Application Notes for Configuring SIP IP Telephony
Contents
1. AVAYA Avaya Solution amp 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 The Asterisk Business Edition PBX supports PBX telephony features standard SIP features and some supplementary SIP features The Avaya 4600 Series IP Telephones and Avaya one X Desktop Edition support the standard SIP features and some supplementary SIP features The Avaya one X Desktop Edition also supports presence and instant messaging Testing was conducted at the Avaya Solution and Interoperability Test Lab JHB Reviewed Solution amp Interoperability Test Lab Application Notes 1 of 47 PV 7 12 2006 2006 Avaya Inc All Rights Reserved Asterisk A V AppN doc 1 Introduction These Application Notes describe the configuration steps for using Avaya 4600 Series IP Telephones and Avaya one X Desktop Edition 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 configur2. 1s implemented by the Asterisk server but is not implemented by the Avaya phone device N C denotes that this feature is implemented by the Asterisk server but was not configured for the test No denotes that this feature is not implemented by the Asterisk server JHB Reviewed Solution amp Interoperability Test Lab Application Notes 4 of 47 PV 7 12 2006 2006 Avaya Inc All Rights Reserved Asterisk A V AppN doc Supported Feature Comments S S Ss gt Ss gt lt Series IP Telephones Supplementary Features lf CalHod Yes Ves 2 Consultation Hold Yes Yes 3 Music On Hold Ses Yes Unattended Transfer via the phone Yes Yes Unattended Transfer viatheserver Yes Yes s sp O O re lt 5 Attended Transfer e e Not applicable for Avaya 4600 Series IP Transfer Instant Messaging N N Telephones not supported by Avaya one X Desktop Edition Call Forward Unconditional via the server Yes Yes 7 Call Forward Unconditional via the Yes N A Not supported by the Avaya 4602SW or Avaya phone one X Desktop Edition Call Forward Busy via the server Yes Yes Call Forward Busy via the phone Not supported by the Avaya 4602SW or Avaya one X Desktop Edition Call Forward No Answer via the server Yes Yes Not supported by the Avaya 4602SW or Avaya one X Desktop Edition Call Forward No Answer via the phone
3. Asterisk A VAppN doc 4 An updated SIP Server License Server dialogue is displayed If the information was not gathered from the DHCP amp HTTP servers the fields would be blank and have to be entered them manually Click the Next button to continue Configuration SIP Server Licensing Server The SIP Softphone has found the following addresses for the SIP Server and Licensing 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 Click the Next button to continue You can create multiple profiles later to help you login using different types of connections JHB Reviewed Solution amp Interoperability Test Lab Application Notes 32 of 47 PV 7 12 2006 2006 Avaya Inc All Rights Reserved Asterisk A V AppN doc 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 Dialing Rules What number do you dial for an outside line 9 Your country code 1 Your area code What number do you dial for long distance 1
4. Notes HTTP services were used provided by Windows Internet Information Services IIS and the file was stored in the Inetpub wwwrooot folder on the Windows server PC The 46xxsettings txt file is a text file that can be edited with Windows Notepad Windows WordPad or other text editor Refer to Figure 2 and Table 4 in the following steps for information on the changes that are required If the telephone is running software prior to 2 2 HTTP is not supported and access to a TFTP server will be needed It is possible to need both a TFTP server and an HTTP HTTPS server until all phones have been upgraded to 2 2 JHB Reviewed Solution amp Interoperability Test Lab Application Notes 20 of 47 PV 7 12 2006 2006 Avaya Inc All Rights Reserved Asterisk A V AppN doc 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 KAFARA SIP Specific Settings HHHHEEEEEEEE HEHEHE HEH Use the following setting to configure SIP specific parameters Examples SET CALLFWDSTAT 3 SET CALLFWDDELAY 5 SET CALLFWDADDR cover avaya com SET COVERAGEADDR cover avaya com SET DIALPLAN A ooo T AAA boo 6 oo am SET DIALWAIT nee Dor M
5. Transfer and Call Park use the instructions in step 1 Restart the Asterisk server after making these changes JHB Reviewed Solution amp Interoperability Test Lab Application Notes 18 of 47 PV 7 12 2006 2006 Avaya Inc All Rights Reserved Asterisk A V AppN doc 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 amp 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 default exten gt lt extension gt hint SIP lt extension gt Edit the extensions conf file with the vi editor or other editor The following is an excerpt from the extensions conf file extensions conf exten a Macro seca ten cUU0Ul sir oo00l exten gt 60001 hint SIP 60001 Save the file and exit the editor Restart the Asterisk server Open the Asterisk CLI as documented in Section 4 2 step 5 Enter restart gracefully CLI gt restart gracefully Preparing for Asterisk restart Restarting Asterisk NOW Asterisk ABE A II Copyright C 1299 2005 Digium Written by Mark Spencer lt markster digium com gt Asterisk Ready Pa Mic All of the SIP features are configured and ready to us
6. VAppN doc l 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 To enable this feature the macro used for standard extensions macro stdexten must be modified to add references to t or T in the Dial command This command is further described in Reference 7 NOTE This will disable shuffling reinvites per the explanation in Reference 8 The explanation of the above options for the Dial command is as follows e t Allow the called user to transfer the call by hitting e T Allow the calling user to transfer the call by hitting Edit the extensions conf file with the vi editor or other editor The following is an excerpt from the extensions conf file used for the testing documented for these Application Notes NOTE Anything that follows a semi colon is a comment extensions conf Original entry without the additional parameters needed to turn on transfer sexten gt s 1 Dial S ARG2 20 Ring the interface 20 seconds maximum Dial command change co allow the calling or called party to transfer exten gt s 1 Dial ARG2 20 tT Ring the inter
