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Acceptance Tests For Asterisk on the MCF54451

Acceptance Tests For Asterisk on the MCF54451. DRAFT – VERSION 0.5 January 6/08. Testing SIP Client. Test Setup Procedure A – Zoiper (sip client) registered to SIP extension on asterisk – (following steps described in setup document) B – Zoiper (sip client)

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Acceptance Tests For Asterisk on the MCF54451

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  1. Acceptance Tests For Asterisk on the MCF54451 DRAFT – VERSION 0.5 January 6/08

  2. Testing SIP Client Test Setup Procedure A – Zoiper (sip client) • registered to SIP extension on asterisk – (following steps described in setup document) B – Zoiper (sip client) • registered to second SIP extension on asterisk – (following steps described in setup document) Test Operation Confirm registration of A and B under Asterisk GUI • (following steps described in setup document) Confirm registration of each Zoiper client • (following steps described in setup document) A calls B -> confirm ringback on A -> confirm B incoming call pop-up window appears Answer B -> confirm full-duplex audio path -> hang up B -> confirm A and B call cleared normally -> confirm registration is maintained Repeat for B calls A scenario

  3. FXS – Testing Inbound Test Setup Procedure A – Zoiper (sip client) • registered to SIP extension on asterisk – (following steps described in setup document) B – FXS Handset Attached to Port “Phone” • FXS analog channel configured as extension (following steps described in setup document) Test Operation A calls B -> confirm ringing Take handset off-hook -> confirm duplex audio path Put handset on-hook -> confirm registration is maintained

  4. FXS – Testing Outbound Test Setup Procedure A – Zoiper (sip client) • registered to SIP extension on asterisk – (following steps described in setup document) B – FXS Handset Attached to Port “Phone” • FXS analog channel configured as extension - (following steps described in setup document) Test Operation Take B off-hook -> confirm dial tone B calls A -> confirm A “ringing” and B “ringback” tone Answer A -> confirm full duplex audio path Hangup A and B -> confirm registration is maintained

  5. FXO Functional Testing - Overview Test Setup Procedure The purpose of this test is to validate ringing, hook state and audio. • Create a Time Interval • Set up incoming call rule to route FXO – to – SIP client • Call FXO line • Answer SIP Extension • Confirm duplex audio path • Terminate test and validate completion

  6. FXO Functional Testing– Setup Time Interval 2. Name Interval ‘ALL’ 3. Set for Mon-to-Sun 1. Select Call Time Interval Menu 4. Check “Entire Day” 5. Click update APPLY CHANGES – VERY IMPORTANT

  7. FXO Functional Testing - Set Up Call Rule 2. Select FXO Trunk 3. Select Time Interval – “ALL” 1. Select Call Rules Menu 4. Destination 6000 (SIP User Extension) 5. Click update APPLY CHANGES – VERY IMPORTANT

  8. FXO Functional Testing - Procedure Test Setup Procedure Follow Steps to setup time interval and set up call rule A – Zoiper (sip client) • registered to SIP extension 6000 on asterisk – (following steps described earlier) B – PSTN Analog Phone Line connected to FXO (line) port of 54451 board • FXO configured as analog trunk (described earlier) • FXO configured to route call to SIP destination 6000 C – PSTN Phone Test Operation C calls PSTN Number B -> Confirm ringback on C -> confirm call is ringing through to A (may take up to 3 rings) Answer A -> Confirm full duplex voice Hangup C and A -> confirm A registration is maintained -> retry to confirm calls cleared normally and state is idle

  9. Notice This Completes Acceptance testing The following section describes how to setup and demo Asterisk features – these features are not supported as part of the project requirements. This documentation has been provided to help with field demos and training.

  10. Cool Demo’s • The capabilities of Asterisk are very complete and most functions have been made to work on the MCF54451-EVB. • Dialing by name • Music on hold • Voicemail • Creating a Ring Group

  11. Dialing by Name Setup Procedure The dialing by name demo sets up the FXO inbound port as a trunk line that gets answered by the Asterisk directory attendant. The directory attendant allows users to spell the name of the person they are trying to reach using their telephone key pad. • Follow previous steps to setup FXO trunk • Follow previous steps to Create a Time Interval • Set up Directory Extension • Change incoming call rule to route FXO – to – Directory Extension • Confirm User Capabilities Testing • Call FXO line • Dial by name

  12. Set Up Directory Extension 2. Configure Directory Extension 6998 3. Check: -Read extension number -Use first name 1. Select Directory Menu 4. Click save APPLY CHANGES – VERY IMPORTANT

  13. Confirm User Capabilities 2. Confirm Name is defined 1. Select User Menu + click user 3. Confirm Directory Capability is set

  14. Modify Incoming Call Rules 2. Click edit 1. Select Incoming Calling Rules menu 4. Select Names Directory 5. Click Update APPLY CHANGES – VERY IMPORTANT Page: 14

  15. Music on Hold Setup Procedure Music on Hold (MoH) plays back music when a call when a call is put on hold. MoH is enabled by default and can be demo’d easily • Follow Previous Steps to Setup User SIP User • Follow Previous Steps to Configure FXS Port • Follow Previous Steps to Configure Zoiper SIP Client Testing • Dial extension / FXS extension from Zoiper SIP Client • Place extension on hold Hold Button

  16. Voicemail Setup Procedure Voicemail allows callers to leave voice messages for extensions that are unavailable. It also allows users to check their voicemail by dialing the voicemail port extension and entering a password. Voicemail also supports: time stamping, envelope stamping, operator assistance, message review capability, message length, message size, mailbox size and email voicemail. • Follow Previous Steps to Setup User SIP User • Follow Previous Steps to Configure FXS Port • Follow Previous Steps to Configure Zoiper SIP Client • Configure General Voicemail Settings • Confirm User Capabilities Testing • Dial extension / FXS extension from Zoiper SIP Client • Leave message • To retrieve messages dial 6999 and enter mailbox / password

  17. Configuring General Voicemail Settings 2. Setup VM Extension 6999 3. Use basic settings 1. Select Voicemail Menu 4. Configure Advanced options 6. Save Settings APPLY CHANGES – VERY IMPORTANT

  18. Confirm User Capabilities 1. Select User Menu + click user 3. Confirm Voicemail Capability is set & Default PIN APPLY ANY CHANGES – VERY IMPORTANT

  19. Creating a Ring Group Setup Procedure Ring Groups allow one extension number to ring in multiple places. This can occur simultaneously or in sequence. This is useful for building systems, medical monitoring and other push-to-call applications because it allows for multiple answer points via single number. • Follow Previous Steps to Setup User SIP Users • Follow Previous Steps to Configure FXS Port • Follow Previous Steps to Configure Zoiper SIP Client • Configure Ring Group Settings Testing • Dial Ring Group extension 6400 • All registered users should ring / answer single user

  20. Configure Ring Group Settings Note: user’s without names will not appear in the available user’s list 2. Name ring group 3. Set up ring group extension 6400 4. Load available users into Ring Group Members 1. Select Ring Groups Menu 4. Select Ring all Simultaneously 5. Configure if not answered goto… 6. Save Settings APPLY CHANGES – VERY IMPORTANT

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