AudioCodes Enabling Technology Products

AC48x

CPE VoIP Toolkit

AC48x CPE VoIP Toolkit Demo Guide for PMC Reference Design
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Contents

Table of Contents
1 2 3 4 About the AC48x CPE VoIP Toolkit............................................................................ 7
1.1 PMC Reference Design .....................................................................................................7

AC48x CPE VoIP Toolkit Demo Requirements.......................................................... 9 Setting Basic Parameters to PAS6x01 EDK ............................................................ 11 Running the SIP Application .................................................................................... 13
4.1 Preparing the Configuration File ......................................................................................13 4.1.1 Connecting to a Proxy Server ............................................................................................13 4.1.2 Connecting In a Direct Call Mode ......................................................................................15 4.2 Using the Management Sample Application ....................................................................18 4.3 Running the SIP Application Directly with Configuration File ..........................................19 4.4 Setting up a Remote Gateway .........................................................................................19 4.5 Test Environment and Setup ...........................................................................................20 4.6 Demo Procedure..............................................................................................................20 4.6.1 Using Flash-only Key Sequence Style...............................................................................20 4.5.2.1 Making an Outgoing Call ....................................................................................20 4.5.2.1 Making an Incoming Call ....................................................................................21 4.5.2.1 Call Hold .............................................................................................................21 4.5.2.1 Call Transfer .......................................................................................................21 4.5.2.1 Semi Attended Transfer .....................................................................................21 4.5.2.1 Call Waiting ........................................................................................................22 4.5.2.1 Call Forwarding ..................................................................................................22 4.5.2.1 Caller ID ..............................................................................................................22 4.5.2.1 Three Way Conference ......................................................................................23 4.6.2 Using Flash + Digit Key Sequence Style ...........................................................................24 4.5.2.1 Making an Outgoing Call ....................................................................................24 4.5.2.1 Making an Incoming Call ....................................................................................24 4.5.2.1 Call Hold .............................................................................................................24 4.5.2.1 Call Transfer .......................................................................................................25 4.5.2.1 Semi Attended Transfer .....................................................................................25 4.5.2.1 Call Waiting ........................................................................................................25 4.5.2.1 Call Forwarding ..................................................................................................26 4.5.2.1 Caller ID ..............................................................................................................26 4.5.2.1 Three-Way Conference ......................................................................................27

A Tulip VoIP Gateway ................................................................................................... 29

A.1 Installing a Tulip VoIP Gateway .......................................................................................29 A.2 Configuring VoIP Parameters ..........................................................................................31 A.2.1 Configuring SIP Proxy .......................................................................................................32 A.2.2 Configuring for SIP Direct Call ...........................................................................................33

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List of Figures
Figure 3-1: Setting IP Address and Network Mask .......................................................................................... 11 Figure 3-2: Opening the Marvell Switch ........................................................................................................... 12 Figure 3-3: Setting the Default Gateway .......................................................................................................... 12 Figure 4-1: Downloading VoIPCfgFile_Proxy_cvt_improved_appl.cfg.............................................................15 Figure 4-2: Downloading VoIPCfgFile_Direct_cvt_improved_appl.cfg ............................................................17 Figure 4-3: Layout of Test Site ......................................................................................................................... 20 Figure A-1: Tulip VoIP Gateway Quick Setup Page ........................................................................................ 29 Figure A-2: Tulip VoIP Gateway Remote Administration Page .......................................................................30 Figure A-3: Tulip VoIP Gateway Line Settings Line 1 .................................................................................. 31 Figure A-4: Tulip VoIP Gateway Line Settings Line 2 .................................................................................. 31 Figure A-5: Tulip VoIP Gateway Signaling Protocol ........................................................................................ 32 Figure A-6: Tulip VoIP Gateway Signaling Protocol ........................................................................................ 33 Figure A-7: Tulip VoIP Gateway Speed Dial ................................................................................................... 33

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Notices

Notice
This is the AC48x CPE VoIP Toolkit Demo Guide for PMC Reference Design. 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. Updates to this document and other documents can be viewed by registered customers at www.audiocodes.com/support. Copyright 2008 AudioCodes Ltd. All rights reserved. This document is subject to change without notice. Refer to the current release notes that may be included with your documentation or hardware delivery. Date Published: Oct-26-2008 Date Printed: Oct-28-2008 Tip: When viewing this manual on CD, Web site or on any other electronic copy, all cross-references are hyperlinked. Click on the page or section numbers (shown in blue) to reach the individual cross-referenced item directly. To return back to the point from where you accessed the cross-reference, press Alt + .