7. What number do you dial for international calls 011 When you make a local call itis necessary to Dial your area code C Dial Dial number as is 17 Display confirmation windowbefore dialing a number JHB Reviewed Solution amp Interoperability Test Lab Application Notes 33 of 47 PV 7 12 2006 2006 Avaya Inc All Rights Reserved Asterisk A V AppN doc 7 The Voicemail Integration dialogue 1s 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 Dashboard 60006 lam Available gt TH Enter Name or Number yY O a lf PB O Communications History Contacts Figure 4 Registered Avaya one X Desktop Edition with Voice Mail Button JHB Reviewed Solution amp Interoperability Test Lab Application Notes 34 of 47 PV 7 12 2006 2006 Avaya Inc All Rights Reserved Asterisk A V AppN doc The Audio Wizard dialogue is displayed Click the Next button to continue Configuration JHB Reviewed Solution amp Interoperability Test Lab Application Notes 35 of 47 PV 7 12 2006 2006 Avaya Inc All Rights Reserved Asterisk A V AppN doc The Audio Wizard Select Soun
8. by the key For example 600004 Ext OK New 60000 Enter the password followed by the key For example 123456 Password tOr On successful registration the telephone s display will display something similar to the following 9 50pm 4 2 06 60000 The first line displays the date and the second line displays the extension number Repeat step 4 for each of the Avaya 4600 Series IP Telephones JHB Reviewed Solution amp Interoperability Test Lab Application Notes 23 of 47 PV 7 12 2006 2006 Avaya Inc All Rights Reserved Asterisk A V AppN doc 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 The steps for this configuration are documented in Reference 3 and further described in these Application Notes The configuration of Avaya one X Desktop Edition via the 46xxsettings txt file requires the use of a DHCP server and an HTTP server Avaya one X Desktop Edition uses the same DHCP and HTTP mechanism used by the Avaya 4600 Series IP Telephones see Section 5 These Applic
9. change this value using the interface on the phone DIALPLAN This parameter defines the dial plan used by the phone In this example 6xxxx defines that extensions will start with the number 5 and will have a length of 5 digits The phone will automatically dial the number once the 5 digit is entered SIPDOMAIN This parameter defines the SIP domain In this example asterisk com is used as the SIP domain SIPROXYSRVR This parameter is for the IP address of the proxy SIPREGISTRAR This parameter takes the same value as the proxy server SIPROXYSRVR NOTE The PHNUMOFSA and SPEAKERSTAT default values do not apply to the 4602SW The 4602SW telephone has a maximum of two 2 line appearances and the speakerphone works as a monitor or one way speaker The CALLFWDSTAT and CALLFWDADDR values also do not apply to the 4602SW NOLL 171 Default NOLL 7 Defaul No of Line Appearances PHNNUMOFSA 3 Default NOLL Defaul Table 4 Avaya 4600 Series IP Telephones SIP Parameters JHB Reviewed Solution amp Interoperability Test Lab Application Notes 22 of 47 PV 7 12 2006 2006 Avaya Inc All Rights Reserved Asterisk A V AppN doc 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
10. for information on the changes that are required 2 An excerpt of the 46xxsettings txt file with the SIP specific parameters is shown in Figure 3 A description of the important parameters is in Table 5 Edit these parameters to configure the SIP settings for Avaya one X Desktop Edition The sample in Figure 3 shows the values used in the compliance test NOTE Lines that start with are comments ERE EE AE E EE HHH HH HH HHH EEE EE EH HH EH HE HH E E E EE E E EH HH E E EE HH EE EE EE H Avaya one X Desktop Edition Variables FEAE EE AE E EE HHH HH HE HH HE EE EE EH HH EE EH EEE E E HH E E E E E E E EE EE EE H Examples tt Off SIPPROAYGRVE 1927 168 0 10 SET WEBLMSRVR 192 168 0 11 SET SP_DIRSRVR ldap east post avaya com SET SP_DIRSRVRPORT 389 SET SP_DIRTOPDN ou People o avaya com SET SP AC 212 SET LOCAL CALL PREFIX AC SET SIPPROXYSRVR SET PHNCC SET PHNDPLENGTH SET PHNIC SET PHNLD SET PHNLDLENGTH SET PHNOL SET SIPPROXYSRVR 10 2 2 60 SET WEBLMSRVR 10 2 2 70 Figure 3 Excerpt from Sample Avaya one X Desktop Edition 46xxsettings txt File JHB Reviewed Solution amp Interoperability Test Lab Application Notes 28 of 47 PV 7 12 2006 2006 Avaya Inc All Rights Reserved Asterisk A V AppN doc Table 5 shows the SIP specific parameters that can be configured The parameters that are critical to configure are SIPPROKYSRVR and WEBLMSRVR SIPPROXYSR
11. friendly method is to invoke the local call forwarding feature on the telephone see Section 5 step 3 Other examples for call forwarding are available in Reference 9 Call forwarding was enabled by modifying the macro used for standard extensions macro stdexten In this example all calls to all extensions are forwarded to extension 60007 Edit the extensions conf file with the vi editor or other editor The following is an excerpt from the extensions conf file used for the testing documented for these Application Notes NOTE Anything that follows a semi colon is a comment extensions conf Comment out the standard dial macro for extensions sexten gt s 1 Dial S ARG2 20 tT Ring the interface 20 seconds maximum gt Uneondire1 oneal forward co 60007 exten gt s 1 Dial SIP 60007 20 tT Ring the interface 20 seconds maximum Save the file and exit the editor Alternatively a single extension can be forwarded by modifying the definition of the extension in the default context In this example only calls to extension 60001 are forwarded unconditionally to extension 60007 Edit the extensions conf file with the vi editor or other editor The following is an excerpt from the extensions conf file used for the testing documented for these Application Notes NOTE Anything that follows a semi colon is a comment extensions conf default pexten 6G000L 1 Macro stdexten cU00l SIPO exten gt
12. in IETF RFC 3265 11 and IETF RFC 3842 Subscribe Notify method 12 will illuminate or extinguish a MWI lamp when new voice messages are left or cleared for an extension respectively This feature was tested using the built in voice mail system that comes with the Asterisk server The method supported by Asterisk for MWI is to notify the client via an unsolicited message As the Avaya devices support the Subscribe Notify method the MWI lamps were never illuminated or extinguished 3 3 Click to Dial While there is a click to dial application supported with Avaya one X Desktop Edition this implementation uses a direct link from Microsoft Internet Explorer using an Internet Explorer Browser Helper Object to Avaya one X Desktop Edition see Section 6 4 step 17 for the configuration of this feature This implementation is not the same as the Click to Dial feature defined for SIP 3 4 DIMF For these Application Notes SIP DTMF signaling defined in IETF RFC 2833 13 1s tested via interaction with the internal voice mail system The server based unattended transfer call park and call pickup features all use DTMF for feature activation DTMF was also tested as part of the testing of these features 3 5 Presence Tracking and Instant Messaging SIP Presence and Instant Messaging IETF RFC 3856 14 RFC 2779 15 and SIMPLE SIP for Instant Messaging and Presence Leveraging Extensions 16 are supported by Avaya one X Deskt
