType; TrunkGroupSettings 0 = 1, 4, 255, , , -1, 255; [ \TrunkGroupSettings ]



[ IpProfile ] FORMAT IpProfile_Index = IpProfile_ProfileName, IpProfile_IpPreference, IpProfile_CodersGroupID, IpProfile_IsFaxUsed, IpProfile_JitterBufMinDelay, IpProfile_JitterBufOptFactor, IpProfile_IPDiffServ, IpProfile_SigIPDiffServ, IpProfile_SCE, IpProfile_RTPRedundancyDepth, IpProfile_RemoteBaseUDPPort, IpProfile_CNGmode, IpProfile_VxxTransportType, IpProfile_NSEMode, IpProfile_IsDTMFUsed, IpProfile_PlayRBTone2IP, IpProfile_EnableEarlyMedia, IpProfile_ProgressIndicator2IP, IpProfile_EnableEchoCanceller, IpProfile_CopyDest2RedirectNumber, IpProfile_MediaSecurityBehaviour, IpProfile_CallLimit, IpProfile_DisconnectOnBrokenConnection, IpProfile_FirstTxDtmfOption, IpProfile_SecondTxDtmfOption, IpProfile_RxDTMFOption, IpProfile_EnableHold, IpProfile_InputGain, IpProfile_VoiceVolume, IpProfile_AddIEInSetup, IpProfile_SBCExtensionCodersGroupID, IpProfile_MediaIPVersionPreference, IpProfile_TranscodingMode, IpProfile_SBCAllowedCodersGroupID, IpProfile_SBCAllowedCodersMode, IpProfile_SBCMediaSecurityBehaviour, IpProfile_SBCRFC2833Behavior, IpProfile_SBCAlternativeDTMFMethod, IpProfile_SBCAssertIdentity, IpProfile_AMDSensitivityParameterSuit, IpProfile_AMDSensitivityLevel, IpProfile_AMDMaxGreetingTime, IpProfile_AMDMaxPostSilenceGreetingTime, IpProfile_SBCDiversionMode, IpProfile_SBCHistoryInfoMode, IpProfile_EnableQSIGTunneling, IpProfile_SBCFaxCodersGroupID, IpProfile_SBCFaxBehavior, IpProfile_SBCFaxOfferMode, IpProfile_SBCFaxAnswerMode, IpProfile_EnableEarly183; IpProfile 1 = MAXCS, 1, 1, 0, 10, 10, 46, 40, 0, 0, 0, 0, 2, 0, 0, 0, 1, -1, 1, 0, 0, -1, 0, 4, -1, 1, 1, 0, 0, , -1, 0, 0, -1, 0, 0, 0, 0, -1, 0, 8, 300, 400, -1, -1, 0, -1, 0, 0, 1, 0; IpProfile 2 = "SIP Trunk", 1, 2, 0, 10, 10, 46, 40, 0, 0, 0, 0, 2, 0, 0, 0, 1, -1, 1, 0, 0, -1, 0, 4, -1, 1, 1, 0, 0, , -1, 0, 0, -1, 0, 0, 0, 0, -1, 0, 8, 300, 400, -1, -1, 0, -1, 0, 0, 1, 0; [ \IpProfile ] [ TargetOfChannel ] FORMAT TargetOfChannel_Index = TargetOfChannel_Destination, TargetOfChannel_Type, TargetOfChannel_HotLineToneDuration, TargetOfChannel_Module, TargetOfChannel_Port; TargetOfChannel 0 = 4085979225, 1, -1, 2, 1; [ \TargetOfChannel ] [ ProxySet ] FORMAT ProxySet_Index = ProxySet_EnableProxyKeepAlive, ProxySet_ProxyKeepAliveTime, ProxySet_ProxyLoadBalancingMethod, ProxySet_IsProxyHotSwap, ProxySet_SRD, ProxySet_ClassificationInput, ProxySet_ProxyRedundancyMode; ProxySet 0 = 0, 60, 0, 0, 0, 0, -1; ProxySet 1 = 0, 60, 0, 0, 0, 0, -1; ProxySet 2 = 0, 60, 0, 0, 0, 0, -1; [ \ProxySet ] [ IPGroup ]