Trademarks
AC logo, Ardito, AudioCoded, AudioCodes, AudioCodes logo, CTI, CTI Squared, InTouch, IPmedia, Mediant, MediaPack, MP-MLQ, NetCoder, Netrake, Nuera, Open Solutions Network, OSN, Stretto, 3GX, TrunkPack, VoicePacketizer, VoIPerfect, What's Inside Matters, Your Gateway To VoIP, are trademarks or registered trademarks of AudioCodes Limited. All other products or trademarks are the property of their respective owners.

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 provided by AudioCodes Distributors, Partners, and Resellers from whom the product was purchased. For Customer support for products purchased directly from AudioCodes, contact support@audiocodes.com.

Abbreviations and Terminology

Each abbreviation, unless widely used, is spelled out in full when first used. Only industrystandard terms are used throughout this manual. The $ symbol indicates hexadecimal notation.

Conventions
The commands viewed after entering help when the standard VT100 terminal emulation program is in monitor mode, are displayed in this document in Courier New font.

Related Documentation
Document # LTRT-77307 LTRT-77706 Document Name VoIPerfect CPE VoIP Toolkit Release Notes VoIPerfect CPE VoIP Toolkit Programmer's Guide

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1. About the AC48x CPE VoIP Toolkit

About the AC48x CPE VoIP Toolkit

AudioCodes AC48x CPE Toolkit is a Linux-based software development kit (SDK) for the AC48x family of Voice over Packet Processors. Based on AudioCodes field proven VoIPerfect software, the CPE Toolkit provides the infrastructure and a sample application for a VoIP Analog Telephone Adapter (ATA), utilizing an AC48x DSP and a Legerity VP880 or SiLAB ProSlic SLIC device. The CPE Toolkit integrates SIP stack and call control for creating a full SIP application.

1.1

PMC Reference Design

PMC EPON and GPON ONUs are cost-optimized, highly integrated reference designs based on PMCs PAS6x01 SoC, a voice sub-stream consisting of AudioCodes AC48x DSP family, Legerity Analog SLIC for telephony interface and complete operational software.

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2. AC48x CPE VoIP Toolkit Demo Requirements

AC48x CPE VoIP Toolkit Demo Requirements

The AC48x CPE Toolkit Demo requires the following: At least one EDK Reference Design board preloaded with AC48x CPE VoIP Toolkit software package which includes: mng_sample_appl - management sample application. acl_main application main task that controls the VoIP task. voip_task the SIP application. Init kernel module. ac48dsp kernel module - includes the VoicePacketizer. le88drv kernel module - Legerity Le88221 SLIC driver. Configuration file.

AudioCodes Tulip VoIP Gateway. PC connected to the EDK with RS-232 serial cable. At least two regular analog phones connected to the RJ-11 connectors on the EDK and to the RJ-11 connectors on the Tulip VoIP Gateway. An Ethernet hub or switch connected to the RJ-45 connectors on the EDK and the Tulip VoIP Gateway. AC48x CPE VoIP Toolkit Documentation.

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3. Setting Basic Parameters to PAS6x01 EDK

Setting Basic Parameters to PAS6x01 EDK

You should be familiar with PAS6x01 EDK Reference Design documentation before handling the VoIP section. The following configuration and actions are required before executing the VoIP applications.

To set Basic parameters to PAS6x01 EDK:

1. 2. 3. 4. 5. Reset the board. In the passhell, run the following command: periph Run the following command: emapper Run the following command: set ip <ipaddr> Run the following command: set netmask <mask> Figure 3-1: Setting IP Address and Network Mask

6.

Reset the board.