13. useful framework to describe product capabilities and compare features supported by various equipment vendors Table 3 provides a summary of SIP features supported on the Avaya 4600 Series IP Telephones and Avaya one X Desktop Edition when connected to the Asterisk Business Edition PBX Asterisk Business Edition PBX is a SIP registrar and acts as a back to back user agent B2BUA The Asterisk server can be configured to support most of the SIP features Based on information from Reference 6 it is possible to configure the server to support some of the remaining more advanced SIP features e g single line extension find me However the details in configuring these advanced SIP features are not readily available For that reason these features are not tested In Table 3 the features that were not tested due to lack of configuration are identified in the Comments column Section 4 Section 5 and Section 6 of these Application Notes describe the steps for configuring the Asterisk server and the Avaya clients to support the SIP features NOTE The features that were tested for these Applications Notes are listed as Yes in the Supported column in Table 3 Yes denotes that the feature is implemented by the Asterisk server and by the Avaya phone device Yes does not denote that the feature works Features that did not work are identified in the Comments column and in Section 7 N A denotes that this feature
14. 3 way conference 3 party added byuser_ Yes Yes rd 3 way conference 3 party calls and is Yes Yes joined by user 16 Cal Park Ves Ves 17 Cal Pick up Ves Yes faum anana TNA es Stree 19 Click to Dial via the phone N A Yes Not supported by the Avaya 4600 Series IP Telephones Other Features Asterisk supports unsolicited MWI notification This FA is incompatible with the Avaya 4600 Series IP Message Waiting Indicator MWI Yes Yes Telephones and Avaya one X Desktop Edition polo om ol a The Avaya devices support Subscribe Notify Limited support provided by Asterisk server not Presence Tracking supported by the Avaya 4600 Series IP Telephones 22 Instant Messaging No No Not supported by the Avaya 4600 Series IP Telephones Table 3 SIP Feature Support with Asterisk Business Edition PBX JHB Reviewed Solution amp Interoperability Test Lab Application Notes 5 of 47 PV 7 12 2006 2006 Avaya Inc All Rights Reserved Asterisk A V AppN doc 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 Message Waiting Indicator MWI Message Waiting Indicator MWI as defined
15. 60001 1 Dial SIP 60007 20 Save the file and exit the editor NOTE It is not possible to test regular calls or the other forwarding features with unconditional forwarding enabled This feature must be disabled before testing the other call features JHB Reviewed Solution amp Interoperability Test Lab Application Notes 16 of 47 PV 7 12 2006 2006 Avaya Inc All Rights Reserved Asterisk A V AppN doc 3 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 The following is an excerpt from the extensions conf file used for the testing documented for these Application Notes NOTE Anything that follows a semi colon is a comment extensions conf Comment out the standard dial macro for no answer exten gt s NOANSWER 1 Voicemail u ARG1 If unavailable send to voicemail w unavail announce Forward on no answer to 60007 exten gt s NOANSWER 1 Dial SIP 60007 20 tT If unavailable send to 60007 Save the file and exit the editor NOTE To test interaction with the voice mail system this feature must be disabled Configu
16. However for simplicity Avaya one X Desktop Edition acquires its license by connecting to the WebLM server installed on an SES server see Figure 1 6 2 Configure the DHCP Server for Avaya one X Desktop Edition OPTIONAL The DHCP server that supports the Avaya 4600 Series IP Telephones can also be used to support Avaya one X Desktop Edition as documented in Reference 1 The DHCP server used in this testing 1s provided with Microsoft Windows 2000 Server The examples shown in these Applications Notes are specific to this DHCP server implementation but also apply to other DHCP servers JHB Reviewed Solution amp Interoperability Test Lab Application Notes 24 of 47 PV 7 12 2006 2006 Avaya Inc All Rights Reserved Asterisk A V AppN doc Edit the DHCP option configured for the Avaya 4600 Series IP Telephones Start the DHCP server client from the Windows Control Panel Start gt Settings gt Control Panel Click on Administrative Tools Click on DHCP 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 0 2 2 x subnet EG DHCP a E Active JHB Reviewed Solution amp Interoperability Test Lab Application Notes 25 of 47 PV 7 12 2006 2006 Avaya Inc All Rights Reserved Asterisk A V AppN doc Additional parameters are shown for the 0 2 2 x DHCP scope on the lef
17. I by entering asterisk c login as admin aming LOY 2s 2250S password admin interop asterisk be admin S su Password root interop asterisk be root asterisk c The Asterisk CLI window is displayed Asterisk ABE A l Copyrigne C 1999 2005 Digium Written by Mark Spencer lt markster digium com gt Asterisk Ready CLI Enter sip show peers An entry will be displayed for all of the users that were defined If the user is registered the user s IP address is listed under the Host column CLI gt sip show peers Name username Host Dyn Nat ACL Mask Status HT286FXS 60014 Unspecified D ZOD Unmonitored 60013 0001 HT488FXS 60012 COATS OOL 00010 60010 60009 60009 60008 60008 60007 60007 60006 60006 60005 60005 60004 60004 60003 60003 60002 60002 60001 60001 14 sip peers i Ga eile ie online Unspecified Unspecified TORZE Unspecified Unspecified Unspecified Unspecified Unspecified Unspecified Unspecified Te a OG NEZA 10722105 ORo r ane Se wets ee ee we ees 2535 295 25e 255 2O MIE Zs Za 25957 235 2595 a sheer ole Unmonitored Unmonitored Unmonitored Unmonitored Unmonitored Unmonitored Unmonitored Unmonitored Unmonitored Unmonitored UnmoniLored Unmonitored Unmonitored The Asterisk server is ready to support SIP clients NOTE If the asterisk program is not running clients will not be able to register with the A