FORMAT IPGroup_Index = IPGroup_Type, IPGroup_Description, IPGroup_ProxySetId, IPGroup_SIPGroupName, IPGroup_ContactUser, IPGroup_EnableSurvivability, IPGroup_ServingIPGroup, IPGroup_SipReRoutingMode, IPGroup_AlwaysUseRouteTable, IPGroup_RoutingMode, IPGroup_SRD, IPGroup_MediaRealm, IPGroup_ClassifyByProxySet, IPGroup_ProfileId, IPGroup_MaxNumOfRegUsers, IPGroup_InboundManSet, IPGroup_OutboundManSet, IPGroup_RegistrationMode, IPGroup_AuthenticationMode, IPGroup_MethodList, IPGroup_EnableSBCClientForking, IPGroup_ContactName; IPGroup 1 = 0, MAXCS, 1, , , 0, -1, 0, 0, 0, 0, , 1, 1, -1, -1, -1, 0, 0, , 0, ; IPGroup 2 = 0, "SIP Trunk", 2, , , 0, -1, 0, 0, 0, 0, , 1, 2, -1, -1, -1, 0, 0, , 0, ; [ \IPGroup ] [ CodersGroup0 ] FORMAT CodersGroup0_Index = CodersGroup0_Name, CodersGroup0_pTime, CodersGroup0_rate, CodersGroup0_PayloadType, CodersGroup0_Sce; CodersGroup0 0 = g711Alaw64k, 20, 255, -1, 0; CodersGroup0 1 = g711Ulaw64k, 20, 255, -1, 0; CodersGroup0 2 = g729, 20, 0, -1, 0; [ \CodersGroup0 ] [ CodersGroup1 ] FORMAT CodersGroup1_Index = CodersGroup1_Name, CodersGroup1_pTime, CodersGroup1_rate, CodersGroup1_PayloadType, CodersGroup1_Sce; CodersGroup1 0 = g711Ulaw64k, 20, 0, -1, 0; CodersGroup1 1 = g711Alaw64k, 20, 0, -1, 0; [ \CodersGroup1 ] [ CodersGroup2 ] FORMAT CodersGroup2_Index = CodersGroup2_Name, CodersGroup2_pTime, CodersGroup2_rate, CodersGroup2_PayloadType, CodersGroup2_Sce; CodersGroup2 0 = g729, 20, 0, -1, 0; CodersGroup2 1 = g711Ulaw64k, 20, 0, -1, 0; CodersGroup2 2 = g711Alaw64k, 20, 0, -1, 0; [ \CodersGroup2 ] [ RoutingRuleGroups ] FORMAT RoutingRuleGroups_Index = RoutingRuleGroups_LCREnable, RoutingRuleGroups_LCRAverageCallLength, RoutingRuleGroups_LCRDefaultCost; RoutingRuleGroups 0 = 0, 0, 1; [ \RoutingRuleGroups ] [ InterfaceTable ]



FORMAT InterfaceTable_Index = InterfaceTable_ApplicationTypes, InterfaceTable_InterfaceMode, InterfaceTable_IPAddress, InterfaceTable_PrefixLength, InterfaceTable_Gateway, InterfaceTable_VlanID, InterfaceTable_InterfaceName, InterfaceTable_PrimaryDNSServerIPAddress, InterfaceTable_SecondaryDNSServerIPAddress, InterfaceTable_UnderlyingInterface; InterfaceTable 0 = 6, 10, 10.30.6.55, 19, 10.30.0.1, 1, Voice, 10.10.0.45, , ; [ \InterfaceTable ] [ DspTemplates ] ; ; *** TABLE DspTemplates *** ; This table contains hidden elements and will not be exposed. ; This table exists on board and will be saved during restarts. ; [ \DspTemplates ] [ IPMediaChannels ] FORMAT IPMediaChannels_Index = IPMediaChannels_ModuleID, IPMediaChannels_DSPChannelsReserved; IPMediaChannels 0 = 6, 20; [ \IPMediaChannels ]

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