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Guide for PMC Reference Design 7. In the passhell, drag and drop the next script into the terminal: Periph mdio write 0x1a 1 0x203e write 0x1a 4 0x77 write 0x10 4 0x77 write 0x11 4 0x77 write 0x12 4 0x77 write 0x13 4 0x7 exit Figure 3-2: Opening the Marvell Switch

8.

Run the following command: ip route add default via <ipaddr> Figure 3-3: Setting the Default Gateway

In a shell window, the VoIP applications are now ready to run.

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4. Running the SIP Application

Running the SIP Application

This chapter describes the usage of the SIP application, which is a complete Analog Telephone Adapter (ATA) SIP application. The SIP application can be executed in one of the following ways: Using the management sample application Running the SIP application directly with the configuration file (stand-alone mode)

The demo can be configured using one of the following: A SIP proxy server (refer to Section 4.1.1 on page 13) Direct SIP calls (refer to Section 4.1.2 on page 15)

4.1
4.1.1

Preparing the Configuration File

Connecting to a Proxy Server
Prepare a SIP configuration file based on the file VoIPCfgFile_Proxy_cvt_improved_appl.cfg. Update the following parameters: # Basic parameters for identifying the users local_ip_address=<EDK IP address> voip/line/0/id=<Line 1 phone number> voip/line/0/auth_name=<Authentication name for Line 1> voip/line/0/auth_password=<Authentication password for Line 1> voip/line/1/id=<Line 2 phone number> voip/line/1/auth_name=<Authentication name for Line 2> voip/line/1/auth_password=<Authentication password for Line 2> # Vocoders voip/codec/0/enabled=1 voip/codec/0/name=PCMU voip/codec/0/ptime=20 voip/codec/1/enabled=1 voip/codec/1/name=PCMA voip/codec/1/ptime=20 voip/codec/2/enabled=1 voip/codec/2/name=G729 voip/codec/2/ptime=20 voip/codec/3/enabled=1 voip/codec/3/name=g723 voip/codec/3/bit_rate_hi=1 voip/codec/3/ptime=30 voip/codec/4/enabled=1 voip/codec/4/name=g726-32 voip/codec/4/ptime=20 # Proxy and Registrar Parameters voip/signalling/sip/port=<local SIP port> voip/signalling/sip/proxy_address=<Proxy IP address> voip/signalling/sip/proxy_port=<Proxy port> voip/signalling/sip/proxy_timeout=<Registration Expiration Time [sec]> voip/signalling/sip/sip_registrar/enabled=1 voip/signalling/sip/sip_registrar/port=<Registrar port> voip/signalling/sip/sip_registrar/addr=<Registrar IP address>

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Guide for PMC Reference Design For example: EDK board with IP address: 10.16.2.76. Two analog phones connected to the board: Line 1 phone number is 123, user name for registration is user1, password for registration is 123456. Line 2 phone number is 124, user name for registration is user2, password for registration is 654321.

Registrar IP address is 10.16.2.19, listens on port 5060. Proxy IP address is 10.16.2.19, listens on port 5060. Coders must be prioritized in the following order: 1. G.711 U-law coder 2. G.711 A-law coder 3. G.729 4. G.723 6.3 kbit/s 5. G.726 32 kbit/s Speed dial must be configured: *01 must be directed to 100000@<Proxy IP address>.

# Basic parameters for identifying the users local_ip_address=10.16.10.76 voip/line/0/id=123 voip/line/0/auth_name= user1 voip/line/0/auth_password=123456 voip/line/1/id=124 voip/line/1/auth_name= user2 voip/line/1/auth_password=654321 # Vocoders voip/codec/0/enabled=1 voip/codec/0/name=PCMU voip/codec/0/ptime=20 voip/codec/1/enabled=1 voip/codec/1/name=PCMA voip/codec/1/ptime=20 voip/codec/2/enabled=1 voip/codec/2/name=G729 voip/codec/2/ptime=20 voip/codec/3/enabled=1 voip/codec/3/name=g723 voip/codec/3/bit_rate_hi=1 voip/codec/3/ptime=30 voip/codec/4/enabled=1 voip/codec/4/name=g726-32 voip/codec/4/ptime=20 # Proxy and Registrar Parameters voip/signalling/sip/port=5060 voip/signalling/sip/proxy_address=10.16.2.19 voip/signalling/sip/proxy_port=5060 voip/signalling/sip/proxy_timeout=3600 voip/signalling/sip/sip_registrar/enabled=1 voip/signalling/sip/sip_registrar/port=5060 voip/signalling/sip/sip_registrar/addr=10.16.2.19 # Phone Book Configuration voip/phonebook/0/number=*01 voip/phonebook/0/destination_type=proxy voip/phonebook/0/user_id=100000