18. USICSRVER as SET MWISRVR NOVO eG Sa i SET PHNNUMOFSA USU SEID REGISTERWAIT sooo SET SIPDOMAIN Tavaya Com SEND SI PEROXYSRVR O2 06s 2028 SEN DIP PORT Poor Dal SIPREGISTRAR 192 168 09 SET SPEAKERSTAT 2 SET RICPMON MLO Looe sO oat RECPMONPORT S005 DIALPLAN 6xxxx SIPDOMAIN asterisk com SIPPROXYSRVR 10 2 2 60 SIPREGISTRAR 10 2 2 60 CALLFWDSTAT 7 CALLFWDADDR 60007 asterisk com Figure 2 Excerpt from Sample Avaya 4600 Series IP Telephone 46xxsettings txt File JHB Reviewed Solution amp Interoperability Test Lab Application Notes 21 of 47 PV 7 12 2006 2006 Avaya Inc All Rights Reserved Asterisk A V AppN doc 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 In this example all three buttons unconditional call forward Call Forward call forward on busy CF wrd Busy and call forward on no answer CFrwd DA are configured on the phone On the phones except the 4602SW the buttons Call Forward CFwrd Busy and CFrwd DA will appear CALLFWDADDR This parameter defines the address to which calls will be forwarded when one of the call forwarding features are invoked NOTE The user can
19. VR Enter the IP address of the Asterisk server WEBLMSRVR Enter the address of the WebLM server LDAP Director Topmost Beas Pp AreaCode DIAL AS 158 Default Table 5 Avaya one X Desktop Edition SIP Parameters JHB Reviewed Solution amp Interoperability Test Lab Application Notes 29 of 47 PV 7 12 2006 2006 Avaya Inc All Rights Reserved Asterisk A V AppN doc 6 4 Configure Avaya one X Desktop Edition After Avaya one X Desktop Edition is installed it must be configured Start Avaya one X Desktop Edition via the menu Start gt Programs gt Avaya SIP Softphone gt Avaya SIP Softphone 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 The main Avaya one X Desktop Edition dialogue and the Configuration dialogue are displayed Click the Next button to continue To complete the initial set up of the Avaya v Dashboard SIP Softphone we need you to answer a Profile few questions If you need assistance please contact your system administrator Username Sasa eae Password ui A s f a A In the future you can change this T settings T information in the Settings section of the lt e SIP Softphone gt Communications Click Next gt to continue gt History gt Contacts While the name of the product has changed to Avaya one X Desktop Edition the s
20. ation Notes will document only the steps to configure the DHCP server to support Avaya one X Desktop Edition and the modifications to the 46xxsettings txt file in support of Avaya one X Desktop Edition NOTE Using a DHCP and HTTP server to configure Avaya one X Desktop Edition is optional The information can be entered manually using the Avaya one X Desktop Edition configuration wizard 6 1 WebLM License Server Avaya one X Desktop Edition must connect to a WebLM license server and acquire a license from the license server Without this license Avaya one X Desktop Edition will not support the full set of SIP features The license server IP address can be entered manually using the Avaya one X Desktop Edition configuration wizard Alternatively the IP address of the license server can be entered as one of the downloadable options in the 46xxsettings txt file The WebLM server software can be installed on several operating systems including Microsoft Windows and Linux The WebLM software for these operating systems is available on the Avaya Support Center site http www avaya com support The installation of the WebLM software is described in Reference 3 but is not covered in these Application Notes WebLM 1s also installed as part of Avaya SIP Enablement Services SES As these Application Notes describe testing the Avaya phones on a third party server it is not expected that an SES would be part of the configuration
21. ation Protocol SIP RFC 3856 Available at http www iett org rfc rfic3856 txt 15 Instant Messaging Presence Protocol Requirements RFC 2779 Available at http www ietf org ric ric2779 txt 16 SIP for Instant Messaging and Presence Leveraging Extensions simple Available at http www ietf org html charters simple charter html JHB Reviewed Solution amp Interoperability Test Lab Application Notes 46 of 47 PV 7 12 2006 2006 Avaya Inc All Rights Reserved Asterisk A VAppN doc 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 The configurations technical data and recommendations provided in these Application Notes are believed to be accurate and dependable but are presented without express or implied warranty Users are responsible for their application of any products specified in these Application Notes Please e mail any questions or comments pertaining to these Application Notes along with the full title name and filename located in the lower right corner directly to the Avaya Solution amp Interoperability Test Lab at interoplabnotes list avaya com JHB Reviewed Solution amp Interoperability Test Lab Applicati
22. ation used in the test is shown in Figure 1 All components are physically connected to a single Avaya C363T PWR Converged Stackable Switch and are administered as a single subnet A PC provides HTTP HTTPS DHCP and TFTP server support The Avaya one X Desktop Edition clients run on separate PCs running Microsoft Windows XP Professional operating system The main difference among the four Avaya 4600 Series IP Telephones 4602S W 4610SW 4620SW amp 4621SW for SIP functionality is the number of line appearances supported by each phone two line appearances for the 4602SW and five for the 4610S W 4620SW and 4621SW The other major differences are that the 4610S W 4620SW and 4621SW telephones provide call log and speed dial applications and a variety of options to view and or modify the current settings for the phone The 4602SW does not support these applications and options Avaya one X Desktop Edition supports up to ten line appearances and provides a Windows based graphical user interface for access to call log speed dials and settings The configuration steps described in these Application Notes apply to all four models of the Avaya 4600 Series IP Telephones and to Avaya one X Desktop Edition Table 1 profiles the network management capabilities of the phones Avaya one X Desktop Edition is the new brand for Avaya SIP Softphone Therefore that name is used in these Application Notes to document the product H
