Avaya Solution & Interoperability Test Lab Application Notes for Configuring SIP IP Telephony Using Avaya 4600 Series IP Telephones, Avaya one-X™ Desktop Edition, and Asterisk Business Edition™ PBX – Issue 1.0 Abstract These Application Notes describe the configuration steps required to configure Avaya 4600 Series IP Telephones and Avaya one-X™ Desktop Edition with the Asterisk Business Edition™ PBX.
1. Introduction These Application Notes describe the configuration steps for using Avaya 4600 Series IP Telephones and Avaya one-X™ Desktop Edition1 with the Asterisk Business Edition PBX. Only those configuration steps pertinent to interoperability of the Asterisk and Avaya equipment are covered. General administration information can be found in the product documentation as well as the specific references listed in Section 9. The configuration used in the test is shown in Figure 1.
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2.
Feature 1. Call Hold 2. Consultation Hold 3. Music On Hold Unattended Transfer – via the phone 4. Unattended Transfer – via the server 5. Attended Transfer Supplementary Features Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes 6. Transfer – Instant Messaging No No Call Forward Unconditional – via the server 7. Call Forward Unconditional – via the phone Call Forward Busy – via the server 8.
3.1. Call Forwarding In addition to the call forwarding features provided by the Asterisk server, the Avaya 4600 Series IP Telephones, except for the 4602SW, support local call forwarding. (Avaya one-X™ Desktop Edition does not support local call forwarding.) These features can be enabled via the 46xxsettings.txt configuration file (see Section 5). 3.2.
Avaya one-X™ Desktop Edition is able to set and track the following states: • • • • • • the user’s station is on-line and idle (registered) [Available] the user’s station is off-line (un-registered) [Offline] the user’s station is on-line but does not want to be tracked [Invisible/Offline] the user’s station is on-line and “on the phone” [On the Phone] the user’s station is on-line and the user is “away from the phone” [Away] the user’s station is on-line and the user is “busy” [Busy] The states that both
4. Configure the Asterisk Business Edition PBX 4.1. Install Asterisk Business Edition PBX Software Asterisk Business Edition PBX must be installed on Red Hat Enterprise 3 or Fedora Core 3. For these Application Notes, Red Hat Enterprise 3 was used. The installation of the software is covered in Reference [4] and Reference [5]. These Application Notes do not cover the installation of the software. 4.2.
Steps 2. Configure the SIP domain. Description Edit the sip.conf file with the vi editor (or other editor). All of the configuration files are organized into different “contexts”. A context is created by placing the context name in brackets (e.g., [general]). A parameter in this file, like all of the .conf files, will be in the following format: [context1] parameter1=value parameter2=value To configure the SIP domain, place the following parameters in the general context.
Steps 3. Configure users on the Asterisk server. Description The configuration of the users is done in two files: • /etc/asterisk/sip.conf • /etc/asterisk/extensions.conf The majority of the configuration is in the sip.conf file. The following parameters must be defined to configure a user.
Steps 4. Configure the users for voice mail. Description The configuration of the voice mail users is done in two files: • /etc/asterisk/sip.conf • /etc/asterisk/voicemail.conf The following parameters must be set in sip.conf to configure a user for voice mail. [] mailbox= Edit the sip.conf file with the vi editor (or other editor). The following is an excerpt from the sip.conf file used for the testing documented for these Application Notes. sip.
Steps Description 5. Verify users configured on the Asterisk server. Log into the Linux system using the administration account (admin) that was created when Linux was installed. Log in as the root account by entering “su -”. Start the Asterisk command line interface (CLI) by entering “asterisk –c” login as: admin admin@10.2.2.60's password: [admin@interop-asterisk-be admin]$ su Password: [root@interop-asterisk-be root]# asterisk –c The Asterisk CLI window is displayed. Asterisk ABE-A.
4.3. Administer the Supplementary Features on the Asterisk Server Additional administration is needed to support the following features: • Unattended transfer (via the server) • Call park • Call pickup • Call forwarding • Shuffling (Reinvite)2 (See Reference [8]) • Presence tracking NOTE: Some of the features listed above can be configured in one sequence. However, there are some features (e.g., Unattended Transfer & Shuffling, Call Forward Unconditional & Call Forward Busy) that are mutually exclusive.
Steps Description 1. Configure the server for Unattended Transfer (via the server). The Asterisk server must be configured to support Unattended Transfer. When the feature is enabled, a user that is active on a call can press the “#” sign and a prompt from the Asterisk IVR system is played. The user is asked to enter the extension to which the call should be transferred. Upon entering a valid extension, the call is transferred.