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AC48x CPE VoIP Toolkit Demo 1. 2. Copy this file to your FTP host directory.

4. Running the SIP Application

In the target terminal, change to directory /var/ftp, and then download the file using an FTP utility. Figure 4-1: Downloading VoIPCfgFile_Proxy_cvt_improved_appl.cfg

4.1.2

Connecting In a Direct Call Mode

Prepare a SIP configuration file based on the file VoIPCfgFile_Direct_cvt_improved_appl.cfg, and update the following parameters: # Basic parameters for identifying the users local_ip_address=<EDK IP address> voip/line/0/id=<Line 1 phone number> voip/line/1/id=<Line 2 phone number> # Vocoders voip/codec/0/enabled=1 voip/codec/0/name=PCMU voip/codec/0/ptime=20 voip/codec/1/enabled=1 voip/codec/1/name=PCMA voip/codec/1/ptime=20 voip/codec/2/enabled=1 voip/codec/2/name=G729 voip/codec/2/ptime=20 voip/codec/3/enabled=1 voip/codec/3/name=g723 voip/codec/3/bit_rate_hi=1 voip/codec/3/ptime=30 voip/codec/4/enabled=1 voip/codec/4/name=g726-32 voip/codec/4/ptime=20 # Proxy and Registrar Parameters voip/signalling/sip/sip_registrar/enabled=<1use registrar, 0Dont use registrar> voip/signalling/sip/use_proxy=<1use proxy, 0Dont use proxy >

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Guide for PMC Reference Design For example: EDK board with IP address 10.16.2.76. Two analog phones connected to the board: Line 1 phone number is 123. Line 2 phone number is 124.

Coders must be prioritized in the following order: 1. G.729 2. G.711 U-law coder 3. G.723 5.3 kbit/s 4. G.726 32 kbit/s 5. G.711 A-law coder Internal calls can be made between the two lines. Calls can be established with remote endpoints: user id 489 located at IP address 10.16.2.50, and user id 777 located at IP address 10.16.2.69.

# Basic parameters for identifying the users local_ip_address=10.16.2.76 voip/line/0/id=123 voip/line/1/id=124 # Vocoders voip/codec/0/enabled=1 voip/codec/0/name=G729 voip/codec/0/ptime=20 voip/codec/1/enabled=1 voip/codec/1/name=PCMU voip/codec/1/ptime=20 voip/codec/2/enabled=1 voip/codec/2/name=g723 voip/codec/2/bit_rate_hi=0 voip/codec/2/ptime=20 voip/codec/3/enabled=1 voip/codec/3/name=g726-32 voip/codec/3/ptime=20 voip/codec/4/enabled=1 voip/codec/4/name=PCMA voip/codec/4/ptime=20 # Proxy and Registrar Parameters voip/signalling/sip/sip_registrar/enabled=0 voip/signalling/sip/use_proxy=0 # Phone Book Configuration voip/phonebook/0/number=489 voip/phonebook/0/destination_type=direct voip/phonebook/0/user_id=489 voip/phonebook/0/user_address=10.16.10.50 voip/phonebook/0/user_port=5060 voip/phonebook/1/number=777 voip/phonebook/1/destination_type=direct voip/phonebook/1/user_id=777 voip/phonebook/1/user_address=10.16.10.69 voip/phonebook/1/user_port=5060 voip/phonebook/2/number=123 voip/phonebook/2/destination_type=local voip/phonebook/2/local_line=0 voip/phonebook/3/number=124 voip/phonebook/3/destination_type=local voip/phonebook/3/local_line=1 1. 2. Copy this file to your FTP host directory. In the target terminal, change to directory /var/ftp, and then download the file using an FTP utility.