23. d 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 Select a sound device from the pull down list to be used for the audio output Select a microphone from the pull down list to be used for the audio input Click the Next button to continue Audio Wizard Ensure that all applications that record or play sound are closed F i jd 5ig maTel C Major Audi JHB Reviewed Solution amp Interoperability Test Lab Application Notes 36 of 47 PV 7 12 2006 2006 Avaya Inc All Rights Reserved Asterisk A V AppN doc 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 Ensure that your speakers or headphones Ensure that your speakers or headphones are connected and turned on are connected and turned on Click Test to play an audio sample to Click Test to play an audio sample to verify that your playback configuration is verify that y
24. e JHB Reviewed Solution amp Interoperability Test Lab Application Notes 19 of 47 PV 7 12 2006 2006 Avaya Inc All Rights Reserved Asterisk A V AppN doc 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 323 software loaded must be optioned to install the SIP software by modifying the SIG value to sip press lt MUTE gt S I G on the telephone keypad and restarting the telephone The installation of the telephone software and the configuration of the telephone require access to a TFTP server or to an HTTP HTTPS server These Application Notes will not provide the details of the administration of Avaya 4600 Series IP Telephones see Reference 1 These Application Notes will provide details of the specific SIP related configuration items Locate the 46xxsettings txt file to administer the SIP specific settings for the Avaya 4600 Series IP Telephones This file is stored on the PC that provides TFTP or HTTP HTTPS services to the telephones The location of this file depends on the software that is used to provide these services For these Application
25. e 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 Dashboard Profile Lab era 0006 asterisk com Password _ settings t Communications History Contacts JHB Reviewed Solution amp Interoperability Test Lab Application Notes 39 of 47 PV 7 12 2006 2006 Avaya Inc All Rights Reserved Asterisk A V AppN doc 15 The Settings Account dialogue is displayed Click on Advanced Mm Settings Account JHB Reviewed Solution amp Interoperability Test Lab Application Notes 40 of 47 PV 7 12 2006 2006 Avaya Inc All Rights Reserved Asterisk A V AppN doc 16 The Settings Advanced Options dialogue is displayed Select Use UDP for the Communications Protocol Mm Settings Advanced Options JHB Reviewed Solution amp Interoperability Test Lab Application Notes 41 of 47 PV 7 12 2006 2006 Avaya Inc All Rights Reserved Asterisk A VAppN doc 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 x Settings sktop Integration E E E M ai JHB Reviewed Solution Interoperability Test Lab Application Notes 42 of 47 PV 7 12 2006 2006 Avaya Inc All Rights Reserved Asteris
26. ebLM license server and acquire a license Otherwise only basic telephony make amp receive calls will work From the local PC open a Command Prompt window and ping the license server From the license server verify that there are available SIP Softphone licenses installed JHB Reviewed Solution amp Interoperability Test Lab Application Notes 44 of 47 PV 7 12 2006 2006 Avaya Inc All Rights Reserved Asterisk A V AppN doc 5 Verify that the Dialing Rules are properly configured Settings gt Dialing Rules When improperly configured external calls may be dialed incorrectly From the same dialogue Settings gt Dialing Rules enable the Display confirmation window before dialing a number to view the number that will be dialed before it is actually dialed e Asterisk Business Edition PBX l 2 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 Verify the dial plan configured in the sip conf file is configured properly to support attended transfers call park shuffling and call forwarding Refer to Section 4 3 for example configurations Also view the dial plan using the show dialplan command in the Asterisk CLI If there are problems with receiving calls verify that call forwarding has not been configured and enabled on the server This may impact receiving calls depending on which forwarding feature is configured see S
27. ection 4 3 Verify that the Asterisk server is running if the telephones are unable to register Use the instructions in see Section 4 2 step 5 to start the server If there are problems dialing to interact with external voice mail systems external IVR systems or any other system that used verify whether the Dial command has been configured to allow to initiate an internal unattended transfer see Section 4 3 step 1 8 Conclusion These Application Notes have described the administration steps required to use Avaya 4600 Series IP Telephones and Avaya one X Desktop Edition with the Asterisk Business Edition PBX Both standard and supplementary features were covered for all of the Avaya 4600 Series IP Telephones and Avaya one X Desktop Edition and presence tracking was covered for Avaya one X Desktop Edition JHB Reviewed Solution amp Interoperability Test Lab Application Notes 45 of 47 PV 7 12 2006 2006 Avaya Inc All Rights Reserved Asterisk A V AppN doc 9 Additional References The following are additional references 1 4600 Series IP Telephone R2 3 LAN Administrator Guide Issue 2 3 Doc ID 555 233 507 April 2006 available at http www avaya com support 2 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 3 Avaya one X Desktop Edition Administration Ap
28. face 20 seconds maximum Save the file and exit the editor NOTES 1 The Call Park feature is implemented as an Unattended Transfer to a known extension Therefore the above configuration is needed also for the Call Park feature 2 Enabling Unattended Transfer will disable shuffling To test shuffling use the instructions in step 7 Restart the Asterisk server after making these changes 3 Enabling the Unattended Transfer feature may interfere with external voice mail systems external IVR systems or any external system where the use of is important The Asterisk system will automatically start an Unattended Transfer when is pressed JHB Reviewed Solution amp Interoperability Test Lab Application Notes 14 of 47 PV 7 12 2006 2006 Avaya Inc All Rights Reserved Asterisk A V AppN doc Zi Configure the server for Call Pickup 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 The following is an excerpt from the features conf file used for the testing documented for these Application Notes NOTE Anything that follows a semi colon is a comment feat
