Can someone decode this for me?

General Help
1

I have a system with Ghost calling. An extension will dial another extension, wait for it to go off hook and then hang up. This might happen 6 times in a 24 hour period and most times after hours. In looking at the Dial app entry in the CDR for one of these calls this appears:
Application: Dial(SIP/2303,HhTtrb(func-apply-sipheaders^s^1))
I am interested in the HhTtrb portion Other calls have the same only Hhtrlb

Thanks for any help

2

Application_Dial - Asterisk Project - Asterisk Project Wiki

  • options
    • A - Play an announcement to the called party, where x is the prompt to be played
      • x - The file to play to the called party
    • a - Immediately answer the calling channel when the called channel answers in all cases. Normally, the calling channel is answered when the called channel answers, but when options such as A() and M() are used, the calling channel is not answered until all actions on the called channel (such as playing an announcement) are completed. This option can be used to answer the calling channel before doing anything on the called channel. You will rarely need to use this option, the default behavior is adequate in most cases.
    • C - Reset the call detail record (CDR) for this call.
    • c - If the Dial() application cancels this call, always set the flag to tell the channel driver that the call is answered elsewhere.
    • d - Allow the calling user to dial a 1 digit extension while waiting for a call to be answered. Exit to that extension if it exists in the current context, or the context defined in the EXITCONTEXT variable, if it exists.
    • D - Send the specified DTMF strings after the called party has answered, but before the call gets bridged. The called DTMF string is sent to the called party, and the calling DTMF string is sent to the calling party. Both arguments can be used alone. If progress is specified, its DTMF is sent immediately after receiving a PROGRESS message.
      • called
      • calling
      • progress
    • e - Execute the h extension for peer after the call ends
    • f - If x is not provided, force the CallerID sent on a call-forward or deflection to the dialplan extension of this Dial() using a dialplan hint. For example, some PSTNs do not allow CallerID to be set to anything other than the numbers assigned to you. If x is provided, force the CallerID sent to x.
      • x
    • F - When the caller hangs up, transfer the called party to the specified destination and start execution at that location.
      • context
      • exten
      • priority
    • F - When the caller hangs up, transfer the called party to the next priority of the current extension and start execution at that location.
    • g - Proceed with dialplan execution at the next priority in the current extension if the destination channel hangs up.
    • G - If the call is answered, transfer the calling party to the specified priority and the called party to the specified priority plus one.
      • context
      • exten
      • priority
    • h - Allow the called party to hang up by sending the DTMF sequence defined for disconnect in features.conf.
    • H - Allow the calling party to hang up by sending the DTMF sequence defined for disconnect in features.conf.
    • i - Asterisk will ignore any forwarding requests it may receive on this dial attempt.
    • I - Asterisk will ignore any connected line update requests or any redirecting party update requests it may receive on this dial attempt.
    • k - Allow the called party to enable parking of the call by sending the DTMF sequence defined for call parking in features.conf.
    • K - Allow the calling party to enable parking of the call by sending the DTMF sequence defined for call parking in features.conf.
    • L - Limit the call to x milliseconds. Play a warning when y milliseconds are left. Repeat the warning every z milliseconds until time expires.
      This option is affected by the following variables:
      • LIMIT_PLAYAUDIO_CALLER - If set, this variable causes Asterisk to play the prompts to the caller.
        • YES default: (true)
        • NO
      • LIMIT_PLAYAUDIO_CALLEE - If set, this variable causes Asterisk to play the prompts to the callee.
        • YES
        • NO default: (true)
      • LIMIT_TIMEOUT_FILE - If specified, filename specifies the sound prompt to play when the timeout is reached. If not set, the time remaining will be announced.
        • FILENAME
      • LIMIT_CONNECT_FILE - If specified, filename specifies the sound prompt to play when the call begins. If not set, the time remaining will be announced.
        • FILENAME
      • LIMIT_WARNING_FILE - If specified, filename specifies the sound prompt to play as a warning when time x is reached. If not set, the time remaining will be announced.
        • FILENAME
      • x - Maximum call time, in milliseconds
      • y - Warning time, in milliseconds
      • z - Repeat time, in milliseconds
    • m - Provide hold music to the calling party until a requested channel answers. A specific music on hold class (as defined in musiconhold.conf) can be specified.
      • class
    • M - Execute the specified macro for the called channel before connecting to the calling channel. Arguments can be specified to the Macro using ^ as a delimiter. The macro can set the variable MACRO_RESULT to specify the following actions after the macro is finished executing:
      • MACRO_RESULT - If set, this action will be taken after the macro finished executing.
        • ABORT - Hangup both legs of the call
        • CONGESTION - Behave as if line congestion was encountered
        • BUSY - Behave as if a busy signal was encountered
        • CONTINUE - Hangup the called party and allow the calling party to continue dialplan execution at the next priority