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4. Running the SIP Application

Figure 4-2: Downloading VoIPCfgFile_Direct_cvt_improved_appl.cfg

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4.2

Using the Management Sample Application

The management sample application enables the user to control the VoIP application. The following commands are available in the sample application CLI: open: Opens the control to the VoIP application. run: Runs the VoIP application. config: Reconfigures the VoIP application. get_port_count: Gets the number of existing ports. set_port_status: Sets the port status (lock/unlock). get_port_status: Gets the port status (lock/unlock). exit: Closes this sample application. help: Provides help on available commands.

Using the sample application CLI commands: /* Downloading the configuration file via FTP */ ./usr/bin/audiocodes/apps/mng_sample_appl VoIP >> VoIP >> open open connection VoIP >> run var/ftp/VoIPCfgFile.cfg . . /* The VoIP application is loading */ . VoIP >> exit /* Downloading the new configuration file via FTP */ ./usr/bin/audiocodes/apps/mng_sample_appl VoIP >> open open connection VoIP >> config var/ftp/new_VoIPCfgFile.cfg . . /* The VoIP application is reloading */ . VoIP >> VoIP >> get_port_count Number of existing ports is: 2 VoIP >> VoIP >> VoIP >> get_port_status 1 Port status of channel 1 is: PORT_UNLOCK VoIP >> VoIP >> VoIP >> set_port_status 1 0 Setting port status of channel 1 to: PORT_LOCK VoIP >> VoIP >> set_port_status 1 1 Setting port status of channel 1 to: PORT_UNLOCK VoIP >>

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4. Running the SIP Application

4.3

Running the SIP Application Directly with Configuration File

In the stand-alone mode, the SIP application reads the configuration file directly and doesn't communicate with the management sample application. Before executing the voip_task application, navigate to the folder containing the AudioCodes sample applications (the exact folder is platform specific), for example: cd /audiocodes/apps Run the SIP application and provide the name of the configuration file (created in Section 4.1 on page 13) as an argument: ./voip task <Configuration File> & To reconfigure the SIP application, the application must be terminated with the command kill and reloaded with -r as the first argument and provided with the name of the new configuration file as the second argument: killall voip task 2>/dev/null ./voip_task -r <New Configuration File> &

Note:

The argument -r stands for reconfiguration.

4.4

Setting up a Remote Gateway

See Appendix A for instructions on how to set up a Tulip VoIP Gateway as a remote endpoint for the SIP session.

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4.5

Test Environment and Setup

Figure 4-3: Layout of Test Site

AudioCodes Tulip VoIP Gateway

Ethernet

Hub/Switch

PON/Ethernet
EDK Reference Design 2 1

4.6

Demo Procedure
The AC48x CPE VoIP Toolkit supports two key sequence styles: Flash-only Flash + digit The demo procedures for these two styles are described in the following subsections.

4.6.1
4.5.2.1
Step 1. 2. 3. 4.

Using Flash-only Key Sequence Style

Making an Outgoing Call
Description From EDK Channel 1, dial the number of Tulip VoIP Gateway's Channel 1 to establish a call. From EDK Channel 2, dial the number of Tulip VoIP Gateway's Channel 2 to establish a call. On-hook session from Step 1. On-hook session from Step 2. Expected Results Call established. Call established. Two sessions with toll quality. Call disconnected. Call disconnected.

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4.5.2.1
Step 1. 2. 3. 4.

Making an Incoming Call

Description From Tulip VoIP Gateway's Channel 1, dial the number of EDK Channel 1 to establish a call. From Tulip VoIP Gateway's Channel 2, dial the number of EDK Channel 2 to establish a call. On-hook session from Step 1. On-hook session from Step 2. Expected Results Call established. Call established. Two sessions with toll quality. Call disconnected. Call disconnected.