29. ility to complete transfers If there are problems with receiving calls verify that call forwarding has not been locally configured This may impact receiving calls depending on which forwarding feature is configured This configuration is done in the 46xxsettings txt configuration file CALLFWDADDR and COVERAGEADDR parameters As the Avaya 4600 Series IP Telephones running the SIP software cannot dial properly verify that the dial plan does not include dialing a For example the default number to dial for call pickup for an Asterisk server is 8 This must be changed to a different number e Avaya one X Desktop Edition l Verify that the Username field for Avaya one X Desktop Edition is in the format of userID domain name instead of userID IP address e g john doe company org instead of john doe 123 22 33 444 If an IP address is used this may affect the ability to complete transfers As the Avaya one X Desktop Edition cannot dial properly verify that the dial plan does not include dialing a For example the default number to dial for call pickup for an Asterisk server is 8 This must be changed to a different number Verify that Avaya one X Desktop Edition has been configured for UDP Settings gt Advanced gt Communications Protocol Otherwise it will not register with the server Verify that Avaya one X Desktop Edition is able to connect to a W
30. k A V AppN doc A warning message is displayed indicating that Avaya one X Desktop Edition must be re started before the changes can take place Click the OK button Some of your changes cannot be applied while the application is running To apply your changes lease restart SIP Softph 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 Dashboard JHB Reviewed Solution amp Interoperability Test Lab Application Notes 43 of 47 PV 7 12 2006 2006 Avaya Inc All Rights Reserved Asterisk A V AppN doc 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 The following steps can be used to verify and or troubleshoot installations in the field e Avaya 4600 Series IP Telephones l 2 Verify that the SIPDOMAIN parameter is configured in the 46xxsettings txt configuration file as this may affect ab
31. nf file The following parameters must be defined to configure a user lt extension number gt comment text after a semi colon are comments type friend host dynamic username lt extension number gt secret lt password gt callerid lt caller ID string gt inthe form of caller ID name lt caller ID gt 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 conf 60001 type friend host dynamic username 60001 secret 60001 iS callerid Avaya 4602SW lt 60001 gt password Save the file and exit the editor The other configuration is in the extensions conf file The following parameters must be set to configure a user default exten gt someezten priority applicationf argl arg2 The following is an excerpt from the extensions conf file used for the testing documented for these Application Notes for the same extension listed above in the sip conf file extensions conf default exten gt 60001 1 Macro stdexten 60001 SIP 60001 Save the file and exit the editor 10 of 47 JHB Reviewed Solution amp Interoperability Test Lab Application Notes Asterisk A V AppN doc PV 7 12 2006 2006 Avaya Inc All Rights Reserved 4 Configure the users for voice mail The configuration of the voice mail users is done in two files e etc aste
32. nf 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 NOTE Anything that follows a semi colon is a comment sip conft 60001 type friend username 60001 secret 60001 mailbox 60001 default callerid Avaya 4602SW lt 60001 gt host dynamic pickupgroup 2 canreinvite yes turn shuffling on Save the file and exit the editor In addition to turn shuffling on the macro used for standard extensions macro stdexten in the extensions conf file must be modified to remove any references to t or T in the Dial command see Reference 7 NOTE This will disable unattended transfer as supported by the server Also shuffling must be turned on for both extensions that are involved in a conversation The following is an excerpt from the extensions conf file used for the testing documented for these Application Notes NOTE Anything that follows a semi colon is a comment extensions conf Entry without the additional parameters needed to turn off transfer exten gt s 1 Dial ARG2 20 Ring the interface 20 seconds maximum Entry with the additional parameters needed to turn on transfer excen gt 8 lpDial S ARGZ 20 11 Ring the interface 20 seconds maximum Save the file and exit the editor NOTE Enabling Shuffling will disable Unattended Transfer and Call Park To test Unattended
33. oftware used for testing still refers to Avaya SIP Softphone This includes the installation folder and Windows program group JHB Reviewed Solution amp Interoperability Test Lab Application Notes 30 of 47 PV 7 12 2006 2006 Avaya Inc All Rights Reserved Asterisk A VAppN doc 2 The Account dialogue is displayed 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 iF The SIP Server License Server dialogue is displayed If the DHCP server was configured properly in Section 6 2 and the 46xxsettings txt file has the proper information as described in Section 6 3 Avaya one X Desktop Edition will get the IP address of the proxy server and the license server Click the Next button to continue SIP Server Licensing Server SIP Softphone will try to find the SIP Server and Licensing Server This may take a minute Skip this step am not using a SIP Server i e am using Peer to Peer communication JHB Reviewed Solution amp Interoperability Test Lab Application Notes 31 of 47 PV 7 12 2006 2006 Avaya Inc All Rights Reserved
34. on Notes 47 of 47 PV 7 12 2006 2006 Avaya Inc All Rights Reserved Asterisk A VAppN doc
35. op Edition but not by the Avaya 4600 Series IP Telephones The Asterisk server does not support instant messaging but does support presence tracking As stated in Table 3 the Asterisk server has limited support for presence tracking Presence can be tracked for all stations not just limited to Softphone stations but the server can only send one of three states e the station is on line and idle registered Available e the station is off line un registered Offline e the station is on line and on the phone On the Phone JHB Reviewed Solution amp Interoperability Test Lab Application Notes 6 of 47 PV 7 12 2006 2006 Avaya Inc All Rights Reserved Asterisk A V AppN doc 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 nvisible 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 1s busy Busy The states that both the Asterisk server and Avaya one X Desktop Edition support are Available Offline and On the Phone JHB Reviewed Solution amp Interoperability Test Lab Application Notes 7 of 47 PV 7 12 2006 2006 Avaya Inc All Rights Reserved A