        • GOTO:[[^]^] - Transfer the call to the specified destination.
      • macro - Name of the macro that should be executed.
      • arg - Macro arguments
    • n - This option is a modifier for the call screening/privacy mode. (See the p and P options.) It specifies that no introductions are to be saved in the priv-callerintros directory.
      • delete - With delete either not specified or set to 0, the recorded introduction will not be deleted if the caller hangs up while the remote party has not yet answered.
        With delete set to 1, the introduction will always be deleted.
    • N - This option is a modifier for the call screening/privacy mode. It specifies that if Caller*ID is present, do not screen the call.
    • o - If x is not provided, specify that the CallerID that was present on the calling channel be stored as the CallerID on the called channel. This was the behavior of Asterisk 1.0 and earlier. If x is provided, specify the CallerID stored on the called channel. Note that o(${CALLERID(all)}) is similar to option o without the parameter.
      • x
    • O - Enables operator services mode. This option only works when bridging a DAHDI channel to another DAHDI channel only. if specified on non-DAHDI interfaces, it will be ignored. When the destination answers (presumably an operator services station), the originator no longer has control of their line. They may hang up, but the switch will not release their line until the destination party (the operator) hangs up.
      • mode - With mode either not specified or set to 1, the originator hanging up will cause the phone to ring back immediately.
        With mode set to 2, when the operator flashes the trunk, it will ring their phone back.
    • p - This option enables screening mode. This is basically Privacy mode without memory.
    • P - Enable privacy mode. Use x as the family/key in the AstDB database if it is provided. The current extension is used if a database family/key is not specified.
      • x
    • r - Default: Indicate ringing to the calling party, even if the called party isn’t actually ringing. Pass no audio to the calling party until the called channel has answered.
      • tone - Indicate progress to calling party. Send audio ‘tone’ from indications.conf
    • S - Hang up the call x seconds after the called party has answered the call.
      • x
    • s - Force the outgoing callerid tag parameter to be set to the string x.
      Works with the f option.
      • x
    • t - Allow the called party to transfer the calling party by sending the DTMF sequence defined in features.conf. This setting does not perform policy enforcement on transfers initiated by other methods.
    • T - Allow the calling party to transfer the called party by sending the DTMF sequence defined in features.conf. This setting does not perform policy enforcement on transfers initiated by other methods.
    • U - Execute via Gosub the routine x for the called channel before connecting to the calling channel. Arguments can be specified to the Gosub using ^ as a delimiter. The Gosub routine can set the variable GOSUB_RESULT to specify the following actions after the Gosub returns.
      • GOSUB_RESULT
        • ABORT - Hangup both legs of the call.
        • CONGESTION - Behave as if line congestion was encountered.
        • BUSY - Behave as if a busy signal was encountered.
        • CONTINUE - Hangup the called party and allow the calling party to continue dialplan execution at the next priority.
        • GOTO:[[^]^] - Transfer the call to the specified destination.
      • x - Name of the subroutine to execute via Gosub
      • arg - Arguments for the Gosub routine
    • u - Works with the f option.
      • x - Force the outgoing callerid presentation indicator parameter to be set to one of the values passed in x: allowed_not_screened allowed_passed_screen allowed_failed_screen allowed prohib_not_screened prohib_passed_screen prohib_failed_screen prohib unavailable
    • w - Allow the called party to enable recording of the call by sending the DTMF sequence defined for one-touch recording in features.conf.
    • W - Allow the calling party to enable recording of the call by sending the DTMF sequence defined for one-touch recording in features.conf.
    • x - Allow the called party to enable recording of the call by sending the DTMF sequence defined for one-touch automixmonitor in features.conf.
    • X - Allow the calling party to enable recording of the call by sending the DTMF sequence defined for one-touch automixmonitor in features.conf.
    • z - On a call forward, cancel any dial timeout which has been set for this call.lockquote
3

Thanks James!

4

Be wary of the T and t options, possible malfeasance, especially when they occur ‘out of hours’

5

How would you correct that?

6

Well you need to consider that the T and t options are for In-Call Feature Codes. Meaning that they allow the use of *2 for attended transfers and ## for blind transfers. This in no way blocks other methods of transfer such as REFER requests.

The dial options used are based on the type of call it is. Internal calls will allow both options because each side can transfer via feature codes. An external inbound call going to an extension will allow the callee (local extension) to transfer but not the caller. Opposite for outbound calls, the caller (local extension) can transfer the call but the callee (PSTN side) cannot. When an external inbound call is forwarded/passed through the system, the option for either party to use feature code transfers is removed. So an incoming call going back out due to call forwarding or followme, for example, loss the ability to use in-call transfer features unless you override the defaults yourself.

7

Thanks Tom for that explanation.

8

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