4.5.2.1
Step 1. 2. 3. 4.

Call Hold
Description From EDK Channel 1, dial the number of Tulip VoIP Gateway's Channel 1 to establish a call. From Tulip Channel 2, dial the number of the EDK Channel 1 to establish a call. Use Flash to switch between the two calls (multi line). On-Hook all phones. Expected Results Call established. Receive an incoming call and establish a call. Successful switching between the calls.

4.5.2.1
Step 1. 2. 3. 4.

Call Transfer
Description From EDK Channel 1, dial the number of Tulip VoIP Gateway's Channel 1 to establish a call. Press Flash Dial the number of EDK Channel 2 to establish a call. On-hook EDK Channel 1. Expected Results Call established. Dial tone received. Call established. Call is established between Tulip VoIP Gateway's Channel 1 and EDK Channel 2.

4.5.2.1
Step 1. 2. 3. 4. 5.

Semi Attended Transfer

Description From EDK Channel 1, dial the number of Tulip VoIP Gateway's Channel 1 to establish a call. Press Flash. Dial the number of EDK Channel 2. On-hook EDK Channel 1. Off-hook EDK Channel 2 Expected Results Call established. Dial tone received. Dont establish a call. EDK Channel 2 continues ringing. Call is established between Tulip VoIP Gateway's Channel 1 and EDK Channel 2.

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4.5.2.1
Step 1. 2. 3. 4. 5.

Call Waiting
Description From EDK Channel 1, dial the number of Tulip VoIP Gateway's Channel 1 to establish a call. From Tulip VoIP Gateway's Channel 2, dial the number of EDK Channel 1. On EDK Channel 1, press Flash. Use Flash to switch between the two calls (multi line). On-Hook all phones. Expected Results Call established. A call waiting tone is played on EDK Channel 1. The call is switched to the call waiting. Successful switching between the calls.

4.5.2.1

Call Forwarding
Call forwarding is disabled by default. To enable it, refer to the AC48x CPE VoIP Toolkit Programmer's Guide (Configuring Services Parameters).

Step 1. 2. 3. 4. 5.

Description From EDK Channel 1, dial the call forward key sequence, for example: *72. From EDK Channel 1, dial the number of Tulip VoIP Gateway's Channel 1. On-hook EDK Channel 1. From Tulip VoIP Gateway's Channel 2, dial the number of EDK Channel 1. From now on, all incoming calls are forwarded. Each time you off-hook EDK Channel 1, the Stutter tone is heard, notifying you that call forwarding is still active. To disable call forwarding, off-hook EDK Channel 1 (Stutter tone is heard) and then dial the call forward key sequence (e.g *72). On-hook EDK Channel 1.

Expected Results Dial tone received. Stutter tone received.

The call is forwarded to Tulip VoIP Gateway's Channel 1.

6. 7.

Call forwarding is disabled.

4.5.2.1
Step 1.

Caller ID
Description From Tulip VoIP Gateway's Channel 1, dial the number of EDK Channel 1. Off-hook EDK Channel 1. From Tulip VoIP Gateway's Channel 2, dial the number of EDK Channel 1. Expected Results Caller ID appears on EDK Channel 1 analog phone screen. Call established. A call waiting tone is played on EDK Channel 1 and Tulip VoIP Gateway's Caller ID is shown on EDK Channel 1 analog phone screen. The call is switched to the waiting call.

2. 3.

4.

On EDK Channel 1, press Flash.

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4. Running the SIP Application

4.5.2.1
Step 1. 2.

Three Way Conference

Description From EDK Channel 1, dial the number of Tulip VoIP Gateway's Channel 1 to establish a call. On EDK Channel 1 press Flash. Expected Results Call established. Tulip VoIP Gateway's Channel 1 is on hold and a dial tone is heard on EDK Channel 1. Call established.

3. 4. 5.

Dial the number of Tulip VoIP Gateway's Channel 2 to establish a call. Press Flash to enable three-way conferencing. On-Hook one of the Tulip VoIP Gateway Channels.

Call remains between EDK Channel 1 and the second Channel of the Tulip VoIP Gateway.

Note:

On AC48802 DSP template, three-way conferencing can only be achieved between a local port and two remote IPs.

