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7) Voice can be audible clearly between two clients.

Manual Asterisk Server setup in Banana PI R4 without adding configuration changes of hal-voice-asterisk code. 

1) Added below changes for porting asterisk code for banana pi r4 board.

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10) Now initiate an outbound call from internal extension to the external sip server(2931).

-- Executing [92931@from-internal:1] NoOp("PJSIP/1001-00000001", "") in new stack
    -- Executing [92931@from-internal:2] Dial("PJSIP/1001-00000001", "PJSIP/2931@linphone_freesip_trunk,60") in new stack
    -- Called PJSIP/2931@linphone_freesip_trunk
    -- PJSIP/linphone_freesip_trunk-00000002 is ringing
    -- Added contact 'sip:1001@192.168.1.9:36312;transport=UDP;rinstance=faa99161e70850c2' to AOR '1001' with expiration of 60 seconds
[Jul 16 11:20:45] WARNING[1830]: db.c:316 db_execute_sql: Error executing SQL (COMMIT): database is locked
       > 0x7f940598d0 -- Strict RTP learning after remote address set to: 5.135.215.43:18034
    -- PJSIP/linphone_freesip_trunk-00000002 answered PJSIP/1001-00000001
       > 0x7f94049790 -- Strict RTP learning after remote address set to: 192.168.1.9:36687
    -- Channel PJSIP/linphone_freesip_trunk-00000002 joined 'simple_bridge' basic-bridge <97c41ec6-e3f4-45e8-ad94-f2f8763abff7>
    -- Channel PJSIP/1001-00000001 joined 'simple_bridge' basic-bridge <97c41ec6-e3f4-45e8-ad94-f2f8763abff7>
    -- Channel PJSIP/linphone_freesip_trunk-00000002 left 'simple_bridge' basic-bridge <97c41ec6-e3f4-45e8-ad94-f2f8763abff7>
    -- Channel PJSIP/1001-00000001 left 'simple_bridge' basic-bridge <97c41ec6-e3f4-45e8-ad94-f2f8763abff7>
  == Spawn extension (from-internal, 92931, 2) exited non-zero on 'PJSIP/1001-00000001'
    -- Added contact 'sip:1001@192.168.1.9:36312;transport=UDP;rinstance=faa99161e70850c2' to AOR '1001' with expiration of 60 seconds 

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SIP registration commands and phone call mechanism after adding the code changes in BPI R4:

Image Added

1) By default, inbound call configuration will be updated in the device. Please refer 11th point of Manual Asterisk Server setup in Banana PI R4 without adding configuration changes of hal-voice-asterisk code for inbound call mechanism using sip clients.

2) No need to start the systemd service as the process is started by hal-voice-asterisk with the existing configuration.

3) No need to add firewall rules to support SIP and RTP ports. We handled the same in utopia and rules will be updated whenever device boots up.

4) Please take two clients install an application which supports voice calls and register those clients with the configuration in the "/etc/asterisk" folder.

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    Usename: 602

    Password: 602

35) Please set below commands from the console to update the outbound configuration. 

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      dmcli eRT getv Device.Services.VoiceService.1.VoiceProfile.1.Line.1.Status

46) Please take some other client and create an account in sip.linphone.org

57) Provide the username and password in the data models as per the data used for account creation using linphone.

8) Initiate outbound call as mentioned in the 10th point of Registering Asterisk as client with external SIP server and making call using external sip server trunk

96) Added below logs for the reference.

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