Steps 2. Configure the server for Call Pickup. Description Normally, no configuration is needed to support the Call Pickup feature. However, the default dial string to invoke the feature is “*8”. The version of the Avaya SIP clients under test cannot dial “*” as part of the dial string. Therefore, the call pickup code was changed. For these Application Notes, Call Pickup dial string was changed to “200”. Edit the features.conf file with the vi editor (or other editor).
Steps Description 4. Configure the server for Call Forwarding Unconditional. Call forwarding is supported by modifying the existing scripts or creating custom scripts in the extensions.conf file. The method used for these Application Notes to enable the call forwarding feature was to change the dial plan so that all calls are forwarded. This method was expedient for this testing.
Steps Description 5. Configure the server for Call Forwarding No Answer. In this example, all calls are forwarded to extension 60007. Normally, calls that are not answered would be forwarded to voice mail. Similar to the Call Forwarding Unconditional feature, the method used to turn on this feature was expedient for this testing. The user can invoke this feature from the telephone (see Section 5, step #3). Edit the extensions.conf file with the vi editor (or other editor).
Steps Description 7. Configure the server for Shuffling/Reinvite. By default, all voice goes through the server. Shuffling can be turned on by modifying the sip.conf file. The following parameters must be set in sip.conf to configure a user for shuffling. [] canreinvite=yes|no [shuffle or don’t shuffle] Edit the sip.conf file with the vi editor (or other editor). The following is an excerpt from the sip.conf file used for the testing documented for these Application Notes.
Steps Description 8. Configure the system to support Presence Tracking. Presence tracking is supported but only in a limited fashion. Asterisk reports presence status based on the actual status of the phone. That is, whether the phone is registered & idle [Available], active on the phone [On the Phone], or un-registered [Offline]. The configuration of this feature is in the extensions.conf file. The following parameters must be set in extensions.conf.
5. Configure the Avaya 4600 IP Telephones The SIP software should be installed in the Avaya 4600 Series IP Telephones using the procedures described in Reference [1]. The SIP specific software can be downloaded from the Avaya Support Center site (http://www.avaya.com/support). Download the 46xxSIP*.zip file (where * is the date) and install the files per the instructions in Reference [1]. Any Avaya 4600 Series IP Telephone that has the H.
Steps Description 2. An excerpt of the 46xxsettings.txt file with the SIP specific parameters is shown in Figure 2. A description of the important parameters is in Table 4. Edit these parameters to configure the SIP settings for the Avaya 4600 Series IP Telephones. The sample in Figure 2 shows the values used in the compliance test. NOTE: Lines that start with “##” are comments.
Steps Description 3. Table 4 shows the SIP specific parameters that can be configured. The parameters that are critical to configure are DIALPLAN, SIPDOMAIN, SIPROXYSRVR, and SIPREGISTRAR. In addition, for testing local call forwarding, also configure CALLFWDSTAT and CALLFWDADDR. CALLFWDSTAT: This parameter defines which call forwarding buttons are configured on the phone.
Steps Description 4. Connect the Avaya 4600 Series IP Telephone to the network and reboot the phone. If TFTP or HTTP/HTTPS support have been properly configured, the phone will download the software and configuration files and prompt the user for the extension and password. Enter the extension followed by the # key. For example, 60000#. Ext.= #=OK New=60000 Enter the password followed by the # key. For example, 123456#.
6. Configure the Avaya one-X™ Desktop Edition Software The Avaya one-X™ Desktop Edition R2.1 software is available on the Avaya Support Center site (http://www.avaya.com/support). The installation and usage instructions for Avaya one-X™ Desktop Edition are documented in Reference [2]. The Avaya one-X™ Desktop Edition can be configured manually using the graphical user interface. As an option, there are some configurations items that can be configured via the 46xxsettings.txt file.
Steps Description 1. Edit the DHCP option configured for the Avaya 4600 Series IP Telephones. Start the DHCP server client from the Windows Control Panel (Start Æ Settings Æ Control Panel). Click on Administrative Tools. Click on DHCP. 2. The DHCP main dialogue is displayed. Click on the “+” next to the DHCP scope that was created for the Avaya 4600 Series IP Telephones. In this example, select the scope for the 10.2.2.x subnet.