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4.6.2
4.5.2.1
Step 1. 2. 3. 4.

Using Flash + Digit Key Sequence Style

Making an Outgoing Call
Description From EDK Channel 1, dial the number of Tulip VoIP Gateway's Channel 1 to establish a call. From EDK Channel 2, dial the number of Tulip VoIP Gateway's Channel 2 to establish a call. On-hook session from Step 1. On-hook session from Step 2. Expected Results Call established. Call established. Two sessions with toll quality. Call disconnected. Call disconnected.

4.5.2.1
Step 1. 2. 3. 4.

Making an Incoming Call

Description From Tulip VoIP Gateway's Channel 1, dial the number of EDK Channel 1 to establish a call. From Tulip VoIP Gateway's Channel 2, dial the number of EDK Channel 2 to establish a call. On-hook session from Step 1. On-hook session from Step 2. Expected Results Call established. Call established. Two sessions with toll quality. Call disconnected. Call disconnected.

4.5.2.1
Step 1. 2.

Call Hold
Description From EDK Channel 1, dial the number of Tulip VoIP Gateway's Channel 1 to establish a call. On EDK Channel 1, press Flash + '1'. Expected Results Call established. Tulip VoIP Gateway's Channel 1 is on hold and a dial tone is heard on EDK Channel 1. Call established. Successful switching between the calls.

3. 4. 5.

Dial the number of Tulip VoIP Gateway's Channel 2 to establish a call. Use Flash + '1' to switch between the two calls (multi line). On-Hook all phones.

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4. Running the SIP Application

4.5.2.1
Step 1. 2. 3. 4.

Call Transfer
Description From EDK Channel 1, dial the number of Tulip VoIP Gateway's Channel 1 to establish a call. Press Flash + '1'. Dial the number of EDK Channel 2 to establish a call. Press Flash + '2'. Expected Results Call established. Dial tone received. Call established. Call is established between Tulip VoIP Gateway's Channel 1 and EDK Channel 2.

4.5.2.1
Step 1. 2. 3. 4. 5.

Semi Attended Transfer

Description From EDK Channel 1, dial the number of Tulip VoIP Gateway's Channel 1 to establish a call. Press Flash + '1'. Dial the number of EDK Channel 2. Press Flash + '2' and hang-up the phone. Pick up EDK Channel 2 phone. Call is established between Tulip VoIP Gateway's Channel 1 and EDK Channel 2. Expected Results Call established. Dial tone received. EDK Channel 2 rings.

4.5.2.1
Step 1. 2. 3. 4. 5.

Call Waiting
Description From EDK Channel 1, dial the number of Tulip VoIP Gateway's Channel 1 to establish a call. From Tulip VoIP Gateway's Channel 2, dial the number of EDK Channel 1. On EDK Channel 1 press Flash + '1'. Use Flash + '1' to switch between the two calls (multi line). On-Hook all phones. Expected Results Call established. A call waiting tone is played on EDK Channel 1. The call is switched to the call waiting. Successful switching between the calls.

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4.5.2.1

Call Forwarding
Call forwarding is disabled by default. To enable it, refer to the AC48x CPE VoIP Toolkit Programmer's Guide (Configuring Services Parameters).

Step 1. 2. 3. 4. 5.

Description From EDK Channel 1, dial the call forward key sequence, for example: *72. From EDK Channel 1, dial the number of Tulip VoIP Gateway's Channel 1. On-hook EDK Channel 1. From Tulip VoIP Gateway's Channel 2, dial the number of EDK Channel 1. From now on, all incoming calls are forwarded. Every time you off-hook EDK Channel 1, the Stutter tone is heard, notifying you that call forwarding is still active. To disable call forwarding, off-hook EDK Channel 1 (Stutter tone is heard) and then dial the call forward key sequence (e.g *72). On-hook EDK Channel 1.

Expected Results Dial tone received. Stutter tone received.

The call is forwarded to Tulip VoIP Gateway's Channel 1.

6. 7.

Call forwarding is disabled.

