Dev-scatch

Overview

Simple development scatch for myself.

Asterisk

No events for recording/live/pause

Because there's no events for pause/unpause, it's hard to know the recording was puased or not.

POST /channels/<channelId>/dtmf doing wrong

When it's getting the negative duration/between, it doesn't accept the call hangup.

POST /channels codec not in order

When the ARI application originating the channel with ARI, it doesn't respect the codec order in /etc/asterisk/pjsip.conf file.

Fixing Dial event

For the first Dial ARI event, it comes with dialstate: "", always.

This should be fixed to dialsatate: "initial" something.

The Dial application doesn't send the DialEnd event.

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This is fine. It was already explained.

Make reason for mandatory

Currently, the Asterisk doesn't set the reason field as a requirement. It makes hard to use the ChannelHangupRequest and ChannelDestroyed event.

Add more Hangup reason

Currently, DELETE /ari/channels/<channelID> supports only few hangup reason.

res/ari/resource_channels.c <source lang=c> if (ast_strlen_zero(args->reason) || !strcmp(args->reason, "normal")) { cause = AST_CAUSE_NORMAL; } else if (!strcmp(args->reason, "busy")) { cause = AST_CAUSE_BUSY; } else if (!strcmp(args->reason, "congestion")) { cause = AST_CAUSE_CONGESTION; } else if (!strcmp(args->reason, "no_answer")) { cause = AST_CAUSE_NOANSWER; } else if(!strcmp(args->reason, "answered_elsewhere")) { cause = AST_CAUSE_ANSWERED_ELSEWHERE; } else { ast_ari_response_error( response, 400, "Invalid Reason", "Invalid reason for hangup provided"); return; } </source>