Steps Description 3. Additional parameters are shown for the 10.2.2.x DHCP scope on the left hand side of the DHCP window. Click on Scope Options. Select the scope option that was created for the Avaya 4600 Series IP Telephones on the right hand side of the DHCP window. In this example, select the scope option 176 46xxOptions. From the menu, select Action Æ Properties. JHB; Reviewed: PV 7/12/2006 Solution & Interoperability Test Lab Application Notes ©2006 Avaya Inc. All Rights Reserved.
Steps Description 4. The Scope Options dialogue is displayed. The value for the 176 46xxOptions option needs to be modified. The default string used by Avaya one-X™ Desktop Edition is in the form of: HTTPSRVR=nnn.nnn.nnn.nnn HTTPSRVR is the IP address of the server that holds the configuration script for Avaya one-X™ Desktop Edition. The TFTPSRV parameter is also set for the Avaya 4600 Series IP Telephones. For this example, the option is set to the following. TFTPSRVR=10.2.2.102,HTTPSRVR=10.2.2.
6.3. Configure 46xxsettings.txt for Avaya one-X™ Desktop Edition (OPTIONAL) Assuming that the DHCP server has been configured to support Avaya one-X™ Desktop Edition (see Section 6.2), the 46xxsettings.txt configuration file must be modified to provide the Avaya one-X™ Desktop Edition configuration information. Steps Description 1. Locate the 46xxsettings.txt file to administer the SIP specific settings for Avaya one-X™ Desktop Edition.
Steps Description 3. Table 5 shows the SIP specific parameters that can be configured. The parameters that are critical to configure are SIPPROXYSRVR and WEBLMSRVR. SIPPROXYSRVR: Enter the IP address of the Asterisk server. WEBLMSRVR: Enter the address of the WebLM server.
6.4. Configure Avaya one-X™ Desktop Edition After Avaya one-X™ Desktop Edition is installed, it must be configured. Steps Description 1. Start Avaya one-X™ Desktop Edition via the menu Start Æ Programs Æ Avaya SIP Softphone Æ Avaya SIP Softphone.4 The first time the program is executed, the Configuration Wizard will be displayed. Edit the 46xxsettings.txt file to administer the Avaya one-X™ Desktop Edition specific settings.
Steps 2. The Account dialogue is displayed. Description Enter the name for the Avaya one-X™ Desktop Edition station. This name will be used for caller identification. Enter the user name that will be used to authenticate with the SIP registrar. NOTE: It is recommended to use a domain name instead of an IP address for the user name. For this example, the name is 60006@asterisk.com instead of 60006@10.2.2.60. Choose the appropriate option for the password. Click the Next button to continue. 3.
Steps Description 4. An updated SIP Server/License Server dialogue is displayed. If the information was not gathered from the DHCP & HTTP servers, the fields would be blank and have to be entered them manually. Click the Next button to continue. 5. The Profile dialogue is displayed. For this example, Lab was entered for the Profile Name. Any name can be used. Select Local Area Network for the Connection Type. The other options are Cable, xDSL, or IDSN and Modem (28800 bps or faster).
Steps Description 6. The Dialing Rules dialogue is displayed. As there are only internal calls made, the default values are used for these Application Notes. To configure the software for calls outside of the Asterisk server (local, long distance, and international), enter the values that correspond to the external dialing rules for the server. Click the Next button to continue. JHB; Reviewed: PV 7/12/2006 Solution & Interoperability Test Lab Application Notes ©2006 Avaya Inc. All Rights Reserved.
Steps Description 7. The Voicemail Integration dialogue is displayed. Avaya one-X™ Desktop Edition can be optioned to dial the voice mail extension. In this example, the default voice mail extension for the Asterisk server is 8500. When this option is configured and the Avaya client is registered with the server, clicking on the voice mail button (see Figure 4) will dial the voice mail system. Click the Next button to continue.
Steps Description 8. The Audio Wizard dialogue is displayed. Click the Next button to continue. JHB; Reviewed: PV 7/12/2006 Solution & Interoperability Test Lab Application Notes ©2006 Avaya Inc. All Rights Reserved. 35 of 47 Asterisk-AVAppN.
Steps Description 9. The Audio Wizard: Select Sound Device dialogue is displayed. The recommendation is to use a USB headset with Avaya one-X™ Desktop Edition. A list of recommended headsets is available on the Avaya Support Center (http://www.avaya.com/support). Depending on how many sound devices are installed or connected to the PC (e.g., built-in sound device, USB headset), there may be one or more audio output (speaker) and audio input (microphone) devices available.
