[asterisk-users] Default delay time for Attended call

Thu Jan 31 10:46:27 CST 2008

A call comes in from the PSTN, Asterisk answers it, it goes to the directory, and then to the extension the caller designates and the user at that extension answers.  The user at the extension then wants to transfer the call to another extension; on the Cisco 7940 they push the “more” soft key, then the “Transfer” soft key, then enter the extension number they want to transfer to, and hit the “dial” soft key.  The user at the new extension answers and the talks to the user doing the transfer.  They agree to transfer the call to the new extension and the person who got the original call then hits the “transfer” soft key and hangs up.  6 seconds later the caller and the new extension can talk to each other.  The line at the new extension is silent for those 6 seconds.

Thank you for help.

Here is the extension.cong file:

[general]

static = yes

writeprotect = no

clearglobalvars = no

[globals]

CONSOLE = Console/dsp

[trunkint]

exten => _9011.,1,Dial(Zap/g1/${EXTEN:1},70,Tt) 

[trunkld]

exten => _91NXXNXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})

[trunklocal]

exten => _9NXXXXXX,1,Dial(Zap/g1/${EXTEN:1},70,Tt)

[trunktollfree]

exten => _91800NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})

exten => _91888NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})

exten => _91877NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})

exten => _91866NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})

[international]

include => longdistance 

include => trunkint

[longdistance]

ignorepat => 9

include => local

include => trunkld

[local]

ignorepat => 9 

include => default 

include => trunklocal 

include => trunktollfree 

include => longdistance

include => parkedcalls

[macro-trunkdial]

exten => s,1,Dial(${ARG1})

exten => s,n,Goto(s-${DIALSTATUS},1)

exten => s-NOANSWER,1,Hangup

exten => s-BUSY,1,Hangup

exten => _s-.,1,NoOp

[macro-stdexten]

exten => s,1,Dial(${ARG2},20)             ; Ring the interface, 20 seconds maximum 

exten => s,2,Goto(s-${DIALSTATUS},1)      ; Jump based on status 

(NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)

exten => s-NOANSWER,1,Voicemail(${ARG1},u)  ; If unavailable, send to voicemail w/ unavail announce

exten => s-BUSY,1,Voicemail(${ARG1},b)  ; If busy, send to voicemail w/ busy announce

exten => _s-.,1,Goto(s-NOANSWER,1)        ; Treat anything else as no answer 

exten => a,1,VoicemailMain(${ARG1})       ; If they press *, send the user into VoicemailMain

[macro-stdPrivacyexten]

exten => s,1,Dial(${ARG2},20|p)  ; Ring the interface, 20 seconds maximum, call screening ; option (or use 

P for databased call screening) 

exten => s,2,Goto(s-${DIALSTATUS},1)  ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)

exten => s-NOANSWER,1,Voicemail(${ARG1},u)  ; If unavailable, send to voicemail w/ unavail announce

exten => s-BUSY,1,Voicemail(${ARG1},b)  ; If busy, send to voicemail w/ busy announce

exten => s-DONTCALL,1,Goto(${ARG3},s,1)  ; Callee chose to send this call to a polite "Don't call again" 

script.

exten => s-TORTURE,1,Goto(${ARG4},s,1)  ; Callee chose to send this call to a telemarketer torture script.

exten => _s-.,1,Goto(s-NOANSWER,1)  ; Treat anything else as no answer 

exten => a,1,VoicemailMain(${ARG1})  ; If they press *, send the user into VoicemailMain

[macro-page];

exten => s,1,ChanIsAvail(${ARG1}|js)  ; j is for Jump and s is for ANY call 

exten => s,n,GoToIf([${AVAILSTATUS} = "1"]?autoanswer:fail) 

exten => s,n(autoanswer),Set(_ALERT_INFO="RA")  ; This is for the PolyComs 

exten => s,n,SIPAddHeader(Call-Info: Answer-After=0)  ; This is for the Grandstream, Snoms, and Others 

exten => s,n,NoOp()  ; Add others here and Post on the Wiki!!!!

exten => s,n,Dial(${ARG1}||)

exten => s,n(fail),Hangup

[default]

exten => _9NXXXXXX,1,Dial(${TRUNK}/${EXTEN:1},70,Tt)

exten => _91NXXNXXXXXX,1,Dial(${TRUNK}/${EXTEN:1},70,Tt)

exten => _9011.,1,Dial(${TRUNK}/${EXTEN:1},70,Tt)

exten => _9911,1,Dial(${TRUNK}/${EXTEN:1},70,Tt)

exten => 6500,1,VoiceMailMain

exten => 6500,1,NoOp(${Exten:6:11})

exten => 6500,2,VoiceMailMain(s${CALLERID(all):1})

exten = o,1,Goto(default,6000,1) 

include => parkedcalls

;[asterisk_guitools]

;exten = executecommand,1,System(${command})

;exten = executecommand,n,Hangup()

;exten = record_vmenu,1,Answer

;exten = record_vmenu,n,Playback(vm-intro)

;exten = record_vmenu,n,Record(${var1})

;exten = record_vmenu,n,Playback(vm-saved)

;exten = record_vmenu,n,Playback(vm-goodbye)

;exten = record_vmenu,n,Hangup

;exten = play_file,1,Answer

;exten = play_file,n,Playback(${var1})

;exten = play_file,n,Hangup

;hasbeensetup = Y

[DID_trunk_1]

include = default

exten = s,1,Answer()

exten = s,n,NoOp(${CALLERID(num)})

exten = s,n,Directory(default||f)

[support]

include = default

exten = _X.,1,Goto(default|6009|1)

exten = s,1,Goto(default|6009|1)

[numberplan-custom-1]

plancomment = DialPlan1

include = default

exten = _9XXXXXXX,1,Macro(trunkdial,${TRUNK}/${EXTEN:1})

comment = _9XXXXXXX,1,Local,standard

exten = ,1,Macro(trunkdial,${TRUNK}/${EXTEN:0})

comment = ,1,Emergency911,custom

exten = _9XXXXXXXXXXX!,1,Macro(trunkdial,${TRUNK}/${EXTEN:1})

comment = _9XXXXXXXXXXX!,1,Longdistance,standard

Date: Thu, 31 Jan 2008 09:04:03 -0500

From: "C F" <shmaltz at gmail.com>

Subject: Re: [asterisk-users] Default delay time for Attended call

      transfer

To: "Asterisk Users Mailing List - Non-Commercial Discussion"

      <asterisk-users at lists.digium.com>

Message-ID:

      <81000b5a0801310604r27004306q6e6ed333acab24ba at mail.gmail.com>

Content-Type: text/plain; charset=ISO-8859-1

First time or second time they hit transfer?

Dial plan config?

2008/1/30 Don Smith <HYPERLINK "mailto:Don_Smith at catalysthealthcare.ca"Don_Smith at catalysthealthcare.ca>:

> 

> 

> 

> 

> Greetings,

> 

> I have an issue with the length of time that passes from when someone hits

> the transfer soft key on a Cisco 7940, after doing an attended transfer, and

> when the recipient's connects with the transferred call.  It appears to be

> around 6 seconds.  Is there a .conf in Asterisk where this time can be

> reduced?

> 

> 

> 

> Thank you for your help

> 

> Don

> 

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