36. our playback configuration is correct correct Adjust the slider below to set the desired Adjust the slider below to set the desired volume volume Talk into the microphone and adjust the volume with the slider Click the Next button to continue Audio Wizard Ensure that your microphone is plugged in and turned on Speak into the microphone a JHB Reviewed Solution amp Interoperability Test Lab Application Notes 37 of 47 PV 7 12 2006 2006 Avaya Inc All Rights Reserved Asterisk A V AppN doc 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 Audio Wizard Click Test to determine the normal noise levels at your current location This test bapa prevent Avaya SIP 3 background Please do not cover your microphone or talk during the test Re run this test if Uausual noise levels occur during tho test 13 The Congratulations dialogue is displayed The configuration is complete Click the Finish button to continue Your initial set up is complete Welcome to SIP To start using SIP Softphone click on Finish JHB Reviewed Solution amp Interoperability Test Lab Application Notes 38 of 47 PV 7 12 2006 2006 Avaya Inc All Rights Reserved Asterisk A V AppN doc The Avaya one X Desktop Edition dialogue is displayed again with its configuration information populated Befor
37. owever the version of the product that was used for testing still has internal references to Avaya SIP Softphone or SIP Softphone JHB Reviewed Solution amp Interoperability Test Lab Application Notes 2 of 47 PV 7 12 2006 2006 Avaya Inc All Rights Reserved Asterisk A V AppN doc Cis 3745 R 10 2 Avaya one Desktop _ Edition 2 Figure 1 Network Configuration Diagram Avaya 4600 Series Avaya one X IP Telephones Desktop Edition a i Manual Administration mechanisms Configuration files wa Configuration files TFTP HTTP _ Avaya one X Table 1 Network Management Capabilities of 46 Series IP Telephones amp Avaya one X M Desktop est 10 dition 1 Avaya Avaya Ava 4602SW 4610SW 4620 JHB Reviewed Solution amp Interoperability Test Lab Application Notes 3 of 47 PV 7 12 2006 2006 Avaya Inc All Rights Reserved Asterisk A V AppN doc 2 Eguipment and Software Validated The following eguipment and software were used for the sample configuration provided Eguipment Avaya C363T PWR Converged Stackable Switch 4 5 14 Avaya 4602SW 4610SW 4620SW 4621SW IP 2 2 2 SIP Telephones Table 2 Equipment and Version 3 Supported SIP Features In addition to standard calling capabilities 1 e make a call answer a call drop a call the Internet Engineering Task Force IETF has defined a supplementary set of calling features often referred to as the SIPPING 19 10 This reference provides a
38. r F Change the directory to etc asterisk root interop asterisk be root ed etc asterisk root interop asterisk be asterisk Continue to configure the SIP domain JHB Reviewed Solution amp Interoperability Test Lab Application Notes 8 of 47 PV 7 12 2006 2006 Avaya Inc All Rights Reserved Asterisk A V AppN doc Configure the SIP domain 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 con files will be in the following format context1 parameteri value parameter2 value To configure the SIP domain place the following parameters in the general context domain lt domain name gt fromdomain lt domain name gt The following is an excerpt from the sip conf file used for the testing documented for these Application Notes sip conft general domain asterisk com fromdomain asterisk com Save the file and exit the editor JHB Reviewed Solution amp Interoperability Test Lab Application Notes 9 of 47 PV 7 12 2006 2006 Avaya Inc All Rights Reserved Asterisk A V AppN doc 3 Configure users on the Asterisk server The configuration of the users is done in two files e etc asterisk sip conf e etc asterisk extensions conf The majority of the configuration is in the sip co
39. re the server for Call Forwarding Busy In this example all calls are forwarded to extension 60007 Normally calls 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 The following is an excerpt from the extensions conf file used for the testing documented for these Application Notes NOTE Anything that follows a semi colon is a comment extensions conf Comment out the standard dial macro for busy exten gt s BUSY 1 Voicemail b ARG1 If busy send to voicemail w busy announce Forward on busy to 60007 exten gt s BUSY 1 Dial SIP 60007 20 tT If busy send to 60007 Save the file and exit the editor NOTE To test interaction with the voice mail system this feature must be disabled JHB Reviewed Solution amp Interoperability Test Lab Application Notes 17 of 47 PV 7 12 2006 2006 Avaya Inc All Rights Reserved Asterisk A V AppN doc 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 lt extension gt canreinvite yes no shuffle or don t shuffle Edit the sip co
40. ril 2006 available at http www avaya com support 4 Asterisk Business Edition Technical Reference Version A Available at http www digium com en products software abe php 5 Asterisk Business Edition QuickStart Guide Available at http www digium com en products software abe php 6 Asterisk WiKi page on voip info org http www voip info org wiki Asterisk 7 Asterisk cmd Dial Available at http www voip info org wiki index php page Asterisk cmd Dial 8 Asterisk SIP media path Available at http www voip info org wiki view Asterisk SIP media path 9 Asterisk call forwarding Available at http www voip info org wiki view Asterisk call forwarding 10 Session Initiation Protocol Service Examples draft ietf sipping service examples 10 SIPPING Working Group Internet Draft Mach 5 2006 expires September 6 2006 available at http tools 1etf org wg sipping draft ietf sipping service examples draft iet sipping service examples 10 txt 11 Session Initiation Protocol SIP Specific Event Notification RFC 3265 Available at http www ietf org rtc ric3265 txt 12 A Message Summary and Message Waiting Indication Event Package for the Session Initiation Protocol SIP RFC 3842 Available at http www ietf org ric ric3842 txt 13 RTP Payload for DTMF Digits Telephony Tones and Telephony Signals RFC 2833 Available at http www iett org ric rfc2833 txt 14 A Presence Event Package for the Session Initi