Steps Description 10. The Audio Wizard: Test Speaker dialogue is displayed. Click the Test button and adjust the volume with the slider. The Test button changes to a Stop button. Click the Stop button once the proper volume for playback is achieved. Click the Next button to continue. 11. The Audio Wizard: Tune Microphone dialogue is displayed. Talk into the microphone and adjust the volume with the slider. Click the Next button to continue.
Steps Description 12. The Audio Wizard: Test Background Noise dialogue is displayed. Click on the Test button. Do not talk during this test. Click the Next button to continue. 13. The Congratulations dialogue is displayed. The configuration is complete. Click the Finish button to continue. JHB; Reviewed: PV 7/12/2006 Solution & Interoperability Test Lab Application Notes ©2006 Avaya Inc. All Rights Reserved. 38 of 47 Asterisk-AVAppN.
Steps Description 14. The Avaya one-X™ Desktop Edition dialogue is displayed again, with its configuration information populated. Before Avaya one-X™ Desktop Edition can be used with the Asterisk server, it must be optioned to use UDP to communicate with the server. Click the Settings button. JHB; Reviewed: PV 7/12/2006 Solution & Interoperability Test Lab Application Notes ©2006 Avaya Inc. All Rights Reserved. 39 of 47 Asterisk-AVAppN.
Steps Description 15. The Settings: Account dialogue is displayed. Click on Advanced. JHB; Reviewed: PV 7/12/2006 Solution & Interoperability Test Lab Application Notes ©2006 Avaya Inc. All Rights Reserved. 40 of 47 Asterisk-AVAppN.
Steps Description 16. The Settings: Advanced Options dialogue is displayed. Select Use UDP for the Communications Protocol. JHB; Reviewed: PV 7/12/2006 Solution & Interoperability Test Lab Application Notes ©2006 Avaya Inc. All Rights Reserved. 41 of 47 Asterisk-AVAppN.
Steps Description 17. Avaya one-X™ Desktop Edition can be configured to support Click to Dial using Microsoft Internet Explorer. Click Desktop Int. from the Settings dialogue. Enable the option Enable dialing from Internet Explorer. Click the Save button to continue. JHB; Reviewed: PV 7/12/2006 Solution & Interoperability Test Lab Application Notes ©2006 Avaya Inc. All Rights Reserved. 42 of 47 Asterisk-AVAppN.
Steps Description 18. A warning message is displayed indicating that Avaya one-X™ Desktop Edition must be restarted before the changes can take place. Click the OK button. 19. Click on the Avaya logo on the Avaya one-X™ Desktop Edition dialogue and a pop-up menu is displayed. Alternatively, right-click on the far right of the Dashboard to bring up the pop-up menu. Click Exit to exit Avaya one-X™ Desktop Edition.
7. Verification Steps All features shown in Table 3 that have a “Yes” in the Supported column were tested. Two problems were found. 1. The MWI feature, as implemented by the Asterisk Business Edition PBX, is not compatible with the Avaya 4600 Series IP Telephones nor with Avaya one-X™ Desktop Edition. Thus, MWI does not function in this combined solution. 2. On completion of an attended transfer, the caller ID information for the transferred call is not sent along with the call.
5. Verify that the Dialing Rules are properly configured (Settings Æ Dialing Rules). When improperly configured, external calls may be dialed incorrectly. From the same dialogue (Settings Æ Dialing Rules), enable the Display confirmation window before dialing a number to view the number that will be dialed before it is actually dialed. • Asterisk Business Edition PBX 1. Verify that the users have been entered into the system properly using the “sip show peers” in the Asterisk CLI (see Section 4.2, step #5).
9. Additional References The following are additional references. [1] [2] [3] [4] [5] [6] [7] [8] [9] [10] [11] [12] [13] [14] [15] [16] 4600 Series IP Telephone R2.3 LAN Administrator Guide, Issue 2.3, Doc ID 555-233507, April 2006, available at http://www.avaya.com/support. Avaya one-X™ Desktop Edition R2.1 Getting Started Guide, February 2006. Avaya one-X™ Desktop Edition Overview, April 2006, available at http://www.avaya.com/support.
©2006 Avaya Inc. All Rights Reserved. Avaya and the Avaya Logo are trademarks of Avaya Inc. All trademarks identified by ® and ™ are registered trademarks or trademarks, respectively, of Avaya Inc. All other trademarks are the property of their respective owners. The information provided in these Application Notes is subject to change without notice.