4.5.2.1
Step 1.

Caller ID
Description From Tulip VoIP Gateway's Channel 1, dial the number of EDK Channel 1. Off-hook EDK Channel 1. From Tulip VoIP Gateway's Channel 2, dial the number of EDK Channel 1. Expected Results Caller ID appears on EDK Channel 1 analog phone screen. Call established. A call waiting tone is played on EDK Channel 1 and Tulip VoIP Gateway's Caller ID is shown on EDK Channel 1 analog phone screen. The call is switched to the waiting call.

2. 3.

4.

On EDK Channel 1 press Flash + '1'.

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4. Running the SIP Application

4.5.2.1
Step 1. 2.

Three-Way Conference
Description From EDK Channel 1, dial the number of Tulip VoIP Gateway's Channel 1 to establish a call. On EDK Channel 1, press Flash + '1'. Expected Results Call established. Tulip VoIP Gateway's Channel 1 is on hold and a dial tone is heard on EDK Channel 1. Call established.

3. 4. 5.

Dial the number of Tulip VoIP Gateway's Channel 2 to establish a call. Press Flash + '3 to enable three-way conferencing. On-Hook one of the Tulip VoIP Gateway Channels.

Call remains between EDK Channel 1 and the second Channel of the Tulip VoIP Gateway.

Note:

On AC48802 DSP template, three-way conferencing can only be achieved between a local port and two remote IPs.

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A. Tulip VoIP Gateway

A
A.1

Tulip VoIP Gateway

Installing a Tulip VoIP Gateway
You can connect a third-party SIP device (i.e. your own), although it is recommended to use an AudioCodes Tulip VoIP Gateway.

To install a Tulip VoIP Gateway:

1. 2. Perform the procedures described in sections 1, 2, and 3 (excluding Section 3.5) in the Tulip VoIP Gateway Telephone Adapter Quick Installation Guide. In the Quick Setup page, perform the following: a. b. c. In the 'Connection Type' drop-down list, select Manual IP Address Ethernet Connection. Fill in the appropriate values for IP Address, Subnet Mask and Default Gateway. Click OK. Figure A-1: Tulip VoIP Gateway Quick Setup Page

3.

In the Advanced page, click Remote Administration, and then configure the following parameters: a. b. c. Check the Using Primary Telnet Port (23) check box. Check the Using Primary HTTP Port (80) check box. Click OK.

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4. 5.

Disconnect the PC from the Tulip VoIP Gateways LAN/PC port, and then connect it to the hub/switch. In your Web browser, enter the IP address of the Tulip VoIP Gateway.

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A. Tulip VoIP Gateway

A.2

Configuring VoIP Parameters

The procedure below describes how to configure the VoIP parameters.

To configure the VoIP parameters:

1. 2. In the Voice Over IP page, click the tab Line Settings. Click the Action icon located on the right of each line, and then configure the appropriate parameters: Figure A-3: Tulip VoIP Gateway Line Settings Line 1

Figure A-4: Tulip VoIP Gateway Line Settings Line 2

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A.2.1

Configuring SIP Proxy

The procedure below describes how to configure the SIP Proxy server.

To configure the SIP Proxy server:

1. 2. In the Voice Over IP page, click the tab Line Settings. Check the Use SIP Proxy check box, and then configure the following parameters: Proxy IP Address or Host Name. 'Proxy Port' (remain with the default value). Figure A-5: Tulip VoIP Gateway Signaling Protocol

3.

Click OK.

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A. Tulip VoIP Gateway

A.2.2

Configuring for SIP Direct Call

The procedure below describes how to configure a direct SIP call.

To configure a direct SIP call:

1. In the Voice Over IP page, click the tab Line Settings, and then clear the check box Use SIP Proxy. Figure A-6: Tulip VoIP Gateway Signaling Protocol

2.

Click the tab Speed Dial, click the Action icon on the right, and then configure the appropriate parameters for each remote endpoint lines. Figure A-7: Tulip VoIP Gateway Speed Dial

3.

Click OK.

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AudioCodes Enabling Technology Products

AC48x

CPE VoIP Toolkit

AC48x CPE VoIP Toolkit Demo Guide for PMC Reference Design
Version 2.6.1

www.audiocodes.com