41. risk sip conf e etc asterisk voicemail conf The following parameters must be set in sip conf to configure a user for voice mail lt extension number gt mailbox lt extension number gt 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 conft 60001 This iS a comment 7 type friend host dynamic username 60001 gt Thais 12s another comment secret 60001 callerid Avaya 4602SW lt 60001 gt mailbox 60001 The following parameters must be defined in voicemail conf to configure a user for voice mail default lt extension number gt gt lt extension number gt lt caller ID name gt Edit the voicemail conf file with the vi editor or other editor The following is an excerpt from the voicemail conf file used for the testing documented for these Application Notes voicemail conf default 60001 gt 60001 Avaya 4602SW Save the file and exit the editor JHB Reviewed Solution amp Interoperability Test Lab Application Notes 11 of 47 PV 7 12 2006 2006 Avaya Inc All Rights Reserved Asterisk A V AppN doc 3 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 CL
42. sterisk A V AppN doc 4 Configure the Asterisk Business Edition PBK 4 1 Install Asterisk Business Edition PBK 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 Administer the Asterisk Server The following steps describe configuration of the Asterisk Business Edition PBX for use with Avaya SIP clients The administration of the server is covered in Reference 4 The installation of the server is covered in References 4 and 5 For additional information and examples on configuring of the server see Reference 6 The Asterisk Business Edition PBX is configured by editing the following configuration files on the Linux system e etc asterisk sip conf e etc asterisk extensions conf e etc asterisk features conf e etc asterisk voicemail conf First log into the Linux system using the administration account admin that was created when Linux was installed Enter the password for the administration account login as admin admin loecuc 00 S password admin interop asterisk be admin Log in as the root account by entering su Enter the password for the root account admin interop asterisk be admin S su Password root interop asterisk be roo
43. sterisk server JHB Reviewed PV 7 12 2006 2006 Avaya Inc All Rights Reserved Solution amp Interoperability Test Lab Application Notes 12 of 47 Asterisk A V AppN doc 4 3 Administer the Supplementary Features on the Asterisk Server Additional administration is needed to support the following features e Unattended transfer via the server e Call park e Call pickup e Call forwarding e Shuffling Reinvite See Reference 8 e Presence tracking NOTE Some of the features listed above can be configured in one sequence However there are some features e g Unattended Transfer amp Shuffling Call Forward Unconditional amp Call Forward Busy that are mutually exclusive The steps listed below document how to configure all of the features At the end there is the step to restart the Asterisk server In some cases enabling one feature will disable another feature Therefore when it is time to test such features the change should be made as described in the appropriate steps below followed by restarting the server Similar to Communications Manager CM Asterisk is involved in the media path on every call Also similar to CM Asterisk can be optioned to shuffle calls so that the media goes directly from one client to the other This is referenced within Asterisk as reinvite JHB Reviewed Solution amp Interoperability Test Lab Application Notes 13 of 47 PV 7 12 2006 2006 Avaya Inc All Rights Reserved Asterisk A
44. t 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 gt Properties LO DHCP Action Yew e gt OlmM X she Tree Scope Options Option Name DHCP Standard 10 2 2 1 S E Scope 10 2 2 0 10 2 2 x Poso UNIS Serve Standard 130 1 1 200 fas Address Pool 76 46xxOptions TFTPSRYR 10 2 2 102 HTTPSRYR None GB Address Leases JHB Reviewed Solution amp Interoperability Test Lab Application Notes 26 of 47 PV 7 12 2006 2006 Avaya Inc All Rights Reserved Asterisk A V AppN doc 4 The Scope Options dialogue 1s displayed The value for the 7176 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 trtpsrv 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 2 102 Click the OK button to save the change Scope Options Aa AA General Advanced _ Available Description O 074 Internet Relay Chat IRC Servers List of IRC O O75 StreetT alk Servers List of Stree ain Oprec
45. tory Assistance STDA Servers List of STDS AEs i EA a r Data entry String value FT PSAVA 10 2 2 102 HT TPSAVR 10 2 2 102 5 Refresh the DHCP server to use the updated scope options To refresh the options select the scope option 176 46xxOptions Then from the menu select Action gt Refresh JHB Reviewed Solution amp Interoperability Test Lab Application Notes 27 of 47 PV 7 12 2006 2006 Avaya Inc All Rights Reserved Asterisk A V AppN doc 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 l Locate the 46xxsettings txt file to administer the SIP specific settings for Avaya one X Desktop Edition This file is stored on the PC that provides HTTP services to the telephones The location of this file depends on the software that is used to provide these services For these Application Notes the HTTP services were provided by Windows Internet Information Services IIS and the 46xxsettings txt file was stored on the Inetpub wwwrooot folder on the Windows server PC The 46xxsettings txt file is a text file that can be edited with Windows Notepad Windows WordPad or other text editor Refer to Figure 3 and Table 5 in the following steps
46. ures conf pickupexten 8 Configure the pickup extension Default is 8 pickupexten 200 Change the pickup extension to 200 Save the file and exit the editor 3 Configure the user for Call Pickup In addition to configuring the server to support the Call Pickup feature the users have to be placed in a pickup group This is done in the sip conf file The following parameters must be set in sip conf lt extension gt pickupgroup lt pickup group gt The following is an excerpt from the sip conf file used for the testing documented for these Application Notes sip conft LOOOL type friend username 60001 secret 60001 maillbox 60001 default callerid Avaya 4602SW lt 60001 gt host dynamic pickupgroup 2 Save the file and exit the editor JHB Reviewed Solution amp Interoperability Test Lab Application Notes 15 of 47 PV 7 12 2006 2006 Avaya Inc All Rights Reserved Asterisk A V AppN doc 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 However this is not the method that would be used in a production system as there is no easy way for the user to control how calls are forwarded A user
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